Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

896 files changed, 33775 insertions, 36158 deletions

diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 839a1439e6..000dd394df 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -11,10 +11,10 @@ import("../../build/webrtc.gni")
 
 source_set("rent_a_codec") {
   sources = [
-    "main/acm2/acm_codec_database.cc",
-    "main/acm2/acm_codec_database.h",
-    "main/acm2/rent_a_codec.cc",
-    "main/acm2/rent_a_codec.h",
+    "acm2/acm_codec_database.cc",
+    "acm2/acm_codec_database.h",
+    "acm2/rent_a_codec.cc",
+    "acm2/rent_a_codec.h",
   ]
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
@@ -44,31 +44,29 @@ source_set("rent_a_codec") {
 
 config("audio_coding_config") {
   include_dirs = [
-    "main/include",
-    "../interface",
+    "include",
+    "../include",
   ]
 }
 
 source_set("audio_coding") {
   sources = [
-    "main/acm2/acm_common_defs.h",
-    "main/acm2/acm_receiver.cc",
-    "main/acm2/acm_receiver.h",
-    "main/acm2/acm_resampler.cc",
-    "main/acm2/acm_resampler.h",
-    "main/acm2/audio_coding_module.cc",
-    "main/acm2/audio_coding_module_impl.cc",
-    "main/acm2/audio_coding_module_impl.h",
-    "main/acm2/call_statistics.cc",
-    "main/acm2/call_statistics.h",
-    "main/acm2/codec_manager.cc",
-    "main/acm2/codec_manager.h",
-    "main/acm2/codec_owner.cc",
-    "main/acm2/codec_owner.h",
-    "main/acm2/initial_delay_manager.cc",
-    "main/acm2/initial_delay_manager.h",
-    "main/include/audio_coding_module.h",
-    "main/include/audio_coding_module_typedefs.h",
+    "acm2/acm_common_defs.h",
+    "acm2/acm_receiver.cc",
+    "acm2/acm_receiver.h",
+    "acm2/acm_resampler.cc",
+    "acm2/acm_resampler.h",
+    "acm2/audio_coding_module.cc",
+    "acm2/audio_coding_module_impl.cc",
+    "acm2/audio_coding_module_impl.h",
+    "acm2/call_statistics.cc",
+    "acm2/call_statistics.h",
+    "acm2/codec_manager.cc",
+    "acm2/codec_manager.h",
+    "acm2/initial_delay_manager.cc",
+    "acm2/initial_delay_manager.h",
+    "include/audio_coding_module.h",
+    "include/audio_coding_module_typedefs.h",
   ]
 
   defines = []
@@ -166,11 +164,11 @@ config("cng_config") {
 source_set("cng") {
   sources = [
     "codecs/cng/audio_encoder_cng.cc",
+    "codecs/cng/audio_encoder_cng.h",
     "codecs/cng/cng_helpfuns.c",
     "codecs/cng/cng_helpfuns.h",
-    "codecs/cng/include/audio_encoder_cng.h",
-    "codecs/cng/include/webrtc_cng.h",
     "codecs/cng/webrtc_cng.c",
+    "codecs/cng/webrtc_cng.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -181,8 +179,8 @@ source_set("cng") {
   ]
 
   deps = [
-    "../../common_audio",
     ":audio_encoder_interface",
+    "../../common_audio",
   ]
 }
 
@@ -204,8 +202,8 @@ source_set("red") {
   ]
 
   deps = [
-    "../../common_audio",
     ":audio_encoder_interface",
+    "../../common_audio",
   ]
 }
 
@@ -219,13 +217,13 @@ config("g711_config") {
 source_set("g711") {
   sources = [
     "codecs/g711/audio_decoder_pcm.cc",
+    "codecs/g711/audio_decoder_pcm.h",
     "codecs/g711/audio_encoder_pcm.cc",
+    "codecs/g711/audio_encoder_pcm.h",
     "codecs/g711/g711.c",
     "codecs/g711/g711.h",
     "codecs/g711/g711_interface.c",
-    "codecs/g711/include/audio_decoder_pcm.h",
-    "codecs/g711/include/audio_encoder_pcm.h",
-    "codecs/g711/include/g711_interface.h",
+    "codecs/g711/g711_interface.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -236,6 +234,7 @@ source_set("g711") {
   ]
 
   deps = [
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
   ]
 }
@@ -250,14 +249,14 @@ config("g722_config") {
 source_set("g722") {
   sources = [
     "codecs/g722/audio_decoder_g722.cc",
+    "codecs/g722/audio_decoder_g722.h",
     "codecs/g722/audio_encoder_g722.cc",
+    "codecs/g722/audio_encoder_g722.h",
     "codecs/g722/g722_decode.c",
     "codecs/g722/g722_enc_dec.h",
     "codecs/g722/g722_encode.c",
     "codecs/g722/g722_interface.c",
-    "codecs/g722/include/audio_decoder_g722.h",
-    "codecs/g722/include/audio_encoder_g722.h",
-    "codecs/g722/include/g722_interface.h",
+    "codecs/g722/g722_interface.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -268,6 +267,7 @@ source_set("g722") {
   ]
 
   deps = [
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
   ]
 }
@@ -286,7 +286,9 @@ source_set("ilbc") {
     "codecs/ilbc/abs_quant_loop.c",
     "codecs/ilbc/abs_quant_loop.h",
     "codecs/ilbc/audio_decoder_ilbc.cc",
+    "codecs/ilbc/audio_decoder_ilbc.h",
     "codecs/ilbc/audio_encoder_ilbc.cc",
+    "codecs/ilbc/audio_encoder_ilbc.h",
     "codecs/ilbc/augmented_cb_corr.c",
     "codecs/ilbc/augmented_cb_corr.h",
     "codecs/ilbc/bw_expand.c",
@@ -351,9 +353,7 @@ source_set("ilbc") {
     "codecs/ilbc/hp_output.c",
     "codecs/ilbc/hp_output.h",
     "codecs/ilbc/ilbc.c",
-    "codecs/ilbc/include/audio_decoder_ilbc.h",
-    "codecs/ilbc/include/audio_encoder_ilbc.h",
-    "codecs/ilbc/include/ilbc.h",
+    "codecs/ilbc/ilbc.h",
     "codecs/ilbc/index_conv_dec.c",
     "codecs/ilbc/index_conv_dec.h",
     "codecs/ilbc/index_conv_enc.c",
@@ -434,8 +434,9 @@ source_set("ilbc") {
   ]
 
   deps = [
-    "../../common_audio",
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
+    "../../common_audio",
   ]
 }
 
@@ -604,6 +605,7 @@ source_set("isac_fix") {
   ]
 
   deps = [
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":isac_common",
     "../../common_audio",
@@ -696,14 +698,15 @@ config("pcm16b_config") {
 source_set("pcm16b") {
   sources = [
     "codecs/pcm16b/audio_decoder_pcm16b.cc",
+    "codecs/pcm16b/audio_decoder_pcm16b.h",
     "codecs/pcm16b/audio_encoder_pcm16b.cc",
-    "codecs/pcm16b/include/audio_decoder_pcm16b.h",
-    "codecs/pcm16b/include/audio_encoder_pcm16b.h",
-    "codecs/pcm16b/include/pcm16b.h",
+    "codecs/pcm16b/audio_encoder_pcm16b.h",
     "codecs/pcm16b/pcm16b.c",
+    "codecs/pcm16b/pcm16b.h",
   ]
 
   deps = [
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":g711",
   ]
@@ -723,16 +726,18 @@ config("opus_config") {
 source_set("webrtc_opus") {
   sources = [
     "codecs/opus/audio_decoder_opus.cc",
+    "codecs/opus/audio_decoder_opus.h",
     "codecs/opus/audio_encoder_opus.cc",
-    "codecs/opus/include/audio_decoder_opus.h",
-    "codecs/opus/include/audio_encoder_opus.h",
-    "codecs/opus/include/opus_interface.h",
+    "codecs/opus/audio_encoder_opus.h",
     "codecs/opus/opus_inst.h",
     "codecs/opus/opus_interface.c",
+    "codecs/opus/opus_interface.h",
   ]
 
   deps = [
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
+    "../../base:rtc_base_approved",
   ]
 
   if (rtc_build_opus) {
diff --git a/webrtc/modules/audio_coding/OWNERS b/webrtc/modules/audio_coding/OWNERS
index f43fb1f3ee..77db17d1c5 100644
--- a/webrtc/modules/audio_coding/OWNERS
+++ b/webrtc/modules/audio_coding/OWNERS
@@ -5,8 +5,6 @@ kwiberg@webrtc.org
 minyue@webrtc.org
 jan.skoglund@webrtc.org
 
-per-file *.isolate=kjellander@webrtc.org
-
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
 per-file *.gyp=*
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
index f7842ce5b1..5f3c07802b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
@@ -15,12 +15,12 @@
 
 // TODO(tlegrand): Change constant input pointers in all functions to constant
 // references, where appropriate.
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
 
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -29,11 +29,6 @@ namespace acm2 {
 
 namespace {
 
-// Checks if the bitrate is valid for the codec.
-bool IsRateValid(int codec_id, int rate) {
-  return ACMCodecDB::database_[codec_id].rate == rate;
-}
-
 // Checks if the bitrate is valid for iSAC.
 bool IsISACRateValid(int rate) {
   return (rate == -1) || ((rate <= 56000) && (rate >= 10000));
@@ -111,7 +106,7 @@ const CodecInst ACMCodecDB::database_[] = {
   {127, "red", 8000, 0, 1, 0},
 #endif
   // To prevent compile errors due to trailing commas.
-  {-1, "Null", -1, -1, -1, -1}
+  {-1, "Null", -1, -1, 0, -1}
 };
 
 // Create database with all codec settings at compile time.
@@ -167,7 +162,7 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
     {1, {0}, 0, 1},
 #endif
     // To prevent compile errors due to trailing commas.
-    {-1, {-1}, -1, -1}
+    {-1, {-1}, -1, 0}
 };
 
 // Create a database of all NetEQ decoders at compile time.
@@ -214,20 +209,6 @@ const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
 #endif
 };
 
-// Get codec information from database.
-// TODO(tlegrand): replace memcpy with a pointer to the data base memory.
-int ACMCodecDB::Codec(int codec_id, CodecInst* codec_inst) {
-  // Error check to see that codec_id is not out of bounds.
-  if (static_cast<size_t>(codec_id) >= RentACodec::NumberOfCodecs()) {
-    return -1;
-  }
-
-  // Copy database information for the codec to the output.
-  memcpy(codec_inst, &database_[codec_id], sizeof(CodecInst));
-
-  return 0;
-}
-
 // Enumerator for error codes when asking for codec database id.
 enum {
   kInvalidCodec = -10,
@@ -249,7 +230,7 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
   }
 
   // Checks the validity of payload type
-  if (!ValidPayloadType(codec_inst.pltype)) {
+  if (!RentACodec::IsPayloadTypeValid(codec_inst.pltype)) {
     return kInvalidPayloadtype;
   }
 
@@ -298,8 +279,7 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
         ? codec_id : kInvalidRate;
   }
 
-  return IsRateValid(codec_id, codec_inst.rate) ?
-      codec_id : kInvalidRate;
+  return database_[codec_id].rate == codec_inst.rate ? codec_id : kInvalidRate;
 }
 
 // Looks for a matching payload name, frequency, and channels in the
@@ -312,7 +292,9 @@ int ACMCodecDB::CodecId(const CodecInst& codec_inst) {
                   codec_inst.channels));
 }
 
-int ACMCodecDB::CodecId(const char* payload_name, int frequency, int channels) {
+int ACMCodecDB::CodecId(const char* payload_name,
+                        int frequency,
+                        size_t channels) {
   for (const CodecInst& ci : RentACodec::Database()) {
     bool name_match = false;
     bool frequency_match = false;
@@ -346,19 +328,6 @@ int ACMCodecDB::ReceiverCodecNumber(const CodecInst& codec_inst) {
   return CodecId(codec_inst);
 }
 
-// Returns the codec sampling frequency for codec with id = "codec_id" in
-// database.
-int ACMCodecDB::CodecFreq(int codec_id) {
-  const size_t i = static_cast<size_t>(codec_id);
-  const auto db = RentACodec::Database();
-  return i < db.size() ? db[i].plfreq : -1;
-}
-
-// Checks if the payload type is in the valid range.
-bool ACMCodecDB::ValidPayloadType(int payload_type) {
-  return (payload_type >= 0) && (payload_type <= 127);
-}
-
 }  // namespace acm2
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/acm2/acm_codec_database.h
index 84c8846a57..6c2db9cfc8 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.h
@@ -13,12 +13,12 @@
  * codecs.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
 namespace webrtc {
@@ -48,20 +48,9 @@ class ACMCodecDB {
     int num_packet_sizes;
     int packet_sizes_samples[kMaxNumPacketSize];
     int basic_block_samples;
-    int channel_support;
+    size_t channel_support;
   };
 
-  // Gets codec information from database at the position in database given by
-  // [codec_id].
-  // Input:
-  //   [codec_id] - number that specifies at what position in the database to
-  //                get the information.
-  // Output:
-  //   [codec_inst] - filled with information about the codec.
-  // Return:
-  //   0 if successful, otherwise -1.
-  static int Codec(int codec_id, CodecInst* codec_inst);
-
   // Returns codec id from database, given the information received in the input
   // [codec_inst].
   // Input:
@@ -71,26 +60,9 @@ class ACMCodecDB {
   //   codec id if successful, otherwise < 0.
   static int CodecNumber(const CodecInst& codec_inst);
   static int CodecId(const CodecInst& codec_inst);
-  static int CodecId(const char* payload_name, int frequency, int channels);
+  static int CodecId(const char* payload_name, int frequency, size_t channels);
   static int ReceiverCodecNumber(const CodecInst& codec_inst);
 
-  // Returns the codec sampling frequency for codec with id = "codec_id" in
-  // database.
-  // TODO(tlegrand): Check if function is needed, or if we can change
-  // to access database directly.
-  // Input:
-  //   [codec_id] - number that specifies at what position in the database to
-  //                get the information.
-  // Return:
-  //   codec sampling frequency if successful, otherwise -1.
-  static int CodecFreq(int codec_id);
-
-  // Check if the payload type is valid, meaning that it is in the valid range
-  // of 0 to 127.
-  // Input:
-  //   [payload_type] - payload type.
-  static bool ValidPayloadType(int payload_type);
-
   // Databases with information about the supported codecs
   // database_ - stored information about all codecs: payload type, name,
   //             sampling frequency, packet size in samples, default channel
@@ -101,15 +73,11 @@ class ACMCodecDB {
   // neteq_decoders_ - list of supported decoders in NetEQ.
   static const CodecInst database_[kMaxNumCodecs];
   static const CodecSettings codec_settings_[kMaxNumCodecs];
-
- private:
   static const NetEqDecoder neteq_decoders_[kMaxNumCodecs];
-
-  friend class RentACodec;
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/acm2/acm_common_defs.h
index 23e3519ed0..483bdd93f1 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/acm2/acm_common_defs.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
 
 #include "webrtc/engine_configurations.h"
 
@@ -29,4 +29,4 @@ const int kIsacPacSize960 = 960;
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc b/webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc
index 607b933deb..607b933deb 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
index fdcfdfc22d..855a39e675 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
 
 #include <assert.h>
 #include <stdio.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
@@ -55,7 +55,7 @@ bool ModifyAndUseThisCodec(CodecInst* codec_param) {
 // G.722 = 94
 bool RemapPltypeAndUseThisCodec(const char* plname,
                                 int plfreq,
-                                int channels,
+                                size_t channels,
                                 int* pltype) {
   if (channels != 1)
     return false;  // Don't use non-mono codecs.
@@ -144,9 +144,10 @@ int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
     int rtp_payload_type,
     AudioDecoder* external_decoder,
     int sample_rate_hz,
-    int num_channels) {
+    int num_channels,
+    const std::string& name) {
   return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
-                                            sample_rate_hz, num_channels);
+                                            sample_rate_hz, num_channels, name);
 }
 
 void AcmReceiveTestOldApi::Run() {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
index 0b5671fe8c..3010ec72b1 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -48,7 +50,8 @@ class AcmReceiveTestOldApi {
   int RegisterExternalReceiveCodec(int rtp_payload_type,
                                    AudioDecoder* external_decoder,
                                    int sample_rate_hz,
-                                   int num_channels);
+                                   int num_channels,
+                                   const std::string& name);
 
   // Runs the test and returns true if successful.
   void Run();
@@ -91,4 +94,4 @@ class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
 
 }  // namespace test
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index cf486ce06a..f45d5d3414 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 
 #include <stdlib.h>  // malloc
 
@@ -21,9 +21,8 @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
@@ -124,27 +123,13 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
       id_(config.id),
       last_audio_decoder_(nullptr),
       previous_audio_activity_(AudioFrame::kVadPassive),
-      current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
       audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       neteq_(NetEq::Create(config.neteq_config)),
-      vad_enabled_(true),
+      vad_enabled_(config.neteq_config.enable_post_decode_vad),
       clock_(config.clock),
-      resampled_last_output_frame_(true),
-      av_sync_(false),
-      initial_delay_manager_(),
-      missing_packets_sync_stream_(),
-      late_packets_sync_stream_() {
+      resampled_last_output_frame_(true) {
   assert(clock_);
-
-  // Make sure we are on the same page as NetEq. Post-decode VAD is disabled by
-  // default in NetEq4, however, Audio Conference Mixer relies on VAD decision
-  // and fails if VAD decision is not provided.
-  if (vad_enabled_)
-    neteq_->EnableVad();
-  else
-    neteq_->DisableVad();
-
   memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
   memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
 }
@@ -160,42 +145,6 @@ int AcmReceiver::SetMinimumDelay(int delay_ms) {
   return -1;
 }
 
-int AcmReceiver::SetInitialDelay(int delay_ms) {
-  if (delay_ms < 0 || delay_ms > 10000) {
-    return -1;
-  }
-  CriticalSectionScoped lock(crit_sect_.get());
-
-  if (delay_ms == 0) {
-    av_sync_ = false;
-    initial_delay_manager_.reset();
-    missing_packets_sync_stream_.reset();
-    late_packets_sync_stream_.reset();
-    neteq_->SetMinimumDelay(0);
-    return 0;
-  }
-
-  if (av_sync_ && initial_delay_manager_->PacketBuffered()) {
-    // Too late for this API. Only works before a call is started.
-    return -1;
-  }
-
-  // Most of places NetEq calls are not within AcmReceiver's critical section to
-  // improve performance. Here, this call has to be placed before the following
-  // block, therefore, we keep it inside critical section. Otherwise, we have to
-  // release |neteq_crit_sect_| and acquire it again, which seems an overkill.
-  if (!neteq_->SetMinimumDelay(delay_ms))
-    return -1;
-
-  const int kLatePacketThreshold = 5;
-  av_sync_ = true;
-  initial_delay_manager_.reset(new InitialDelayManager(delay_ms,
-                                                       kLatePacketThreshold));
-  missing_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
-  late_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
-  return 0;
-}
-
 int AcmReceiver::SetMaximumDelay(int delay_ms) {
   if (neteq_->SetMaximumDelay(delay_ms))
     return 0;
@@ -207,74 +156,52 @@ int AcmReceiver::LeastRequiredDelayMs() const {
   return neteq_->LeastRequiredDelayMs();
 }
 
-int AcmReceiver::current_sample_rate_hz() const {
+rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
   CriticalSectionScoped lock(crit_sect_.get());
-  return current_sample_rate_hz_;
+  return last_packet_sample_rate_hz_;
+}
+
+int AcmReceiver::last_output_sample_rate_hz() const {
+  return neteq_->last_output_sample_rate_hz();
 }
 
 int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
-                              const uint8_t* incoming_payload,
-                              size_t length_payload) {
+                              rtc::ArrayView<const uint8_t> incoming_payload) {
   uint32_t receive_timestamp = 0;
-  InitialDelayManager::PacketType packet_type =
-      InitialDelayManager::kUndefinedPacket;
-  bool new_codec = false;
   const RTPHeader* header = &rtp_header.header;  // Just a shorthand.
 
   {
     CriticalSectionScoped lock(crit_sect_.get());
 
-    const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload);
+    const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload[0]);
     if (!decoder) {
       LOG_F(LS_ERROR) << "Payload-type "
                       << static_cast<int>(header->payloadType)
                       << " is not registered.";
       return -1;
     }
-    const int sample_rate_hz = ACMCodecDB::CodecFreq(decoder->acm_codec_id);
+    const int sample_rate_hz = [&decoder] {
+      const auto ci = RentACodec::CodecIdFromIndex(decoder->acm_codec_id);
+      return ci ? RentACodec::CodecInstById(*ci)->plfreq : -1;
+    }();
     receive_timestamp = NowInTimestamp(sample_rate_hz);
 
-    if (IsCng(decoder->acm_codec_id)) {
-      // If this is a CNG while the audio codec is not mono skip pushing in
-      // packets into NetEq.
-      if (last_audio_decoder_ && last_audio_decoder_->channels > 1)
+    // If this is a CNG while the audio codec is not mono, skip pushing in
+    // packets into NetEq.
+    if (IsCng(decoder->acm_codec_id) && last_audio_decoder_ &&
+        last_audio_decoder_->channels > 1)
         return 0;
-      packet_type = InitialDelayManager::kCngPacket;
-    } else if (decoder->acm_codec_id ==
-               *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
-      packet_type = InitialDelayManager::kAvtPacket;
-    } else {
-      if (decoder != last_audio_decoder_) {
-        // This is either the first audio packet or send codec is changed.
-        // Therefore, either NetEq buffer is empty or will be flushed when this
-        // packet is inserted.
-        new_codec = true;
-        last_audio_decoder_ = decoder;
-      }
-      packet_type = InitialDelayManager::kAudioPacket;
+    if (!IsCng(decoder->acm_codec_id) &&
+        decoder->acm_codec_id !=
+            *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
+      last_audio_decoder_ = decoder;
+      last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz);
     }
 
-    if (av_sync_) {
-      assert(initial_delay_manager_.get());
-      assert(missing_packets_sync_stream_.get());
-      // This updates |initial_delay_manager_| and specifies an stream of
-      // sync-packets, if required to be inserted. We insert the sync-packets
-      // when AcmReceiver lock is released and |decoder_lock_| is acquired.
-      initial_delay_manager_->UpdateLastReceivedPacket(
-          rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz,
-          missing_packets_sync_stream_.get());
-    }
   }  // |crit_sect_| is released.
 
-  // If |missing_packets_sync_stream_| is allocated then we are in AV-sync and
-  // we may need to insert sync-packets. We don't check |av_sync_| as we are
-  // outside AcmReceiver's critical section.
-  if (missing_packets_sync_stream_.get()) {
-    InsertStreamOfSyncPackets(missing_packets_sync_stream_.get());
-  }
-
-  if (neteq_->InsertPacket(rtp_header, incoming_payload, length_payload,
-                           receive_timestamp) < 0) {
+  if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
+      0) {
     LOG(LERROR) << "AcmReceiver::InsertPacket "
                 << static_cast<int>(header->payloadType)
                 << " Failed to insert packet";
@@ -286,30 +213,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   enum NetEqOutputType type;
   size_t samples_per_channel;
-  int num_channels;
-  bool return_silence = false;
-
-  {
-    // Accessing members, take the lock.
-    CriticalSectionScoped lock(crit_sect_.get());
-
-    if (av_sync_) {
-      assert(initial_delay_manager_.get());
-      assert(late_packets_sync_stream_.get());
-      return_silence = GetSilence(desired_freq_hz, audio_frame);
-      uint32_t timestamp_now = NowInTimestamp(current_sample_rate_hz_);
-      initial_delay_manager_->LatePackets(timestamp_now,
-                                          late_packets_sync_stream_.get());
-    }
-  }
-
-  // If |late_packets_sync_stream_| is allocated then we have been in AV-sync
-  // mode and we might have to insert sync-packets.
-  if (late_packets_sync_stream_.get()) {
-    InsertStreamOfSyncPackets(late_packets_sync_stream_.get());
-    if (return_silence)  // Silence generated, don't pull from NetEq.
-      return 0;
-  }
+  size_t num_channels;
 
   // Accessing members, take the lock.
   CriticalSectionScoped lock(crit_sect_.get());
@@ -324,23 +228,18 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
     return -1;
   }
 
-  // NetEq always returns 10 ms of audio.
-  current_sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
+  const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
 
   // Update if resampling is required.
-  bool need_resampling = (desired_freq_hz != -1) &&
-      (current_sample_rate_hz_ != desired_freq_hz);
+  const bool need_resampling =
+      (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
 
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
-    int samples_per_channel_int =
-        resampler_.Resample10Msec(last_audio_buffer_.get(),
-                                  current_sample_rate_hz_,
-                                  desired_freq_hz,
-                                  num_channels,
-                                  AudioFrame::kMaxDataSizeSamples,
-                                  temp_output);
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        num_channels, AudioFrame::kMaxDataSizeSamples, temp_output);
     if (samples_per_channel_int < 0) {
       LOG(LERROR) << "AcmReceiver::GetAudio - "
                      "Resampling last_audio_buffer_ failed.";
@@ -354,13 +253,9 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   // TODO(henrik.lundin) Glitches in the output may appear if the output rate
   // from NetEq changes. See WebRTC issue 3923.
   if (need_resampling) {
-    int samples_per_channel_int =
-        resampler_.Resample10Msec(audio_buffer_.get(),
-                                  current_sample_rate_hz_,
-                                  desired_freq_hz,
-                                  num_channels,
-                                  AudioFrame::kMaxDataSizeSamples,
-                                  audio_frame->data_);
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        num_channels, AudioFrame::kMaxDataSizeSamples, audio_frame->data_);
     if (samples_per_channel_int < 0) {
       LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
       return -1;
@@ -406,16 +301,17 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
 
 int32_t AcmReceiver::AddCodec(int acm_codec_id,
                               uint8_t payload_type,
-                              int channels,
+                              size_t channels,
                               int sample_rate_hz,
-                              AudioDecoder* audio_decoder) {
+                              AudioDecoder* audio_decoder,
+                              const std::string& name) {
   const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
     if (acm_codec_id == -1)
       return NetEqDecoder::kDecoderArbitrary;  // External decoder.
-    const rtc::Maybe<RentACodec::CodecId> cid =
+    const rtc::Optional<RentACodec::CodecId> cid =
         RentACodec::CodecIdFromIndex(acm_codec_id);
     RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
-    const rtc::Maybe<NetEqDecoder> ned =
+    const rtc::Optional<NetEqDecoder> ned =
         RentACodec::NetEqDecoderFromCodecId(*cid, channels);
     RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
     return *ned;
@@ -447,10 +343,10 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id,
 
   int ret_val;
   if (!audio_decoder) {
-    ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type);
+    ret_val = neteq_->RegisterPayloadType(neteq_decoder, name, payload_type);
   } else {
-    ret_val = neteq_->RegisterExternalDecoder(audio_decoder, neteq_decoder,
-                                              payload_type, sample_rate_hz);
+    ret_val = neteq_->RegisterExternalDecoder(
+        audio_decoder, neteq_decoder, name, payload_type, sample_rate_hz);
   }
   if (ret_val != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
@@ -503,6 +399,7 @@ int AcmReceiver::RemoveAllCodecs() {
 
   // No codec is registered, invalidate last audio decoder.
   last_audio_decoder_ = nullptr;
+  last_packet_sample_rate_hz_ = rtc::Optional<int>();
   return ret_val;
 }
 
@@ -516,8 +413,10 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) {
     LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
     return -1;
   }
-  if (last_audio_decoder_ == &it->second)
+  if (last_audio_decoder_ == &it->second) {
     last_audio_decoder_ = nullptr;
+    last_packet_sample_rate_hz_ = rtc::Optional<int>();
+  }
   decoders_.erase(it);
   return 0;
 }
@@ -528,42 +427,16 @@ void AcmReceiver::set_id(int id) {
 }
 
 bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
-  if (av_sync_) {
-    assert(initial_delay_manager_.get());
-    if (initial_delay_manager_->buffering()) {
-      return initial_delay_manager_->GetPlayoutTimestamp(timestamp);
-    }
-  }
   return neteq_->GetPlayoutTimestamp(timestamp);
 }
 
-int AcmReceiver::last_audio_codec_id() const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  return last_audio_decoder_ ? last_audio_decoder_->acm_codec_id : -1;
-}
-
-int AcmReceiver::RedPayloadType() const {
-  const auto red_index =
-      RentACodec::CodecIndexFromId(RentACodec::CodecId::kRED);
-  if (red_index) {
-    CriticalSectionScoped lock(crit_sect_.get());
-    for (const auto& decoder_pair : decoders_) {
-      const Decoder& decoder = decoder_pair.second;
-      if (decoder.acm_codec_id == *red_index)
-        return decoder.payload_type;
-    }
-  }
-  LOG(WARNING) << "RED is not registered.";
-  return -1;
-}
-
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
   CriticalSectionScoped lock(crit_sect_.get());
   if (!last_audio_decoder_) {
     return -1;
   }
-  memcpy(codec, &ACMCodecDB::database_[last_audio_decoder_->acm_codec_id],
-         sizeof(CodecInst));
+  *codec = *RentACodec::CodecInstById(
+      *RentACodec::CodecIdFromIndex(last_audio_decoder_->acm_codec_id));
   codec->pltype = last_audio_decoder_->payload_type;
   codec->channels = last_audio_decoder_->channels;
   codec->plfreq = last_audio_decoder_->sample_rate_hz;
@@ -603,8 +476,8 @@ int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
     return -1;
   }
   const Decoder& decoder = it->second;
-  memcpy(codec, &ACMCodecDB::database_[decoder.acm_codec_id],
-         sizeof(CodecInst));
+  *codec = *RentACodec::CodecInstById(
+      *RentACodec::CodecIdFromIndex(decoder.acm_codec_id));
   codec->pltype = decoder.payload_type;
   codec->channels = decoder.channels;
   codec->plfreq = decoder.sample_rate_hz;
@@ -626,74 +499,20 @@ std::vector<uint16_t> AcmReceiver::GetNackList(
 }
 
 void AcmReceiver::ResetInitialDelay() {
-  {
-    CriticalSectionScoped lock(crit_sect_.get());
-    av_sync_ = false;
-    initial_delay_manager_.reset(NULL);
-    missing_packets_sync_stream_.reset(NULL);
-    late_packets_sync_stream_.reset(NULL);
-  }
   neteq_->SetMinimumDelay(0);
   // TODO(turajs): Should NetEq Buffer be flushed?
 }
 
-// This function is called within critical section, no need to acquire a lock.
-bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
-  assert(av_sync_);
-  assert(initial_delay_manager_.get());
-  if (!initial_delay_manager_->buffering()) {
-    return false;
-  }
-
-  // We stop accumulating packets, if the number of packets or the total size
-  // exceeds a threshold.
-  int num_packets;
-  int max_num_packets;
-  const float kBufferingThresholdScale = 0.9f;
-  neteq_->PacketBufferStatistics(&num_packets, &max_num_packets);
-  if (num_packets > max_num_packets * kBufferingThresholdScale) {
-    initial_delay_manager_->DisableBuffering();
-    return false;
-  }
-
-  // Update statistics.
-  call_stats_.DecodedBySilenceGenerator();
-
-  // Set the values if already got a packet, otherwise set to default values.
-  if (last_audio_decoder_) {
-    current_sample_rate_hz_ =
-        ACMCodecDB::database_[last_audio_decoder_->acm_codec_id].plfreq;
-    frame->num_channels_ = last_audio_decoder_->channels;
-  } else {
-    frame->num_channels_ = 1;
-  }
-
-  // Set the audio frame's sampling frequency.
-  if (desired_sample_rate_hz > 0) {
-    frame->sample_rate_hz_ = desired_sample_rate_hz;
-  } else {
-    frame->sample_rate_hz_ = current_sample_rate_hz_;
-  }
-
-  frame->samples_per_channel_ =
-      static_cast<size_t>(frame->sample_rate_hz_ / 100);  // Always 10 ms.
-  frame->speech_type_ = AudioFrame::kCNG;
-  frame->vad_activity_ = AudioFrame::kVadPassive;
-  size_t samples = frame->samples_per_channel_ * frame->num_channels_;
-  memset(frame->data_, 0, samples * sizeof(int16_t));
-  return true;
-}
-
 const AcmReceiver::Decoder* AcmReceiver::RtpHeaderToDecoder(
     const RTPHeader& rtp_header,
-    const uint8_t* payload) const {
+    uint8_t payload_type) const {
   auto it = decoders_.find(rtp_header.payloadType);
   const auto red_index =
       RentACodec::CodecIndexFromId(RentACodec::CodecId::kRED);
   if (red_index &&  // This ensures that RED is defined in WebRTC.
       it != decoders_.end() && it->second.acm_codec_id == *red_index) {
     // This is a RED packet, get the payload of the audio codec.
-    it = decoders_.find(payload[0] & 0x7F);
+    it = decoders_.find(payload_type & 0x7F);
   }
 
   // Check if the payload is registered.
@@ -711,23 +530,6 @@ uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
       (decoder_sampling_rate / 1000) * now_in_ms);
 }
 
-// This function only interacts with |neteq_|, therefore, it does not have to
-// be within critical section of AcmReceiver. It is inserting packets
-// into NetEq, so we call it when |decode_lock_| is acquired. However, this is
-// not essential as sync-packets do not interact with codecs (especially BWE).
-void AcmReceiver::InsertStreamOfSyncPackets(
-    InitialDelayManager::SyncStream* sync_stream) {
-  assert(sync_stream);
-  assert(av_sync_);
-  for (int n = 0; n < sync_stream->num_sync_packets; ++n) {
-    neteq_->InsertSyncPacket(sync_stream->rtp_info,
-                             sync_stream->receive_timestamp);
-    ++sync_stream->rtp_info.header.sequenceNumber;
-    sync_stream->rtp_info.header.timestamp += sync_stream->timestamp_step;
-    sync_stream->receive_timestamp += sync_stream->timestamp_step;
-  }
-}
-
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   CriticalSectionScoped lock(crit_sect_.get());
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 4775b8c6d9..b150612f69 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -8,23 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
 
 #include <map>
+#include <string>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -42,7 +44,7 @@ class AcmReceiver {
     uint8_t payload_type;
     // This field is meaningful for codecs where both mono and
     // stereo versions are registered under the same ID.
-    int channels;
+    size_t channels;
     int sample_rate_hz;
   };
 
@@ -66,8 +68,7 @@ class AcmReceiver {
   //                           <0 if NetEq returned an error.
   //
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                   const uint8_t* incoming_payload,
-                   size_t length_payload);
+                   rtc::ArrayView<const uint8_t> incoming_payload);
 
   //
   // Asks NetEq for 10 milliseconds of decoded audio.
@@ -115,9 +116,10 @@ class AcmReceiver {
   //
   int AddCodec(int acm_codec_id,
                uint8_t payload_type,
-               int channels,
+               size_t channels,
                int sample_rate_hz,
-               AudioDecoder* audio_decoder);
+               AudioDecoder* audio_decoder,
+               const std::string& name);
 
   //
   // Sets a minimum delay for packet buffer. The given delay is maintained,
@@ -151,29 +153,18 @@ class AcmReceiver {
   int LeastRequiredDelayMs() const;
 
   //
-  // Sets an initial delay of |delay_ms| milliseconds. This introduces a playout
-  // delay. Silence (zero signal) is played out until equivalent of |delay_ms|
-  // millisecond of audio is buffered. Then, NetEq maintains the delay.
-  //
-  // Input:
-  //   - delay_ms             : initial delay in milliseconds.
-  //
-  // Return value             : 0 if OK.
-  //                           <0 if NetEq returned an error.
-  //
-  int SetInitialDelay(int delay_ms);
-
-  //
   // Resets the initial delay to zero.
   //
   void ResetInitialDelay();
 
-  //
-  // Get the current sampling frequency in Hz.
-  //
-  // Return value             : Sampling frequency in Hz.
-  //
-  int current_sample_rate_hz() const;
+  // Returns the sample rate of the decoder associated with the last incoming
+  // packet. If no packet of a registered non-CNG codec has been received, the
+  // return value is empty. Also, if the decoder was unregistered since the last
+  // packet was inserted, the return value is empty.
+  rtc::Optional<int> last_packet_sample_rate_hz() const;
+
+  // Returns last_output_sample_rate_hz from the NetEq instance.
+  int last_output_sample_rate_hz() const;
 
   //
   // Get the current network statistics from NetEq.
@@ -231,13 +222,6 @@ class AcmReceiver {
   bool GetPlayoutTimestamp(uint32_t* timestamp);
 
   //
-  // Return the index of the codec associated with the last non-CNG/non-DTMF
-  // received payload. If no non-CNG/non-DTMF payload is received -1 is
-  // returned.
-  //
-  int last_audio_codec_id() const;  // TODO(turajs): can be inline.
-
-  //
   // Get the audio codec associated with the last non-CNG/non-DTMF received
   // payload. If no non-CNG/non-DTMF packet is received -1 is returned,
   // otherwise return 0.
@@ -245,11 +229,6 @@ class AcmReceiver {
   int LastAudioCodec(CodecInst* codec) const;
 
   //
-  // Return payload type of RED if it is registered, otherwise return -1;
-  //
-  int RedPayloadType() const;
-
-  //
   // Get a decoder given its registered payload-type.
   //
   // Input:
@@ -296,24 +275,16 @@ class AcmReceiver {
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
  private:
-  bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
-  int GetNumSyncPacketToInsert(uint16_t received_squence_number);
-
   const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
-                                    const uint8_t* payload) const
+                                    uint8_t payload_type) const
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
-  void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream);
-
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
   const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
-  int current_sample_rate_hz_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
   // Used in GetAudio, declared as member to avoid allocating every 10ms.
   // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
@@ -326,23 +297,11 @@ class AcmReceiver {
   bool vad_enabled_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
-
-  // Indicates if a non-zero initial delay is set, and the receiver is in
-  // AV-sync mode.
-  bool av_sync_;
-  rtc::scoped_ptr<InitialDelayManager> initial_delay_manager_;
-
-  // The following are defined as members to avoid creating them in every
-  // iteration. |missing_packets_sync_stream_| is *ONLY* used in InsertPacket().
-  // |late_packets_sync_stream_| is only used in GetAudio(). Both of these
-  // member variables are allocated only when we AV-sync is enabled, i.e.
-  // initial delay is set.
-  rtc::scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
-  rtc::scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
+  rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index f0caacce10..24ecc694ff 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -8,20 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 
 #include <algorithm>  // std::min
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -88,14 +86,14 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
   void TearDown() override {}
 
   void InsertOnePacketOfSilence(int codec_id) {
-    CodecInst codec;
-    ACMCodecDB::Codec(codec_id, &codec);
+    CodecInst codec =
+        *RentACodec::CodecInstById(*RentACodec::CodecIdFromIndex(codec_id));
     if (timestamp_ == 0) {  // This is the first time inserting audio.
       ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     } else {
-      CodecInst current_codec;
-      ASSERT_EQ(0, acm_->SendCodec(&current_codec));
-      if (!CodecsEqual(codec, current_codec))
+      auto current_codec = acm_->SendCodec();
+      ASSERT_TRUE(current_codec);
+      if (!CodecsEqual(codec, *current_codec))
         ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     }
     AudioFrame frame;
@@ -121,7 +119,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
       ASSERT_TRUE(i);
       ASSERT_EQ(
           0, receiver_->AddCodec(*i, codecs_[*i].pltype, codecs_[*i].channels,
-                                 codecs_[*i].plfreq, nullptr));
+                                 codecs_[*i].plfreq, nullptr, ""));
     }
   }
 
@@ -142,8 +140,9 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
       rtp_header_.type.Audio.isCNG = true;
     rtp_header_.header.timestamp = timestamp;
 
-    int ret_val = receiver_->InsertPacket(rtp_header_, payload_data,
-                                          payload_len_bytes);
+    int ret_val = receiver_->InsertPacket(
+        rtp_header_,
+        rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
     if (ret_val < 0) {
       assert(false);
       return -1;
@@ -164,13 +163,18 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
   FrameType last_frame_type_;
 };
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
+#else
+#define MAYBE_AddCodecGetCodec AddCodecGetCodec
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
   // Add codec.
   for (size_t n = 0; n < codecs_.size(); ++n) {
     if (n & 0x1)  // Just add codecs with odd index.
       EXPECT_EQ(0,
                 receiver_->AddCodec(n, codecs_[n].pltype, codecs_[n].channels,
-                                    codecs_[n].plfreq, NULL));
+                                    codecs_[n].plfreq, NULL, ""));
   }
   // Get codec and compare.
   for (size_t n = 0; n < codecs_.size(); ++n) {
@@ -188,7 +192,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
   }
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
+#else
+#define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
   const CodecIdInst codec1(RentACodec::CodecId::kPCMA);
   CodecInst codec2 = codec1.inst;
   ++codec2.pltype;
@@ -197,9 +206,9 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
   // Register the same codec with different payloads.
   EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec1.inst.pltype,
                                    codec1.inst.channels, codec1.inst.plfreq,
-                                   nullptr));
+                                   nullptr, ""));
   EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec2.pltype, codec2.channels,
-                                   codec2.plfreq, NULL));
+                                   codec2.plfreq, NULL, ""));
 
   // Both payload types should exist.
   EXPECT_EQ(0,
@@ -209,7 +218,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
   EXPECT_EQ(true, CodecsEqual(codec2, test_codec));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
+#else
+#define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
+#endif
+TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
   const CodecIdInst codec1(RentACodec::CodecId::kPCMU);
   CodecIdInst codec2(RentACodec::CodecId::kPCMA);
   codec2.inst.pltype = codec1.inst.pltype;
@@ -218,10 +232,10 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
   // Register the same payload type with different codec ID.
   EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec1.inst.pltype,
                                    codec1.inst.channels, codec1.inst.plfreq,
-                                   nullptr));
+                                   nullptr, ""));
   EXPECT_EQ(0, receiver_->AddCodec(codec2.id, codec2.inst.pltype,
                                    codec2.inst.channels, codec2.inst.plfreq,
-                                   nullptr));
+                                   nullptr, ""));
 
   // Make sure that the last codec is used.
   EXPECT_EQ(0,
@@ -229,12 +243,17 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
   EXPECT_EQ(true, CodecsEqual(codec2.inst, test_codec));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
+#else
+#define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
   const CodecIdInst codec(RentACodec::CodecId::kPCMA);
   const int payload_type = codec.inst.pltype;
   EXPECT_EQ(
       0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
-                             codec.inst.plfreq, nullptr));
+                             codec.inst.plfreq, nullptr, ""));
 
   // Remove non-existing codec should not fail. ACM1 legacy.
   EXPECT_EQ(0, receiver_->RemoveCodec(payload_type + 1));
@@ -247,7 +266,12 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
   EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &ci));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_SampleRate DISABLED_SampleRate
+#else
+#define MAYBE_SampleRate SampleRate
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
   const RentACodec::CodecId kCodecId[] = {RentACodec::CodecId::kISAC,
                                           RentACodec::CodecId::kISACSWB};
   AddSetOfCodecs(kCodecId);
@@ -261,18 +285,22 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
     for (int k = 0; k < num_10ms_frames; ++k) {
       EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
     }
-    EXPECT_EQ(std::min(32000, codec.inst.plfreq),
-              receiver_->current_sample_rate_hz());
+    EXPECT_EQ(codec.inst.plfreq, receiver_->last_output_sample_rate_hz());
   }
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
   receiver_->EnableVad();
   EXPECT_TRUE(receiver_->vad_enabled());
   const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
   ASSERT_EQ(
       0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
-                             codec.inst.plfreq, nullptr));
+                             codec.inst.plfreq, nullptr, ""));
   const int kNumPackets = 5;
   const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
   AudioFrame frame;
@@ -294,14 +322,13 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
   EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
 }
 
-#ifdef WEBRTC_CODEC_ISAC
-#define IF_ISAC_FLOAT(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
 #else
-#define IF_ISAC_FLOAT(x) DISABLED_##x
+#define MAYBE_LastAudioCodec LastAudioCodec
 #endif
-
-TEST_F(AcmReceiverTestOldApi,
-       DISABLED_ON_ANDROID(IF_ISAC_FLOAT(LastAudioCodec))) {
+#if defined(WEBRTC_CODEC_ISAC)
+TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
   const RentACodec::CodecId kCodecId[] = {
       RentACodec::CodecId::kISAC, RentACodec::CodecId::kPCMA,
       RentACodec::CodecId::kISACSWB, RentACodec::CodecId::kPCM16Bswb32kHz};
@@ -331,7 +358,7 @@ TEST_F(AcmReceiverTestOldApi,
 
   // Has received, only, DTX. Last Audio codec is undefined.
   EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
-  EXPECT_EQ(-1, receiver_->last_audio_codec_id());
+  EXPECT_FALSE(receiver_->last_packet_sample_rate_hz());
 
   for (auto id : kCodecId) {
     const CodecIdInst c(id);
@@ -345,7 +372,8 @@ TEST_F(AcmReceiverTestOldApi,
     // of type "speech."
     ASSERT_TRUE(packet_sent_);
     ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
-    EXPECT_EQ(c.id, receiver_->last_audio_codec_id());
+    EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
+              receiver_->last_packet_sample_rate_hz());
 
     // Set VAD on to send DTX. Then check if the "Last Audio codec" returns
     // the expected codec.
@@ -357,11 +385,13 @@ TEST_F(AcmReceiverTestOldApi,
       InsertOnePacketOfSilence(c.id);
       ASSERT_TRUE(packet_sent_);
     }
-    EXPECT_EQ(c.id, receiver_->last_audio_codec_id());
+    EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
+              receiver_->last_packet_sample_rate_hz());
     EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
     EXPECT_TRUE(CodecsEqual(c.inst, codec));
   }
 }
+#endif
 
 }  // namespace acm2
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
index cbcad85f5b..dfc3ef7e27 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 
 #include <assert.h>
 #include <string.h>
@@ -28,11 +28,10 @@ ACMResampler::~ACMResampler() {
 int ACMResampler::Resample10Msec(const int16_t* in_audio,
                                  int in_freq_hz,
                                  int out_freq_hz,
-                                 int num_audio_channels,
+                                 size_t num_audio_channels,
                                  size_t out_capacity_samples,
                                  int16_t* out_audio) {
-  size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
-  int out_length = out_freq_hz * num_audio_channels / 100;
+  size_t in_length = in_freq_hz * num_audio_channels / 100;
   if (in_freq_hz == out_freq_hz) {
     if (out_capacity_samples < in_length) {
       assert(false);
@@ -44,24 +43,20 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
 
   if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
                                     num_audio_channels) != 0) {
-    LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
-              num_audio_channels);
+    LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", " << out_freq_hz
+                  << ", " << num_audio_channels << ") failed.";
     return -1;
   }
 
-  out_length =
+  int out_length =
       resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
   if (out_length == -1) {
-    LOG_FERR4(LS_ERROR,
-              Resample,
-              in_audio,
-              in_length,
-              out_audio,
-              out_capacity_samples);
+    LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
+                  << out_audio << ", " << out_capacity_samples << ") failed.";
     return -1;
   }
 
-  return out_length / num_audio_channels;
+  return static_cast<int>(out_length / num_audio_channels);
 }
 
 }  // namespace acm2
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/webrtc/modules/audio_coding/acm2/acm_resampler.h
index a19b0c4569..268db8b752 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/typedefs.h"
@@ -25,7 +25,7 @@ class ACMResampler {
   int Resample10Msec(const int16_t* in_audio,
                      int in_freq_hz,
                      int out_freq_hz,
-                     int num_audio_channels,
+                     size_t num_audio_channels,
                      size_t out_capacity_samples,
                      int16_t* out_audio);
 
@@ -36,4 +36,4 @@ class ACMResampler {
 }  // namespace acm2
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
index ac38dc011d..3a89a77487 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
 
 #include <assert.h>
 #include <stdio.h>
@@ -17,7 +17,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
index 3e65ec6c2d..ce68196a3f 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
+++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
 
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
@@ -88,4 +88,4 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
 
 }  // namespace test
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 77ee0f2789..c4dd349cc4 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
@@ -56,8 +56,8 @@ int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
 int AudioCodingModule::Codec(const char* payload_name,
                              CodecInst* codec,
                              int sampling_freq_hz,
-                             int channels) {
-  rtc::Maybe<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
+                             size_t channels) {
+  rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
       payload_name, sampling_freq_hz, channels);
   if (ci) {
     *codec = *ci;
@@ -76,12 +76,13 @@ int AudioCodingModule::Codec(const char* payload_name,
 
 int AudioCodingModule::Codec(const char* payload_name,
                              int sampling_freq_hz,
-                             int channels) {
-  rtc::Maybe<acm2::RentACodec::CodecId> ci = acm2::RentACodec::CodecIdByParams(
-      payload_name, sampling_freq_hz, channels);
+                             size_t channels) {
+  rtc::Optional<acm2::RentACodec::CodecId> ci =
+      acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
+                                        channels);
   if (!ci)
     return -1;
-  rtc::Maybe<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
+  rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
   return i ? *i : -1;
 }
 
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
index b36c064800..ac302f0fe3 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
 
 #include <assert.h>
 #include <stdlib.h>
@@ -17,10 +17,10 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
@@ -97,7 +97,7 @@ void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
   if (value != last_value_ || first_time_) {
     first_time_ = false;
     last_value_ = value;
-    RTC_HISTOGRAM_COUNTS_100(histogram_name_, value);
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
   }
 }
 
@@ -133,7 +133,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   if (!HaveValidEncoder("Process"))
     return -1;
 
-  AudioEncoder* audio_encoder = codec_manager_.CurrentEncoder();
+  AudioEncoder* audio_encoder = rent_a_codec_.GetEncoderStack();
   // Scale the timestamp to the codec's RTP timestamp rate.
   uint32_t rtp_timestamp =
       first_frame_ ? input_data.input_timestamp
@@ -149,7 +149,9 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
 
   encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());
   encoded_info = audio_encoder->Encode(
-      rtp_timestamp, input_data.audio, input_data.length_per_channel,
+      rtp_timestamp, rtc::ArrayView<const int16_t>(
+                         input_data.audio, input_data.audio_channel *
+                                               input_data.length_per_channel),
       encode_buffer_.size(), encode_buffer_.data());
   encode_buffer_.SetSize(encoded_info.encoded_bytes);
   bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);
@@ -196,19 +198,43 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
 // Can be called multiple times for Codec, CNG, RED.
 int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.RegisterEncoder(send_codec);
+  if (!codec_manager_.RegisterEncoder(send_codec)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (!sp->speech_encoder && codec_manager_.GetCodecInst()) {
+    // We have no speech encoder, but we have a specification for making one.
+    AudioEncoder* enc =
+        rent_a_codec_.RentEncoder(*codec_manager_.GetCodecInst());
+    if (!enc)
+      return -1;
+    sp->speech_encoder = enc;
+  }
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  return 0;
 }
 
 void AudioCodingModuleImpl::RegisterExternalSendCodec(
     AudioEncoder* external_speech_encoder) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  codec_manager_.RegisterEncoder(external_speech_encoder);
+  auto* sp = codec_manager_.GetStackParams();
+  sp->speech_encoder = external_speech_encoder;
+  rent_a_codec_.RentEncoderStack(sp);
 }
 
 // Get current send codec.
-int AudioCodingModuleImpl::SendCodec(CodecInst* current_codec) const {
+rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.GetCodecInst(current_codec);
+  auto* ci = codec_manager_.GetCodecInst();
+  if (ci) {
+    return rtc::Optional<CodecInst>(*ci);
+  }
+  auto* enc = codec_manager_.GetStackParams()->speech_encoder;
+  if (enc) {
+    return rtc::Optional<CodecInst>(CodecManager::ForgeCodecInst(enc));
+  }
+  return rtc::Optional<CodecInst>();
 }
 
 // Get current send frequency.
@@ -217,19 +243,21 @@ int AudioCodingModuleImpl::SendFrequency() const {
                "SendFrequency()");
   CriticalSectionScoped lock(acm_crit_sect_.get());
 
-  if (!codec_manager_.CurrentEncoder()) {
+  const auto* enc = rent_a_codec_.GetEncoderStack();
+  if (!enc) {
     WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                  "SendFrequency Failed, no codec is registered");
     return -1;
   }
 
-  return codec_manager_.CurrentEncoder()->SampleRateHz();
+  return enc->SampleRateHz();
 }
 
 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  if (codec_manager_.CurrentEncoder()) {
-    codec_manager_.CurrentEncoder()->SetTargetBitrate(bitrate_bps);
+  auto* enc = rent_a_codec_.GetEncoderStack();
+  if (enc) {
+    enc->SetTargetBitrate(bitrate_bps);
   }
 }
 
@@ -296,10 +324,12 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
   }
 
   // Check whether we need an up-mix or down-mix?
-  bool remix = ptr_frame->num_channels_ !=
-               codec_manager_.CurrentEncoder()->NumChannels();
+  const size_t current_num_channels =
+      rent_a_codec_.GetEncoderStack()->NumChannels();
+  const bool same_num_channels =
+      ptr_frame->num_channels_ == current_num_channels;
 
-  if (remix) {
+  if (!same_num_channels) {
     if (ptr_frame->num_channels_ == 1) {
       if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
         return -1;
@@ -314,14 +344,13 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
   const int16_t* ptr_audio = ptr_frame->data_;
 
   // For pushing data to primary, point the |ptr_audio| to correct buffer.
-  if (codec_manager_.CurrentEncoder()->NumChannels() !=
-      ptr_frame->num_channels_)
+  if (!same_num_channels)
     ptr_audio = input_data->buffer;
 
   input_data->input_timestamp = ptr_frame->timestamp_;
   input_data->audio = ptr_audio;
   input_data->length_per_channel = ptr_frame->samples_per_channel_;
-  input_data->audio_channel = codec_manager_.CurrentEncoder()->NumChannels();
+  input_data->audio_channel = current_num_channels;
 
   return 0;
 }
@@ -333,13 +362,14 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
 // is required, |*ptr_out| points to |in_frame|.
 int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
                                                const AudioFrame** ptr_out) {
-  bool resample = (in_frame.sample_rate_hz_ !=
-                   codec_manager_.CurrentEncoder()->SampleRateHz());
+  const auto* enc = rent_a_codec_.GetEncoderStack();
+  const bool resample = in_frame.sample_rate_hz_ != enc->SampleRateHz();
 
   // This variable is true if primary codec and secondary codec (if exists)
   // are both mono and input is stereo.
-  bool down_mix = (in_frame.num_channels_ == 2) &&
-                  (codec_manager_.CurrentEncoder()->NumChannels() == 1);
+  // TODO(henrik.lundin): This condition should probably be
+  //   in_frame.num_channels_ > enc->NumChannels()
+  const bool down_mix = in_frame.num_channels_ == 2 && enc->NumChannels() == 1;
 
   if (!first_10ms_data_) {
     expected_in_ts_ = in_frame.timestamp_;
@@ -349,10 +379,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
     // TODO(turajs): Do we need a warning here.
     expected_codec_ts_ +=
         (in_frame.timestamp_ - expected_in_ts_) *
-        static_cast<uint32_t>(
-            (static_cast<double>(
-                 codec_manager_.CurrentEncoder()->SampleRateHz()) /
-             static_cast<double>(in_frame.sample_rate_hz_)));
+        static_cast<uint32_t>(static_cast<double>(enc->SampleRateHz()) /
+                              static_cast<double>(in_frame.sample_rate_hz_));
     expected_in_ts_ = in_frame.timestamp_;
   }
 
@@ -391,8 +419,7 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
     dest_ptr_audio = preprocess_frame_.data_;
 
     int samples_per_channel = resampler_.Resample10Msec(
-        src_ptr_audio, in_frame.sample_rate_hz_,
-        codec_manager_.CurrentEncoder()->SampleRateHz(),
+        src_ptr_audio, in_frame.sample_rate_hz_, enc->SampleRateHz(),
         preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
         dest_ptr_audio);
 
@@ -403,8 +430,7 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
     }
     preprocess_frame_.samples_per_channel_ =
         static_cast<size_t>(samples_per_channel);
-    preprocess_frame_.sample_rate_hz_ =
-        codec_manager_.CurrentEncoder()->SampleRateHz();
+    preprocess_frame_.sample_rate_hz_ = enc->SampleRateHz();
   }
 
   expected_codec_ts_ +=
@@ -420,17 +446,21 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
 
 bool AudioCodingModuleImpl::REDStatus() const {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.red_enabled();
+  return codec_manager_.GetStackParams()->use_red;
 }
 
 // Configure RED status i.e on/off.
-int AudioCodingModuleImpl::SetREDStatus(
+int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
 #ifdef WEBRTC_CODEC_RED
-    bool enable_red) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.SetCopyRed(enable_red) ? 0 : -1;
+  if (!codec_manager_.SetCopyRed(enable_red)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  return 0;
 #else
-    bool /* enable_red */) {
   WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
                "  WEBRTC_CODEC_RED is undefined");
   return -1;
@@ -443,18 +473,29 @@ int AudioCodingModuleImpl::SetREDStatus(
 
 bool AudioCodingModuleImpl::CodecFEC() const {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.codec_fec_enabled();
+  return codec_manager_.GetStackParams()->use_codec_fec;
 }
 
 int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.SetCodecFEC(enable_codec_fec);
+  if (!codec_manager_.SetCodecFEC(enable_codec_fec)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  if (enable_codec_fec) {
+    return sp->use_codec_fec ? 0 : -1;
+  } else {
+    RTC_DCHECK(!sp->use_codec_fec);
+    return 0;
+  }
 }
 
 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   if (HaveValidEncoder("SetPacketLossRate")) {
-    codec_manager_.CurrentEncoder()->SetProjectedPacketLossRate(loss_rate /
+    rent_a_codec_.GetEncoderStack()->SetProjectedPacketLossRate(loss_rate /
                                                                 100.0);
   }
   return 0;
@@ -469,14 +510,22 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
   // Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
   RTC_DCHECK_EQ(enable_dtx, enable_vad);
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.SetVAD(enable_dtx, mode);
+  if (!codec_manager_.SetVAD(enable_dtx, mode)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  return 0;
 }
 
 // Get VAD/DTX settings.
 int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
                                ACMVADMode* mode) const {
   CriticalSectionScoped lock(acm_crit_sect_.get());
-  codec_manager_.VAD(dtx_enabled, vad_enabled, mode);
+  const auto* sp = codec_manager_.GetStackParams();
+  *dtx_enabled = *vad_enabled = sp->use_cng;
+  *mode = sp->vad_mode;
   return 0;
 }
 
@@ -510,7 +559,7 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
     if (IsCodecRED(db[i]) || IsCodecCN(db[i])) {
       if (receiver_.AddCodec(static_cast<int>(i),
                              static_cast<uint8_t>(db[i].pltype), 1,
-                             db[i].plfreq, nullptr) < 0) {
+                             db[i].plfreq, nullptr, db[i].plname) < 0) {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                      "Cannot register master codec.");
         return -1;
@@ -523,25 +572,16 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
 
 // Get current receive frequency.
 int AudioCodingModuleImpl::ReceiveFrequency() const {
-  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
-               "ReceiveFrequency()");
-
-  CriticalSectionScoped lock(acm_crit_sect_.get());
-
-  int codec_id = receiver_.last_audio_codec_id();
-
-  return codec_id < 0 ? receiver_.current_sample_rate_hz() :
-                        ACMCodecDB::database_[codec_id].plfreq;
+  const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
+  return last_packet_sample_rate ? *last_packet_sample_rate
+                                 : receiver_.last_output_sample_rate_hz();
 }
 
 // Get current playout frequency.
 int AudioCodingModuleImpl::PlayoutFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutFrequency()");
-
-  CriticalSectionScoped lock(acm_crit_sect_.get());
-
-  return receiver_.current_sample_rate_hz();
+  return receiver_.last_output_sample_rate_hz();
 }
 
 // Register possible receive codecs, can be called multiple times,
@@ -549,7 +589,7 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
 int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   RTC_DCHECK(receiver_initialized_);
-  if (codec.channels > 2 || codec.channels < 0) {
+  if (codec.channels > 2) {
     LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
     return -1;
   }
@@ -564,7 +604,7 @@ int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
   RTC_CHECK(codec_index) << "Invalid codec ID: " << static_cast<int>(*codec_id);
 
   // Check if the payload-type is valid.
-  if (!ACMCodecDB::ValidPayloadType(codec.pltype)) {
+  if (!RentACodec::IsPayloadTypeValid(codec.pltype)) {
     LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
                     << codec.plname;
     return -1;
@@ -572,16 +612,19 @@ int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
 
   // Get |decoder| associated with |codec|. |decoder| is NULL if |codec| does
   // not own its decoder.
-  return receiver_.AddCodec(*codec_index, codec.pltype, codec.channels,
-                            codec.plfreq,
-                            codec_manager_.GetAudioDecoder(codec));
+  return receiver_.AddCodec(
+      *codec_index, codec.pltype, codec.channels, codec.plfreq,
+      STR_CASE_CMP(codec.plname, "isac") == 0 ? rent_a_codec_.RentIsacDecoder()
+                                              : nullptr,
+      codec.plname);
 }
 
 int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
     int rtp_payload_type,
     AudioDecoder* external_decoder,
     int sample_rate_hz,
-    int num_channels) {
+    int num_channels,
+    const std::string& name) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   RTC_DCHECK(receiver_initialized_);
   if (num_channels > 2 || num_channels < 0) {
@@ -590,14 +633,14 @@ int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
   }
 
   // Check if the payload-type is valid.
-  if (!ACMCodecDB::ValidPayloadType(rtp_payload_type)) {
+  if (!RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
     LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
                     << " for external decoder.";
     return -1;
   }
 
   return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
-                            sample_rate_hz, external_decoder);
+                            sample_rate_hz, external_decoder, name);
 }
 
 // Get current received codec.
@@ -610,7 +653,9 @@ int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
                                           const size_t payload_length,
                                           const WebRtcRTPHeader& rtp_header) {
-  return receiver_.InsertPacket(rtp_header, incoming_payload, payload_length);
+  return receiver_.InsertPacket(
+      rtp_header,
+      rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
 }
 
 // Minimum playout delay (Used for lip-sync).
@@ -699,8 +744,6 @@ int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
   if (!HaveValidEncoder("SetOpusApplication")) {
     return -1;
   }
-  if (!codec_manager_.CurrentEncoderIsOpus())
-    return -1;
   AudioEncoder::Application app;
   switch (application) {
     case kVoip:
@@ -713,7 +756,7 @@ int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
       FATAL();
       return 0;
   }
-  return codec_manager_.CurrentEncoder()->SetApplication(app) ? 0 : -1;
+  return rent_a_codec_.GetEncoderStack()->SetApplication(app) ? 0 : -1;
 }
 
 // Informs Opus encoder of the maximum playback rate the receiver will render.
@@ -722,9 +765,7 @@ int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
   if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
     return -1;
   }
-  if (!codec_manager_.CurrentEncoderIsOpus())
-    return -1;
-  codec_manager_.CurrentEncoder()->SetMaxPlaybackRate(frequency_hz);
+  rent_a_codec_.GetEncoderStack()->SetMaxPlaybackRate(frequency_hz);
   return 0;
 }
 
@@ -733,9 +774,7 @@ int AudioCodingModuleImpl::EnableOpusDtx() {
   if (!HaveValidEncoder("EnableOpusDtx")) {
     return -1;
   }
-  if (!codec_manager_.CurrentEncoderIsOpus())
-    return -1;
-  return codec_manager_.CurrentEncoder()->SetDtx(true) ? 0 : -1;
+  return rent_a_codec_.GetEncoderStack()->SetDtx(true) ? 0 : -1;
 }
 
 int AudioCodingModuleImpl::DisableOpusDtx() {
@@ -743,9 +782,7 @@ int AudioCodingModuleImpl::DisableOpusDtx() {
   if (!HaveValidEncoder("DisableOpusDtx")) {
     return -1;
   }
-  if (!codec_manager_.CurrentEncoderIsOpus())
-    return -1;
-  return codec_manager_.CurrentEncoder()->SetDtx(false) ? 0 : -1;
+  return rent_a_codec_.GetEncoderStack()->SetDtx(false) ? 0 : -1;
 }
 
 int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
@@ -753,7 +790,7 @@ int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
 }
 
 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
-  if (!codec_manager_.CurrentEncoder()) {
+  if (!rent_a_codec_.GetEncoderStack()) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "%s failed: No send codec is registered.", caller_name);
     return false;
@@ -765,17 +802,6 @@ int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
   return receiver_.RemoveCodec(payload_type);
 }
 
-int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
-  {
-    CriticalSectionScoped lock(acm_crit_sect_.get());
-    // Initialize receiver, if it is not initialized. Otherwise, initial delay
-    // is reset upon initialization of the receiver.
-    if (!receiver_initialized_)
-      InitializeReceiverSafe();
-  }
-  return receiver_.SetInitialDelay(delay_ms);
-}
-
 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
   return receiver_.EnableNack(max_nack_list_size);
 }
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index f20861398b..926671f199 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/base/buffer.h"
@@ -18,10 +19,9 @@
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 
 namespace webrtc {
 
@@ -48,7 +48,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
       AudioEncoder* external_speech_encoder) override;
 
   // Get current send codec.
-  int SendCodec(CodecInst* current_codec) const override;
+  rtc::Optional<CodecInst> SendCodec() const override;
 
   // Get current send frequency.
   int SendFrequency() const override;
@@ -124,7 +124,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   int RegisterExternalReceiveCodec(int rtp_payload_type,
                                    AudioDecoder* external_decoder,
                                    int sample_rate_hz,
-                                   int num_channels) override;
+                                   int num_channels,
+                                   const std::string& name) override;
 
   // Get current received codec.
   int ReceiveCodec(CodecInst* current_codec) const override;
@@ -150,10 +151,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // Smallest latency NetEq will maintain.
   int LeastRequiredDelayMs() const override;
 
-  // Impose an initial delay on playout. ACM plays silence until |delay_ms|
-  // audio is accumulated in NetEq buffer, then starts decoding payloads.
-  int SetInitialPlayoutDelay(int delay_ms) override;
-
   // Get playout timestamp.
   int PlayoutTimestamp(uint32_t* timestamp) override;
 
@@ -192,7 +189,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
     uint32_t input_timestamp;
     const int16_t* audio;
     size_t length_per_channel;
-    uint8_t audio_channel;
+    size_t audio_channel;
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
@@ -252,6 +249,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
   ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_);
   CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_);
+  RentACodec rent_a_codec_ GUARDED_BY(acm_crit_sect_);
 
   // This is to keep track of CN instances where we can send DTMFs.
   uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
@@ -282,4 +280,4 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
 }  // namespace acm2
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 879fb839fe..6f82a96ee5 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -8,22 +8,24 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <stdio.h>
 #include <string.h>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/md5digest.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
@@ -33,14 +35,12 @@
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using ::testing::AtLeast;
 using ::testing::Invoke;
@@ -237,7 +237,12 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
 
 // Check if the statistics are initialized correctly. Before any call to ACM
 // all fields have to be zero.
-TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_InitializedToZero DISABLED_InitializedToZero
+#else
+#define MAYBE_InitializedToZero InitializedToZero
+#endif
+TEST_F(AudioCodingModuleTestOldApi, MAYBE_InitializedToZero) {
   RegisterCodec();
   AudioDecodingCallStats stats;
   acm_->GetDecodingCallStatistics(&stats);
@@ -249,34 +254,15 @@ TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
   EXPECT_EQ(0, stats.decoded_plc_cng);
 }
 
-// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
-// should result in generating silence, check the associated field.
-TEST_F(AudioCodingModuleTestOldApi,
-       DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
-  RegisterCodec();
-  AudioDecodingCallStats stats;
-  const int kInitialDelay = 100;
-
-  acm_->SetInitialPlayoutDelay(kInitialDelay);
-
-  int num_calls = 0;
-  for (int time_ms = 0; time_ms < kInitialDelay;
-       time_ms += kFrameSizeMs, ++num_calls) {
-    InsertPacketAndPullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(0, stats.calls_to_neteq);
-  EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
-  EXPECT_EQ(0, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-}
-
 // Insert some packets and pull audio. Check statistics are valid. Then,
 // simulate packet loss and check if PLC and PLC-to-CNG statistics are
 // correctly updated.
-TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(NetEqCalls)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NetEqCalls DISABLED_NetEqCalls
+#else
+#define MAYBE_NetEqCalls NetEqCalls
+#endif
+TEST_F(AudioCodingModuleTestOldApi, MAYBE_NetEqCalls) {
   RegisterCodec();
   AudioDecodingCallStats stats;
   const int kNumNormalCalls = 10;
@@ -314,7 +300,7 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
   EXPECT_EQ(id_, audio_frame.id_);
   EXPECT_EQ(0u, audio_frame.timestamp_);
-  EXPECT_GT(audio_frame.num_channels_, 0);
+  EXPECT_GT(audio_frame.num_channels_, 0u);
   EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
             audio_frame.samples_per_channel_);
   EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
@@ -343,15 +329,9 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
 }
 
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
-#else
-#define IF_ISAC(x) DISABLED_##x
-#endif
-
 // Verifies that the RTP timestamp series is not reset when the codec is
 // changed.
-TEST_F(AudioCodingModuleTestOldApi,
-       IF_ISAC(TimestampSeriesContinuesWhenCodecChanges)) {
+TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
   RegisterCodec();  // This registers the default codec.
   uint32_t expected_ts = input_frame_.timestamp_;
   int blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
@@ -383,6 +363,7 @@ TEST_F(AudioCodingModuleTestOldApi,
     expected_ts += codec_.pacsize;
   }
 }
+#endif
 
 // Introduce this class to set different expectations on the number of encoded
 // bytes. This class expects all encoded packets to be 9 bytes (matching one
@@ -481,11 +462,9 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
 
   AudioCodingModuleMtTestOldApi()
       : AudioCodingModuleTestOldApi(),
-        send_thread_(ThreadWrapper::CreateThread(CbSendThread, this, "send")),
-        insert_packet_thread_(ThreadWrapper::CreateThread(
-            CbInsertPacketThread, this, "insert_packet")),
-        pull_audio_thread_(ThreadWrapper::CreateThread(
-            CbPullAudioThread, this, "pull_audio")),
+        send_thread_(CbSendThread, this, "send"),
+        insert_packet_thread_(CbInsertPacketThread, this, "insert_packet"),
+        pull_audio_thread_(CbPullAudioThread, this, "pull_audio"),
         test_complete_(EventWrapper::Create()),
         send_count_(0),
         insert_packet_count_(0),
@@ -503,19 +482,19 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   }
 
   void StartThreads() {
-    ASSERT_TRUE(send_thread_->Start());
-    send_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(insert_packet_thread_->Start());
-    insert_packet_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(pull_audio_thread_->Start());
-    pull_audio_thread_->SetPriority(kRealtimePriority);
+    send_thread_.Start();
+    send_thread_.SetPriority(rtc::kRealtimePriority);
+    insert_packet_thread_.Start();
+    insert_packet_thread_.SetPriority(rtc::kRealtimePriority);
+    pull_audio_thread_.Start();
+    pull_audio_thread_.SetPriority(rtc::kRealtimePriority);
   }
 
   void TearDown() {
     AudioCodingModuleTestOldApi::TearDown();
-    pull_audio_thread_->Stop();
-    send_thread_->Stop();
-    insert_packet_thread_->Stop();
+    pull_audio_thread_.Stop();
+    send_thread_.Stop();
+    insert_packet_thread_.Stop();
   }
 
   EventTypeWrapper RunTest() {
@@ -595,9 +574,9 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
     return true;
   }
 
-  rtc::scoped_ptr<ThreadWrapper> send_thread_;
-  rtc::scoped_ptr<ThreadWrapper> insert_packet_thread_;
-  rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
+  rtc::PlatformThread send_thread_;
+  rtc::PlatformThread insert_packet_thread_;
+  rtc::PlatformThread pull_audio_thread_;
   const rtc::scoped_ptr<EventWrapper> test_complete_;
   int send_count_;
   int insert_packet_count_;
@@ -607,7 +586,12 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
 };
 
-TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
 
@@ -680,7 +664,11 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   }
 
   void InsertAudio() {
-    memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
+    // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
+    // this call confuses the number of samples with the number of bytes, and
+    // ends up copying only half of what it should.
+    memcpy(input_frame_.data_, audio_loop_.GetNextBlock().data(),
+           kNumSamples10ms);
     AudioCodingModuleTestOldApi::InsertAudio();
   }
 
@@ -707,9 +695,16 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
+#endif
 
 class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
  protected:
@@ -720,12 +715,10 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
 
   AcmReRegisterIsacMtTestOldApi()
       : AudioCodingModuleTestOldApi(),
-        receive_thread_(
-            ThreadWrapper::CreateThread(CbReceiveThread, this, "receive")),
-        codec_registration_thread_(
-            ThreadWrapper::CreateThread(CbCodecRegistrationThread,
-                                        this,
-                                        "codec_registration")),
+        receive_thread_(CbReceiveThread, this, "receive"),
+        codec_registration_thread_(CbCodecRegistrationThread,
+                                   this,
+                                   "codec_registration"),
         test_complete_(EventWrapper::Create()),
         crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
         codec_registered_(false),
@@ -761,16 +754,16 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
   }
 
   void StartThreads() {
-    ASSERT_TRUE(receive_thread_->Start());
-    receive_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(codec_registration_thread_->Start());
-    codec_registration_thread_->SetPriority(kRealtimePriority);
+    receive_thread_.Start();
+    receive_thread_.SetPriority(rtc::kRealtimePriority);
+    codec_registration_thread_.Start();
+    codec_registration_thread_.SetPriority(rtc::kRealtimePriority);
   }
 
   void TearDown() {
     AudioCodingModuleTestOldApi::TearDown();
-    receive_thread_->Stop();
-    codec_registration_thread_->Stop();
+    receive_thread_.Stop();
+    codec_registration_thread_.Stop();
   }
 
   EventTypeWrapper RunTest() {
@@ -798,9 +791,9 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
       // Encode new frame.
       uint32_t input_timestamp = rtp_header_.header.timestamp;
       while (info.encoded_bytes == 0) {
-        info = isac_encoder_->Encode(
-            input_timestamp, audio_loop_.GetNextBlock(), kNumSamples10ms,
-            max_encoded_bytes, encoded.get());
+        info =
+            isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),
+                                  max_encoded_bytes, encoded.get());
         input_timestamp += 160;  // 10 ms at 16 kHz.
       }
       EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
@@ -849,8 +842,8 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
     return true;
   }
 
-  rtc::scoped_ptr<ThreadWrapper> receive_thread_;
-  rtc::scoped_ptr<ThreadWrapper> codec_registration_thread_;
+  rtc::PlatformThread receive_thread_;
+  rtc::PlatformThread codec_registration_thread_;
   const rtc::scoped_ptr<EventWrapper> test_complete_;
   const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   bool codec_registered_ GUARDED_BY(crit_sect_);
@@ -861,9 +854,16 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
+#endif
 
 // Disabling all of these tests on iOS until file support has been added.
 // See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
@@ -871,13 +871,16 @@ TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
 
 class AcmReceiverBitExactnessOldApi : public ::testing::Test {
  public:
-  static std::string PlatformChecksum(std::string win64,
-                                      std::string android,
-                                      std::string others) {
+  static std::string PlatformChecksum(std::string others,
+                                      std::string win64,
+                                      std::string android_arm32,
+                                      std::string android_arm64) {
 #if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
     return win64;
-#elif defined(WEBRTC_ANDROID)
-    return android;
+#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
+    return android_arm32;
+#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
+    return android_arm64;
 #else
     return others;
 #endif
@@ -889,6 +892,7 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
     AudioDecoder* external_decoder;
     int sample_rate_hz;
     int num_channels;
+    std::string name;
   };
 
   void Run(int output_freq_hz,
@@ -924,83 +928,76 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
     for (const auto& ed : external_decoders) {
       ASSERT_EQ(0, test.RegisterExternalReceiveCodec(
                        ed.rtp_payload_type, ed.external_decoder,
-                       ed.sample_rate_hz, ed.num_channels));
+                       ed.sample_rate_hz, ed.num_channels, ed.name));
     }
     test.Run();
 
     std::string checksum_string = checksum.Finish();
     EXPECT_EQ(checksum_ref, checksum_string);
+
+    // Delete the output file.
+    remove(output_file_name.c_str());
   }
 };
 
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISAC)) && \
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
-#define IF_ALL_CODECS(x) x
-#else
-#define IF_ALL_CODECS(x) DISABLED_##x
-#endif
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_8kHzOutput DISABLED_8kHzOutput
-#else
-#define MAYBE_8kHzOutput 8kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_8kHzOutput)) {
-  Run(8000, PlatformChecksum("dcee98c623b147ebe1b40dd30efa896e",
+TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
+  Run(8000, PlatformChecksum("908002dc01fc4eb1d2be24eb1d3f354b",
+                             "dcee98c623b147ebe1b40dd30efa896e",
                              "adc92e173f908f93b96ba5844209815a",
-                             "908002dc01fc4eb1d2be24eb1d3f354b"),
+                             "ba16137d3a5a1e637252289c57522bfe"),
       std::vector<ExternalDecoder>());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_16kHzOutput DISABLED_16kHzOutput
-#else
-#define MAYBE_16kHzOutput 16kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_16kHzOutput)) {
-  Run(16000, PlatformChecksum("f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
+TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
+  Run(16000, PlatformChecksum("a909560b5ca49fa472b17b7b277195e9",
+                              "f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
                               "8cffa6abcb3e18e33b9d857666dff66a",
-                              "a909560b5ca49fa472b17b7b277195e9"),
+                              "66ee001e23534d4dcf5d0f81f916c93b"),
       std::vector<ExternalDecoder>());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_32kHzOutput DISABLED_32kHzOutput
-#else
-#define MAYBE_32kHzOutput 32kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_32kHzOutput)) {
-  Run(32000, PlatformChecksum("306e0d990ee6e92de3fbecc0123ece37",
+TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
+  Run(32000, PlatformChecksum("441aab4b347fb3db4e9244337aca8d8e",
+                              "306e0d990ee6e92de3fbecc0123ece37",
                               "3e126fe894720c3f85edadcc91964ba5",
-                              "441aab4b347fb3db4e9244337aca8d8e"),
+                              "9c6ff204b14152c48fe41d5ab757943b"),
       std::vector<ExternalDecoder>());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_48kHzOutput DISABLED_48kHzOutput
-#else
-#define MAYBE_48kHzOutput 48kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_48kHzOutput)) {
-  Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
+TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
+  Run(48000, PlatformChecksum("4ee2730fa1daae755e8a8fd3abd779ec",
+                              "aa7c232f63a67b2a72703593bdd172e0",
                               "0155665e93067c4e89256b944dd11999",
-                              "4ee2730fa1daae755e8a8fd3abd779ec"),
+                              "fc4f0da8844cd808d822bbddf3b9c285"),
       std::vector<ExternalDecoder>());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_48kHzOutputExternalDecoder DISABLED_48kHzOutputExternalDecoder
-#else
-#define MAYBE_48kHzOutputExternalDecoder 48kHzOutputExternalDecoder
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi,
-       IF_ALL_CODECS(MAYBE_48kHzOutputExternalDecoder)) {
+TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
+  // Class intended to forward a call from a mock DecodeInternal to Decode on
+  // the real decoder's Decode. DecodeInternal for the real decoder isn't
+  // public.
+  class DecodeForwarder {
+   public:
+    DecodeForwarder(AudioDecoder* decoder) : decoder_(decoder) {}
+    int Decode(const uint8_t* encoded,
+               size_t encoded_len,
+               int sample_rate_hz,
+               int16_t* decoded,
+               AudioDecoder::SpeechType* speech_type) {
+      return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
+                              decoder_->PacketDuration(encoded, encoded_len) *
+                                  decoder_->Channels() * sizeof(int16_t),
+                              decoded, speech_type);
+    }
+
+   private:
+    AudioDecoder* const decoder_;
+  };
+
   AudioDecoderPcmU decoder(1);
+  DecodeForwarder decode_forwarder(&decoder);
   MockAudioDecoder mock_decoder;
   // Set expectations on the mock decoder and also delegate the calls to the
   // real decoder.
@@ -1010,9 +1007,9 @@ TEST_F(AcmReceiverBitExactnessOldApi,
   EXPECT_CALL(mock_decoder, Channels())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Channels));
-  EXPECT_CALL(mock_decoder, Decode(_, _, _, _, _, _))
+  EXPECT_CALL(mock_decoder, DecodeInternal(_, _, _, _, _))
       .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Decode));
+      .WillRepeatedly(Invoke(&decode_forwarder, &DecodeForwarder::Decode));
   EXPECT_CALL(mock_decoder, HasDecodePlc())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc));
@@ -1024,16 +1021,19 @@ TEST_F(AcmReceiverBitExactnessOldApi,
   ed.external_decoder = &mock_decoder;
   ed.sample_rate_hz = 8000;
   ed.num_channels = 1;
+  ed.name = "MockPCMU";
   std::vector<ExternalDecoder> external_decoders;
   external_decoders.push_back(ed);
 
-  Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
+  Run(48000, PlatformChecksum("4ee2730fa1daae755e8a8fd3abd779ec",
+                              "aa7c232f63a67b2a72703593bdd172e0",
                               "0155665e93067c4e89256b944dd11999",
-                              "4ee2730fa1daae755e8a8fd3abd779ec"),
+                              "fc4f0da8844cd808d822bbddf3b9c285"),
       external_decoders);
 
   EXPECT_CALL(mock_decoder, Die());
 }
+#endif
 
 // This test verifies bit exactness for the send-side of ACM. The test setup is
 // a chain of three different test classes:
@@ -1138,6 +1138,9 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
 
     // Verify number of packets produced.
     EXPECT_EQ(expected_packets, packet_count_);
+
+    // Delete the output file.
+    remove(output_file_name.c_str());
   }
 
   // Returns a pointer to the next packet. Returns NULL if the source is
@@ -1209,65 +1212,57 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
   rtc::Md5Digest payload_checksum_;
 };
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_IsacWb30ms DISABLED_IsacWb30ms
-#else
-#define MAYBE_IsacWb30ms IsacWb30ms
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(MAYBE_IsacWb30ms)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "0b58f9eeee43d5891f5f6c75e77984a3",
           "c7e5bdadfa2871df95639fcc297cf23d",
           "0499ca260390769b3172136faad925b9",
-          "0b58f9eeee43d5891f5f6c75e77984a3"),
+          "866abf524acd2807efbe65e133c23f95"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "3c79f16f34218271f3dca4e2b1dfe1bb",
           "d42cb5195463da26c8129bbfe73a22e6",
           "83de248aea9c3c2bd680b6952401b4ca",
           "3c79f16f34218271f3dca4e2b1dfe1bb"),
-      33,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+      33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_IsacWb60ms DISABLED_IsacWb60ms
-#else
-#define MAYBE_IsacWb60ms IsacWb60ms
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(MAYBE_IsacWb60ms)) {
+TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "1ad29139a04782a33daad8c2b9b35875",
           "14d63c5f08127d280e722e3191b73bdd",
           "8da003e16c5371af2dc2be79a50f9076",
-          "1ad29139a04782a33daad8c2b9b35875"),
+          "ef75e900e6f375e3061163c53fd09a63"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "9e0a0ab743ad987b55b8e14802769c56",
           "ebe04a819d3a9d83a83a17f271e1139a",
           "97aeef98553b5a4b5a68f8b716e8eaf0",
           "9e0a0ab743ad987b55b8e14802769c56"),
-      16,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+      16, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-#ifdef WEBRTC_CODEC_ISAC
-#define IF_ISAC_FLOAT(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_IsacSwb30ms DISABLED_IsacSwb30ms
 #else
-#define IF_ISAC_FLOAT(x) DISABLED_##x
+#define MAYBE_IsacSwb30ms IsacSwb30ms
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi,
-       DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
+#if defined(WEBRTC_CODEC_ISAC)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacSwb30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
-          "2b3c387d06f00b7b7aad4c9be56fb83d",
-          "",
-          "5683b58da0fbf2063c7adc2e6bfb3fb8"),
+          "5683b58da0fbf2063c7adc2e6bfb3fb8",
+          "2b3c387d06f00b7b7aad4c9be56fb83d", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
-          "bcc2041e7744c7ebd9f701866856849c",
-          "",
-          "ce86106a93419aefb063097108ec94ab"),
+          "ce86106a93419aefb063097108ec94ab",
+          "bcc2041e7744c7ebd9f701866856849c", "android_arm32_payload",
+          "android_arm64_payload"),
       33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
 TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
@@ -1349,101 +1344,96 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
       test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-#ifdef WEBRTC_CODEC_ILBC
-#define IF_ILBC(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_Ilbc_30ms DISABLED_Ilbc_30ms
 #else
-#define IF_ILBC(x) DISABLED_##x
+#define MAYBE_Ilbc_30ms Ilbc_30ms
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
+#if defined(WEBRTC_CODEC_ILBC)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7b6ec10910debd9af08011d3ed5249f7",
-          "android_audio",
-          "7b6ec10910debd9af08011d3ed5249f7"),
+          "7b6ec10910debd9af08011d3ed5249f7", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "cfae2e9f6aba96e145f2bcdd5050ce78",
-          "android_payload",
-          "cfae2e9f6aba96e145f2bcdd5050ce78"),
-      33,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+          "cfae2e9f6aba96e145f2bcdd5050ce78", "android_arm32_payload",
+          "android_arm64_payload"),
+      33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-#ifdef WEBRTC_CODEC_G722
-#define IF_G722(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_G722_20ms DISABLED_G722_20ms
 #else
-#define IF_G722(x) DISABLED_##x
+#define MAYBE_G722_20ms G722_20ms
 #endif
-
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
+#if defined(WEBRTC_CODEC_G722)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7d759436f2533582950d148b5161a36c",
-          "android_audio",
-          "7d759436f2533582950d148b5161a36c"),
+          "7d759436f2533582950d148b5161a36c", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "fc68a87e1380614e658087cb35d5ca10",
-          "android_payload",
-          "fc68a87e1380614e658087cb35d5ca10"),
-      50,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+          "fc68a87e1380614e658087cb35d5ca10", "android_arm32_payload",
+          "android_arm64_payload"),
+      50, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-TEST_F(AcmSenderBitExactnessOldApi,
-       DISABLED_ON_ANDROID(IF_G722(G722_stereo_20ms))) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms
+#else
+#define MAYBE_G722_stereo_20ms G722_stereo_20ms
+#endif
+#if defined(WEBRTC_CODEC_G722)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7190ee718ab3d80eca181e5f7140c210",
-          "android_audio",
-          "7190ee718ab3d80eca181e5f7140c210"),
+          "7190ee718ab3d80eca181e5f7140c210", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "66516152eeaa1e650ad94ff85f668dac",
-          "android_payload",
-          "66516152eeaa1e650ad94ff85f668dac"),
-      50,
-      test::AcmReceiveTestOldApi::kStereoOutput);
+          "66516152eeaa1e650ad94ff85f668dac", "android_arm32_payload",
+          "android_arm64_payload"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
-#else
-#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
 #endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
+
+TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "855041f2490b887302bce9d544731849",
+          "855041f2490b887302bce9d544731849",
           "1e1a0fce893fef2d66886a7f09e2ebce",
-          "855041f2490b887302bce9d544731849"),
+          "7417a66c28be42d5d9b2d64e0c191585"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "d781cce1ab986b618d0da87226cdde30",
+          "d781cce1ab986b618d0da87226cdde30",
           "1a1fe04dd12e755949987c8d729fb3e0",
-          "d781cce1ab986b618d0da87226cdde30"),
-      50,
-      test::AcmReceiveTestOldApi::kStereoOutput);
+          "47b0b04f1d03076b857c86c72c2c298b"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_Opus_stereo_20ms_voip DISABLED_Opus_stereo_20ms_voip
-#else
-#define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
+TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
   // If not set, default will be kAudio in case of stereo.
   EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "9b9e12bc3cc793740966e11cbfa8b35b",
+          "9b9e12bc3cc793740966e11cbfa8b35b",
           "57412a4b5771d19ff03ec35deffe7067",
-          "9b9e12bc3cc793740966e11cbfa8b35b"),
+          "7ad0bbefcaa87e23187bf4a56d2f3513"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "c7340b1189652ab6b5e80dade7390cb4",
+          "c7340b1189652ab6b5e80dade7390cb4",
           "cdfe85939c411d12b61701c566e22d26",
-          "c7340b1189652ab6b5e80dade7390cb4"),
-      50,
-      test::AcmReceiveTestOldApi::kStereoOutput);
+          "7a678fbe46df5bf0c67e88264a2d9275"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
 // This test is for verifying the SetBitRate function. The bitrate is changed at
@@ -1528,7 +1518,12 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
 
 // The result on the Android platforms is inconsistent for this test case.
 // On android_rel the result is different from android and android arm64 rel.
-TEST_F(AcmSetBitRateOldApi, DISABLED_ON_ANDROID(Opus_48khz_20ms_100kbps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
+#else
+#define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
+#endif
+TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
   Run(100000, 100888);
 }
@@ -1657,16 +1652,15 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
       .Times(AtLeast(1))
       .WillRepeatedly(
           Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket));
-  EXPECT_CALL(mock_encoder, Max10MsFramesInAPacket())
-      .Times(AtLeast(1))
-      .WillRepeatedly(
-          Invoke(&encoder, &AudioEncoderPcmU::Max10MsFramesInAPacket));
   EXPECT_CALL(mock_encoder, GetTargetBitrate())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
   EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _, _))
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::EncodeInternal));
+  EXPECT_CALL(mock_encoder, SetFec(_))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SetFec));
   ASSERT_NO_FATAL_FAILURE(
       SetUpTestExternalEncoder(&mock_encoder, codec_inst.pltype));
   Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
@@ -1724,6 +1718,9 @@ class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test,
 
     // This is where the actual test is executed.
     receive_test.Run();
+
+    // Delete output file.
+    remove(output_file_name.c_str());
   }
 
   // Inherited from test::PacketSource.
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.cc b/webrtc/modules/audio_coding/acm2/call_statistics.cc
index 4c3e9fc393..4441932c8c 100644
--- a/webrtc/modules/audio_coding/main/acm2/call_statistics.cc
+++ b/webrtc/modules/audio_coding/acm2/call_statistics.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 
 #include <assert.h>
 
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.h b/webrtc/modules/audio_coding/acm2/call_statistics.h
index 2aece0ff40..888afea0a7 100644
--- a/webrtc/modules/audio_coding/main/acm2/call_statistics.h
+++ b/webrtc/modules/audio_coding/acm2/call_statistics.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 //
 // This class is for book keeping of calls to ACM. It is not useful to log API
@@ -60,4 +60,4 @@ class CallStatistics {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc b/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
index 2bee96465d..9ba0774ce1 100644
--- a/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
@@ -9,7 +9,7 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.cc b/webrtc/modules/audio_coding/acm2/codec_manager.cc
new file mode 100644
index 0000000000..ad67377d42
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.cc
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+namespace acm2 {
+
+namespace {
+
+// Check if the given codec is a valid to be registered as send codec.
+int IsValidSendCodec(const CodecInst& send_codec) {
+  int dummy_id = 0;
+  if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "Wrong number of channels (%" PRIuS ", only mono and stereo "
+                 "are supported)",
+                 send_codec.channels);
+    return -1;
+  }
+
+  auto maybe_codec_id = RentACodec::CodecIdByInst(send_codec);
+  if (!maybe_codec_id) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "Invalid codec setting for the send codec.");
+    return -1;
+  }
+
+  // Telephone-event cannot be a send codec.
+  if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "telephone-event cannot be a send codec");
+    return -1;
+  }
+
+  if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels)
+           .value_or(false)) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "%" PRIuS " number of channels not supportedn for %s.",
+                 send_codec.channels, send_codec.plname);
+    return -1;
+  }
+  return RentACodec::CodecIndexFromId(*maybe_codec_id).value_or(-1);
+}
+
+bool IsOpus(const CodecInst& codec) {
+  return
+#ifdef WEBRTC_CODEC_OPUS
+      !STR_CASE_CMP(codec.plname, "opus") ||
+#endif
+      false;
+}
+
+}  // namespace
+
+CodecManager::CodecManager() {
+  thread_checker_.DetachFromThread();
+}
+
+CodecManager::~CodecManager() = default;
+
+bool CodecManager::RegisterEncoder(const CodecInst& send_codec) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int codec_id = IsValidSendCodec(send_codec);
+
+  // Check for reported errors from function IsValidSendCodec().
+  if (codec_id < 0) {
+    return false;
+  }
+
+  int dummy_id = 0;
+  switch (RentACodec::RegisterRedPayloadType(
+      &codec_stack_params_.red_payload_types, send_codec)) {
+    case RentACodec::RegistrationResult::kOk:
+      return true;
+    case RentACodec::RegistrationResult::kBadFreq:
+      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                   "RegisterSendCodec() failed, invalid frequency for RED"
+                   " registration");
+      return false;
+    case RentACodec::RegistrationResult::kSkip:
+      break;
+  }
+  switch (RentACodec::RegisterCngPayloadType(
+      &codec_stack_params_.cng_payload_types, send_codec)) {
+    case RentACodec::RegistrationResult::kOk:
+      return true;
+    case RentACodec::RegistrationResult::kBadFreq:
+      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                   "RegisterSendCodec() failed, invalid frequency for CNG"
+                   " registration");
+      return false;
+    case RentACodec::RegistrationResult::kSkip:
+      break;
+  }
+
+  if (IsOpus(send_codec)) {
+    // VAD/DTX not supported.
+    codec_stack_params_.use_cng = false;
+  }
+
+  send_codec_inst_ = rtc::Optional<CodecInst>(send_codec);
+  codec_stack_params_.speech_encoder = nullptr;  // Caller must recreate it.
+  return true;
+}
+
+CodecInst CodecManager::ForgeCodecInst(
+    const AudioEncoder* external_speech_encoder) {
+  CodecInst ci;
+  ci.channels = external_speech_encoder->NumChannels();
+  ci.plfreq = external_speech_encoder->SampleRateHz();
+  ci.pacsize = rtc::CheckedDivExact(
+      static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
+                       ci.plfreq),
+      100);
+  ci.pltype = -1;  // Not valid.
+  ci.rate = -1;    // Not valid.
+  static const char kName[] = "external";
+  memcpy(ci.plname, kName, sizeof(kName));
+  return ci;
+}
+
+bool CodecManager::SetCopyRed(bool enable) {
+  if (enable && codec_stack_params_.use_codec_fec) {
+    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
+                 "Codec internal FEC and RED cannot be co-enabled.");
+    return false;
+  }
+  if (enable && send_codec_inst_ &&
+      codec_stack_params_.red_payload_types.count(send_codec_inst_->plfreq) <
+          1) {
+    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
+                 "Cannot enable RED at %i Hz.", send_codec_inst_->plfreq);
+    return false;
+  }
+  codec_stack_params_.use_red = enable;
+  return true;
+}
+
+bool CodecManager::SetVAD(bool enable, ACMVADMode mode) {
+  // Sanity check of the mode.
+  RTC_DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
+             mode == VADVeryAggr);
+
+  // Check that the send codec is mono. We don't support VAD/DTX for stereo
+  // sending.
+  const bool stereo_send =
+      codec_stack_params_.speech_encoder
+          ? (codec_stack_params_.speech_encoder->NumChannels() != 1)
+          : false;
+  if (enable && stereo_send) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
+                 "VAD/DTX not supported for stereo sending");
+    return false;
+  }
+
+  // TODO(kwiberg): This doesn't protect Opus when injected as an external
+  // encoder.
+  if (send_codec_inst_ && IsOpus(*send_codec_inst_)) {
+    // VAD/DTX not supported, but don't fail.
+    enable = false;
+  }
+
+  codec_stack_params_.use_cng = enable;
+  codec_stack_params_.vad_mode = mode;
+  return true;
+}
+
+bool CodecManager::SetCodecFEC(bool enable_codec_fec) {
+  if (enable_codec_fec && codec_stack_params_.use_red) {
+    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
+                 "Codec internal FEC and RED cannot be co-enabled.");
+    return false;
+  }
+
+  codec_stack_params_.use_codec_fec = enable_codec_fec;
+  return true;
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.h b/webrtc/modules/audio_coding/acm2/codec_manager.h
new file mode 100644
index 0000000000..9227e13f09
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+
+#include <map>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+class AudioDecoder;
+class AudioEncoder;
+
+namespace acm2 {
+
+class CodecManager final {
+ public:
+  CodecManager();
+  ~CodecManager();
+
+  // Parses the given specification. On success, returns true and updates the
+  // stored CodecInst and stack parameters; on error, returns false.
+  bool RegisterEncoder(const CodecInst& send_codec);
+
+  static CodecInst ForgeCodecInst(const AudioEncoder* external_speech_encoder);
+
+  const CodecInst* GetCodecInst() const {
+    return send_codec_inst_ ? &*send_codec_inst_ : nullptr;
+  }
+  const RentACodec::StackParameters* GetStackParams() const {
+    return &codec_stack_params_;
+  }
+  RentACodec::StackParameters* GetStackParams() { return &codec_stack_params_; }
+
+  bool SetCopyRed(bool enable);
+
+  bool SetVAD(bool enable, ACMVADMode mode);
+
+  bool SetCodecFEC(bool enable_codec_fec);
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  rtc::Optional<CodecInst> send_codec_inst_;
+  RentACodec::StackParameters codec_stack_params_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(CodecManager);
+};
+
+}  // namespace acm2
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc b/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
new file mode 100644
index 0000000000..dce8f38842
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+
+namespace webrtc {
+namespace acm2 {
+
+using ::testing::Return;
+
+namespace {
+
+// Create a MockAudioEncoder with some reasonable default behavior.
+rtc::scoped_ptr<MockAudioEncoder> CreateMockEncoder() {
+  auto enc = rtc_make_scoped_ptr(new MockAudioEncoder);
+  EXPECT_CALL(*enc, SampleRateHz()).WillRepeatedly(Return(8000));
+  EXPECT_CALL(*enc, NumChannels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(*enc, Max10MsFramesInAPacket()).WillRepeatedly(Return(1));
+  EXPECT_CALL(*enc, Die());
+  return enc;
+}
+
+}  // namespace
+
+TEST(CodecManagerTest, ExternalEncoderFec) {
+  auto enc0 = CreateMockEncoder();
+  auto enc1 = CreateMockEncoder();
+  {
+    ::testing::InSequence s;
+    EXPECT_CALL(*enc0, SetFec(false)).WillOnce(Return(true));
+    EXPECT_CALL(*enc0, Mark("A"));
+    EXPECT_CALL(*enc0, SetFec(true)).WillOnce(Return(true));
+    EXPECT_CALL(*enc1, SetFec(true)).WillOnce(Return(true));
+    EXPECT_CALL(*enc1, SetFec(false)).WillOnce(Return(true));
+    EXPECT_CALL(*enc0, Mark("B"));
+    EXPECT_CALL(*enc0, SetFec(false)).WillOnce(Return(true));
+  }
+
+  CodecManager cm;
+  RentACodec rac;
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+  cm.GetStackParams()->speech_encoder = enc0.get();
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+  enc0->Mark("A");
+  EXPECT_EQ(true, cm.SetCodecFEC(true));
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_TRUE(cm.GetStackParams()->use_codec_fec);
+  cm.GetStackParams()->speech_encoder = enc1.get();
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_TRUE(cm.GetStackParams()->use_codec_fec);
+
+  EXPECT_EQ(true, cm.SetCodecFEC(false));
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  enc0->Mark("B");
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+  cm.GetStackParams()->speech_encoder = enc0.get();
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
index 786fb2e527..0c31b83eb3 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
+++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
index c6942ec285..32dd1260f1 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
+++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -117,4 +117,4 @@ class InitialDelayManager {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
index e973593eb4..d86d221851 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
@@ -11,7 +11,7 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
new file mode 100644
index 0000000000..5695fd6e08
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
@@ -0,0 +1,307 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+
+#include <utility>
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#ifdef WEBRTC_CODEC_G722
+#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#endif
+#ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#endif
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#ifdef WEBRTC_CODEC_RED
+#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#endif
+#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+
+namespace webrtc {
+namespace acm2 {
+
+rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
+    const char* payload_name,
+    int sampling_freq_hz,
+    size_t channels) {
+  return CodecIdFromIndex(
+      ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
+}
+
+rtc::Optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
+  rtc::Optional<int> mi = CodecIndexFromId(codec_id);
+  return mi ? rtc::Optional<CodecInst>(Database()[*mi])
+            : rtc::Optional<CodecInst>();
+}
+
+rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
+    const CodecInst& codec_inst) {
+  return CodecIdFromIndex(ACMCodecDB::CodecNumber(codec_inst));
+}
+
+rtc::Optional<CodecInst> RentACodec::CodecInstByParams(const char* payload_name,
+                                                       int sampling_freq_hz,
+                                                       size_t channels) {
+  rtc::Optional<CodecId> codec_id =
+      CodecIdByParams(payload_name, sampling_freq_hz, channels);
+  if (!codec_id)
+    return rtc::Optional<CodecInst>();
+  rtc::Optional<CodecInst> ci = CodecInstById(*codec_id);
+  RTC_DCHECK(ci);
+
+  // Keep the number of channels from the function call. For most codecs it
+  // will be the same value as in default codec settings, but not for all.
+  ci->channels = channels;
+
+  return ci;
+}
+
+bool RentACodec::IsCodecValid(const CodecInst& codec_inst) {
+  return ACMCodecDB::CodecNumber(codec_inst) >= 0;
+}
+
+rtc::Optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
+                                                       size_t num_channels) {
+  auto i = CodecIndexFromId(codec_id);
+  return i ? rtc::Optional<bool>(
+                 ACMCodecDB::codec_settings_[*i].channel_support >=
+                 num_channels)
+           : rtc::Optional<bool>();
+}
+
+rtc::ArrayView<const CodecInst> RentACodec::Database() {
+  return rtc::ArrayView<const CodecInst>(ACMCodecDB::database_,
+                                         NumberOfCodecs());
+}
+
+rtc::Optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
+    CodecId codec_id,
+    size_t num_channels) {
+  rtc::Optional<int> i = CodecIndexFromId(codec_id);
+  if (!i)
+    return rtc::Optional<NetEqDecoder>();
+  const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
+  return rtc::Optional<NetEqDecoder>(
+      (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
+          ? NetEqDecoder::kDecoderOpus_2ch
+          : ned);
+}
+
+RentACodec::RegistrationResult RentACodec::RegisterCngPayloadType(
+    std::map<int, int>* pt_map,
+    const CodecInst& codec_inst) {
+  if (STR_CASE_CMP(codec_inst.plname, "CN") != 0)
+    return RegistrationResult::kSkip;
+  switch (codec_inst.plfreq) {
+    case 8000:
+    case 16000:
+    case 32000:
+    case 48000:
+      (*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
+      return RegistrationResult::kOk;
+    default:
+      return RegistrationResult::kBadFreq;
+  }
+}
+
+RentACodec::RegistrationResult RentACodec::RegisterRedPayloadType(
+    std::map<int, int>* pt_map,
+    const CodecInst& codec_inst) {
+  if (STR_CASE_CMP(codec_inst.plname, "RED") != 0)
+    return RegistrationResult::kSkip;
+  switch (codec_inst.plfreq) {
+    case 8000:
+      (*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
+      return RegistrationResult::kOk;
+    default:
+      return RegistrationResult::kBadFreq;
+  }
+}
+
+namespace {
+
+// Returns a new speech encoder, or null on error.
+// TODO(kwiberg): Don't handle errors here (bug 5033)
+rtc::scoped_ptr<AudioEncoder> CreateEncoder(
+    const CodecInst& speech_inst,
+    LockedIsacBandwidthInfo* bwinfo) {
+#if defined(WEBRTC_CODEC_ISACFX)
+  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderIsacFix(speech_inst, bwinfo));
+#endif
+#if defined(WEBRTC_CODEC_ISAC)
+  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderIsac(speech_inst, bwinfo));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderOpus(speech_inst));
+#endif
+  if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderPcmU(speech_inst));
+  if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderPcmA(speech_inst));
+  if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderPcm16B(speech_inst));
+#ifdef WEBRTC_CODEC_ILBC
+  if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderIlbc(speech_inst));
+#endif
+#ifdef WEBRTC_CODEC_G722
+  if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderG722(speech_inst));
+#endif
+  LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
+  return rtc::scoped_ptr<AudioEncoder>();
+}
+
+rtc::scoped_ptr<AudioEncoder> CreateRedEncoder(AudioEncoder* encoder,
+                                               int red_payload_type) {
+#ifdef WEBRTC_CODEC_RED
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type;
+  config.speech_encoder = encoder;
+  return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCopyRed(config));
+#else
+  return rtc::scoped_ptr<AudioEncoder>();
+#endif
+}
+
+rtc::scoped_ptr<AudioEncoder> CreateCngEncoder(AudioEncoder* encoder,
+                                               int payload_type,
+                                               ACMVADMode vad_mode) {
+  AudioEncoderCng::Config config;
+  config.num_channels = encoder->NumChannels();
+  config.payload_type = payload_type;
+  config.speech_encoder = encoder;
+  switch (vad_mode) {
+    case VADNormal:
+      config.vad_mode = Vad::kVadNormal;
+      break;
+    case VADLowBitrate:
+      config.vad_mode = Vad::kVadLowBitrate;
+      break;
+    case VADAggr:
+      config.vad_mode = Vad::kVadAggressive;
+      break;
+    case VADVeryAggr:
+      config.vad_mode = Vad::kVadVeryAggressive;
+      break;
+    default:
+      FATAL();
+  }
+  return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCng(config));
+}
+
+rtc::scoped_ptr<AudioDecoder> CreateIsacDecoder(
+    LockedIsacBandwidthInfo* bwinfo) {
+#if defined(WEBRTC_CODEC_ISACFX)
+  return rtc_make_scoped_ptr(new AudioDecoderIsacFix(bwinfo));
+#elif defined(WEBRTC_CODEC_ISAC)
+  return rtc_make_scoped_ptr(new AudioDecoderIsac(bwinfo));
+#else
+  FATAL() << "iSAC is not supported.";
+  return rtc::scoped_ptr<AudioDecoder>();
+#endif
+}
+
+}  // namespace
+
+RentACodec::RentACodec() = default;
+RentACodec::~RentACodec() = default;
+
+AudioEncoder* RentACodec::RentEncoder(const CodecInst& codec_inst) {
+  rtc::scoped_ptr<AudioEncoder> enc =
+      CreateEncoder(codec_inst, &isac_bandwidth_info_);
+  if (!enc)
+    return nullptr;
+  speech_encoder_ = std::move(enc);
+  return speech_encoder_.get();
+}
+
+RentACodec::StackParameters::StackParameters() {
+  // Register the default payload types for RED and CNG.
+  for (const CodecInst& ci : RentACodec::Database()) {
+    RentACodec::RegisterCngPayloadType(&cng_payload_types, ci);
+    RentACodec::RegisterRedPayloadType(&red_payload_types, ci);
+  }
+}
+
+RentACodec::StackParameters::~StackParameters() = default;
+
+AudioEncoder* RentACodec::RentEncoderStack(StackParameters* param) {
+  RTC_DCHECK(param->speech_encoder);
+
+  if (param->use_codec_fec) {
+    // Switch FEC on. On failure, remember that FEC is off.
+    if (!param->speech_encoder->SetFec(true))
+      param->use_codec_fec = false;
+  } else {
+    // Switch FEC off. This shouldn't fail.
+    const bool success = param->speech_encoder->SetFec(false);
+    RTC_DCHECK(success);
+  }
+
+  auto pt = [&param](const std::map<int, int>& m) {
+    auto it = m.find(param->speech_encoder->SampleRateHz());
+    return it == m.end() ? rtc::Optional<int>()
+                         : rtc::Optional<int>(it->second);
+  };
+  auto cng_pt = pt(param->cng_payload_types);
+  param->use_cng =
+      param->use_cng && cng_pt && param->speech_encoder->NumChannels() == 1;
+  auto red_pt = pt(param->red_payload_types);
+  param->use_red = param->use_red && red_pt;
+
+  if (param->use_cng || param->use_red) {
+    // The RED and CNG encoders need to be in sync with the speech encoder, so
+    // reset the latter to ensure its buffer is empty.
+    param->speech_encoder->Reset();
+  }
+  encoder_stack_ = param->speech_encoder;
+  if (param->use_red) {
+    red_encoder_ = CreateRedEncoder(encoder_stack_, *red_pt);
+    if (red_encoder_)
+      encoder_stack_ = red_encoder_.get();
+  } else {
+    red_encoder_.reset();
+  }
+  if (param->use_cng) {
+    cng_encoder_ = CreateCngEncoder(encoder_stack_, *cng_pt, param->vad_mode);
+    encoder_stack_ = cng_encoder_.get();
+  } else {
+    cng_encoder_.reset();
+  }
+  return encoder_stack_;
+}
+
+AudioDecoder* RentACodec::RentIsacDecoder() {
+  if (!isac_decoder_)
+    isac_decoder_ = CreateIsacDecoder(&isac_bandwidth_info_);
+  return isac_decoder_.get();
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
new file mode 100644
index 0000000000..b1dcc9196c
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
@@ -0,0 +1,249 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+
+#include <stddef.h>
+#include <map>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/typedefs.h"
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+#else
+// Dummy implementation, for when we don't have iSAC.
+namespace webrtc {
+class LockedIsacBandwidthInfo {};
+}
+#endif
+
+namespace webrtc {
+
+struct CodecInst;
+
+namespace acm2 {
+
+class RentACodec {
+ public:
+  enum class CodecId {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    kISAC,
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    kISACSWB,
+#endif
+    // Mono
+    kPCM16B,
+    kPCM16Bwb,
+    kPCM16Bswb32kHz,
+    // Stereo
+    kPCM16B_2ch,
+    kPCM16Bwb_2ch,
+    kPCM16Bswb32kHz_2ch,
+    // Mono
+    kPCMU,
+    kPCMA,
+    // Stereo
+    kPCMU_2ch,
+    kPCMA_2ch,
+#ifdef WEBRTC_CODEC_ILBC
+    kILBC,
+#endif
+#ifdef WEBRTC_CODEC_G722
+    kG722,      // Mono
+    kG722_2ch,  // Stereo
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    kOpus,  // Mono and stereo
+#endif
+    kCNNB,
+    kCNWB,
+    kCNSWB,
+#ifdef ENABLE_48000_HZ
+    kCNFB,
+#endif
+    kAVT,
+#ifdef WEBRTC_CODEC_RED
+    kRED,
+#endif
+    kNumCodecs,  // Implementation detail. Don't use.
+
+// Set unsupported codecs to -1.
+#if !defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX)
+    kISAC = -1,
+#endif
+#ifndef WEBRTC_CODEC_ISAC
+    kISACSWB = -1,
+#endif
+    // 48 kHz not supported, always set to -1.
+    kPCM16Bswb48kHz = -1,
+#ifndef WEBRTC_CODEC_ILBC
+    kILBC = -1,
+#endif
+#ifndef WEBRTC_CODEC_G722
+    kG722 = -1,      // Mono
+    kG722_2ch = -1,  // Stereo
+#endif
+#ifndef WEBRTC_CODEC_OPUS
+    kOpus = -1,  // Mono and stereo
+#endif
+#ifndef WEBRTC_CODEC_RED
+    kRED = -1,
+#endif
+#ifndef ENABLE_48000_HZ
+    kCNFB = -1,
+#endif
+
+    kNone = -1
+  };
+
+  enum class NetEqDecoder {
+    kDecoderPCMu,
+    kDecoderPCMa,
+    kDecoderPCMu_2ch,
+    kDecoderPCMa_2ch,
+    kDecoderILBC,
+    kDecoderISAC,
+    kDecoderISACswb,
+    kDecoderPCM16B,
+    kDecoderPCM16Bwb,
+    kDecoderPCM16Bswb32kHz,
+    kDecoderPCM16Bswb48kHz,
+    kDecoderPCM16B_2ch,
+    kDecoderPCM16Bwb_2ch,
+    kDecoderPCM16Bswb32kHz_2ch,
+    kDecoderPCM16Bswb48kHz_2ch,
+    kDecoderPCM16B_5ch,
+    kDecoderG722,
+    kDecoderG722_2ch,
+    kDecoderRED,
+    kDecoderAVT,
+    kDecoderCNGnb,
+    kDecoderCNGwb,
+    kDecoderCNGswb32kHz,
+    kDecoderCNGswb48kHz,
+    kDecoderArbitrary,
+    kDecoderOpus,
+    kDecoderOpus_2ch,
+  };
+
+  static inline size_t NumberOfCodecs() {
+    return static_cast<size_t>(CodecId::kNumCodecs);
+  }
+
+  static inline rtc::Optional<int> CodecIndexFromId(CodecId codec_id) {
+    const int i = static_cast<int>(codec_id);
+    return i >= 0 && i < static_cast<int>(NumberOfCodecs())
+               ? rtc::Optional<int>(i)
+               : rtc::Optional<int>();
+  }
+
+  static inline rtc::Optional<CodecId> CodecIdFromIndex(int codec_index) {
+    return static_cast<size_t>(codec_index) < NumberOfCodecs()
+               ? rtc::Optional<RentACodec::CodecId>(
+                     static_cast<RentACodec::CodecId>(codec_index))
+               : rtc::Optional<RentACodec::CodecId>();
+  }
+
+  static rtc::Optional<CodecId> CodecIdByParams(const char* payload_name,
+                                                int sampling_freq_hz,
+                                                size_t channels);
+  static rtc::Optional<CodecInst> CodecInstById(CodecId codec_id);
+  static rtc::Optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
+  static rtc::Optional<CodecInst> CodecInstByParams(const char* payload_name,
+                                                    int sampling_freq_hz,
+                                                    size_t channels);
+  static bool IsCodecValid(const CodecInst& codec_inst);
+
+  static inline bool IsPayloadTypeValid(int payload_type) {
+    return payload_type >= 0 && payload_type <= 127;
+  }
+
+  static rtc::ArrayView<const CodecInst> Database();
+
+  static rtc::Optional<bool> IsSupportedNumChannels(CodecId codec_id,
+                                                    size_t num_channels);
+
+  static rtc::Optional<NetEqDecoder> NetEqDecoderFromCodecId(
+      CodecId codec_id,
+      size_t num_channels);
+
+  // Parse codec_inst and extract payload types. If the given CodecInst was for
+  // the wrong sort of codec, return kSkip; otherwise, if the rate was illegal,
+  // return kBadFreq; otherwise, update the given RTP timestamp rate (Hz) ->
+  // payload type map and return kOk.
+  enum class RegistrationResult { kOk, kSkip, kBadFreq };
+  static RegistrationResult RegisterCngPayloadType(std::map<int, int>* pt_map,
+                                                   const CodecInst& codec_inst);
+  static RegistrationResult RegisterRedPayloadType(std::map<int, int>* pt_map,
+                                                   const CodecInst& codec_inst);
+
+  RentACodec();
+  ~RentACodec();
+
+  // Creates and returns an audio encoder built to the given specification.
+  // Returns null in case of error. The returned encoder is live until the next
+  // successful call to this function, or until the Rent-A-Codec is destroyed.
+  AudioEncoder* RentEncoder(const CodecInst& codec_inst);
+
+  struct StackParameters {
+    StackParameters();
+    ~StackParameters();
+
+    AudioEncoder* speech_encoder = nullptr;
+    bool use_codec_fec = false;
+    bool use_red = false;
+    bool use_cng = false;
+    ACMVADMode vad_mode = VADNormal;
+
+    // Maps from RTP timestamp rate (in Hz) to payload type.
+    std::map<int, int> cng_payload_types;
+    std::map<int, int> red_payload_types;
+  };
+
+  // Creates and returns an audio encoder stack constructed to the given
+  // specification. If the specification isn't compatible with the encoder, it
+  // will be changed to match (things will be switched off). The returned
+  // encoder is live until the next successful call to this function, or until
+  // the Rent-A-Codec is destroyed.
+  AudioEncoder* RentEncoderStack(StackParameters* param);
+
+  // The last return value of RentEncoderStack, or null if it hasn't been
+  // called.
+  AudioEncoder* GetEncoderStack() const { return encoder_stack_; }
+
+  // Creates and returns an iSAC decoder, which will remain live until the
+  // Rent-A-Codec is destroyed. Subsequent calls will simply return the same
+  // object.
+  AudioDecoder* RentIsacDecoder();
+
+ private:
+  rtc::scoped_ptr<AudioEncoder> speech_encoder_;
+  rtc::scoped_ptr<AudioEncoder> cng_encoder_;
+  rtc::scoped_ptr<AudioEncoder> red_encoder_;
+  rtc::scoped_ptr<AudioDecoder> isac_decoder_;
+  AudioEncoder* encoder_stack_ = nullptr;
+  LockedIsacBandwidthInfo isac_bandwidth_info_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(RentACodec);
+};
+
+}  // namespace acm2
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
index 6c232615a7..e838488e53 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
@@ -8,36 +8,34 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <cstring>
-
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/arraysize.h"
-#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/codec_owner.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 
 namespace webrtc {
 namespace acm2 {
 
 using ::testing::Return;
-using ::testing::InSequence;
 
 namespace {
 const int kDataLengthSamples = 80;
 const int kPacketSizeSamples = 2 * kDataLengthSamples;
 const int16_t kZeroData[kDataLengthSamples] = {0};
-const CodecInst kDefaultCodecInst =
-    {0, "pcmu", 8000, kPacketSizeSamples, 1, 64000};
+const CodecInst kDefaultCodecInst = {0, "pcmu", 8000, kPacketSizeSamples,
+                                     1, 64000};
 const int kCngPt = 13;
 }  // namespace
 
-class CodecOwnerTest : public ::testing::Test {
+class RentACodecTestF : public ::testing::Test {
  protected:
-  CodecOwnerTest() : timestamp_(0) {}
-
   void CreateCodec() {
-    ASSERT_TRUE(
-        codec_owner_.SetEncoders(kDefaultCodecInst, kCngPt, VADNormal, -1));
+    speech_encoder_ = rent_a_codec_.RentEncoder(kDefaultCodecInst);
+    ASSERT_TRUE(speech_encoder_);
+    RentACodec::StackParameters param;
+    param.use_cng = true;
+    param.speech_encoder = speech_encoder_;
+    encoder_ = rent_a_codec_.RentEncoderStack(&param);
   }
 
   void EncodeAndVerify(size_t expected_out_length,
@@ -46,8 +44,8 @@ class CodecOwnerTest : public ::testing::Test {
                        int expected_send_even_if_empty) {
     uint8_t out[kPacketSizeSamples];
     AudioEncoder::EncodedInfo encoded_info;
-    encoded_info = codec_owner_.Encoder()->Encode(
-        timestamp_, kZeroData, kDataLengthSamples, kPacketSizeSamples, out);
+    encoded_info =
+        encoder_->Encode(timestamp_, kZeroData, kPacketSizeSamples, out);
     timestamp_ += kDataLengthSamples;
     EXPECT_TRUE(encoded_info.redundant.empty());
     EXPECT_EQ(expected_out_length, encoded_info.encoded_bytes);
@@ -59,43 +57,10 @@ class CodecOwnerTest : public ::testing::Test {
                 encoded_info.send_even_if_empty);
   }
 
-  // Verify that the speech encoder's Reset method is called when CNG or RED
-  // (or both) are switched on, but not when they're switched off.
-  void TestCngAndRedResetSpeechEncoder(bool use_cng, bool use_red) {
-    MockAudioEncoder speech_encoder;
-    EXPECT_CALL(speech_encoder, NumChannels())
-        .WillRepeatedly(Return(1));
-    EXPECT_CALL(speech_encoder, Max10MsFramesInAPacket())
-        .WillRepeatedly(Return(2));
-    EXPECT_CALL(speech_encoder, SampleRateHz())
-        .WillRepeatedly(Return(8000));
-    {
-      InSequence s;
-      EXPECT_CALL(speech_encoder, Mark("start off"));
-      EXPECT_CALL(speech_encoder, Mark("switch on"));
-      if (use_cng || use_red)
-        EXPECT_CALL(speech_encoder, Reset());
-      EXPECT_CALL(speech_encoder, Mark("start on"));
-      if (use_cng || use_red)
-        EXPECT_CALL(speech_encoder, Reset());
-      EXPECT_CALL(speech_encoder, Mark("switch off"));
-      EXPECT_CALL(speech_encoder, Die());
-    }
-
-    int cng_pt = use_cng ? 17 : -1;
-    int red_pt = use_red ? 19 : -1;
-    speech_encoder.Mark("start off");
-    codec_owner_.SetEncoders(&speech_encoder, -1, VADNormal, -1);
-    speech_encoder.Mark("switch on");
-    codec_owner_.ChangeCngAndRed(cng_pt, VADNormal, red_pt);
-    speech_encoder.Mark("start on");
-    codec_owner_.SetEncoders(&speech_encoder, cng_pt, VADNormal, red_pt);
-    speech_encoder.Mark("switch off");
-    codec_owner_.ChangeCngAndRed(-1, VADNormal, -1);
-  }
-
-  CodecOwner codec_owner_;
-  uint32_t timestamp_;
+  RentACodec rent_a_codec_;
+  AudioEncoder* speech_encoder_ = nullptr;
+  AudioEncoder* encoder_ = nullptr;
+  uint32_t timestamp_ = 0;
 };
 
 // This test verifies that CNG frames are delivered as expected. Since the frame
@@ -107,7 +72,7 @@ class CodecOwnerTest : public ::testing::Test {
 // AudioEncoder::EncodedInfo::send_even_if_empty set to true. (The reason to
 // produce an empty frame is to drive sending of DTMF packets in the RTP/RTCP
 // module.)
-TEST_F(CodecOwnerTest, VerifyCngFrames) {
+TEST_F(RentACodecTestF, VerifyCngFrames) {
   CreateCodec();
   uint32_t expected_timestamp = timestamp_;
   // Verify no frame.
@@ -136,75 +101,122 @@ TEST_F(CodecOwnerTest, VerifyCngFrames) {
   }
 }
 
-TEST_F(CodecOwnerTest, ExternalEncoder) {
-  MockAudioEncoder external_encoder;
-  codec_owner_.SetEncoders(&external_encoder, -1, VADNormal, -1);
+TEST(RentACodecTest, ExternalEncoder) {
   const int kSampleRateHz = 8000;
+  MockAudioEncoder external_encoder;
+  EXPECT_CALL(external_encoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(external_encoder, NumChannels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(external_encoder, SetFec(false)).WillRepeatedly(Return(true));
+
+  RentACodec rac;
+  RentACodec::StackParameters param;
+  param.speech_encoder = &external_encoder;
+  EXPECT_EQ(&external_encoder, rac.RentEncoderStack(&param));
   const int kPacketSizeSamples = kSampleRateHz / 100;
   int16_t audio[kPacketSizeSamples] = {0};
   uint8_t encoded[kPacketSizeSamples];
   AudioEncoder::EncodedInfo info;
-  EXPECT_CALL(external_encoder, SampleRateHz())
-      .WillRepeatedly(Return(kSampleRateHz));
 
   {
-    InSequence s;
+    ::testing::InSequence s;
     info.encoded_timestamp = 0;
     EXPECT_CALL(external_encoder,
-                EncodeInternal(0, audio, arraysize(encoded), encoded))
+                EncodeInternal(0, rtc::ArrayView<const int16_t>(audio),
+                               arraysize(encoded), encoded))
         .WillOnce(Return(info));
     EXPECT_CALL(external_encoder, Mark("A"));
     EXPECT_CALL(external_encoder, Mark("B"));
     info.encoded_timestamp = 2;
     EXPECT_CALL(external_encoder,
-                EncodeInternal(2, audio, arraysize(encoded), encoded))
+                EncodeInternal(2, rtc::ArrayView<const int16_t>(audio),
+                               arraysize(encoded), encoded))
         .WillOnce(Return(info));
     EXPECT_CALL(external_encoder, Die());
   }
 
-  info = codec_owner_.Encoder()->Encode(0, audio, arraysize(audio),
-                                        arraysize(encoded), encoded);
+  info = rac.GetEncoderStack()->Encode(0, audio, arraysize(encoded), encoded);
   EXPECT_EQ(0u, info.encoded_timestamp);
   external_encoder.Mark("A");
 
   // Change to internal encoder.
   CodecInst codec_inst = kDefaultCodecInst;
   codec_inst.pacsize = kPacketSizeSamples;
-  ASSERT_TRUE(codec_owner_.SetEncoders(codec_inst, -1, VADNormal, -1));
+  param.speech_encoder = rac.RentEncoder(codec_inst);
+  ASSERT_TRUE(param.speech_encoder);
+  EXPECT_EQ(param.speech_encoder, rac.RentEncoderStack(&param));
+
   // Don't expect any more calls to the external encoder.
-  info = codec_owner_.Encoder()->Encode(1, audio, arraysize(audio),
-                                        arraysize(encoded), encoded);
+  info = rac.GetEncoderStack()->Encode(1, audio, arraysize(encoded), encoded);
   external_encoder.Mark("B");
 
   // Change back to external encoder again.
-  codec_owner_.SetEncoders(&external_encoder, -1, VADNormal, -1);
-  info = codec_owner_.Encoder()->Encode(2, audio, arraysize(audio),
-                                        arraysize(encoded), encoded);
+  param.speech_encoder = &external_encoder;
+  EXPECT_EQ(&external_encoder, rac.RentEncoderStack(&param));
+  info = rac.GetEncoderStack()->Encode(2, audio, arraysize(encoded), encoded);
   EXPECT_EQ(2u, info.encoded_timestamp);
 }
 
-TEST_F(CodecOwnerTest, CngResetsSpeechEncoder) {
+// Verify that the speech encoder's Reset method is called when CNG or RED
+// (or both) are switched on, but not when they're switched off.
+void TestCngAndRedResetSpeechEncoder(bool use_cng, bool use_red) {
+  MockAudioEncoder speech_encoder;
+  EXPECT_CALL(speech_encoder, NumChannels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(speech_encoder, Max10MsFramesInAPacket())
+      .WillRepeatedly(Return(2));
+  EXPECT_CALL(speech_encoder, SampleRateHz()).WillRepeatedly(Return(8000));
+  EXPECT_CALL(speech_encoder, SetFec(false)).WillRepeatedly(Return(true));
+  {
+    ::testing::InSequence s;
+    EXPECT_CALL(speech_encoder, Mark("disabled"));
+    EXPECT_CALL(speech_encoder, Mark("enabled"));
+    if (use_cng || use_red)
+      EXPECT_CALL(speech_encoder, Reset());
+    EXPECT_CALL(speech_encoder, Die());
+  }
+
+  RentACodec::StackParameters param1, param2;
+  param1.speech_encoder = &speech_encoder;
+  param2.speech_encoder = &speech_encoder;
+  param2.use_cng = use_cng;
+  param2.use_red = use_red;
+  speech_encoder.Mark("disabled");
+  RentACodec rac;
+  rac.RentEncoderStack(&param1);
+  speech_encoder.Mark("enabled");
+  rac.RentEncoderStack(&param2);
+}
+
+TEST(RentACodecTest, CngResetsSpeechEncoder) {
   TestCngAndRedResetSpeechEncoder(true, false);
 }
 
-TEST_F(CodecOwnerTest, RedResetsSpeechEncoder) {
+TEST(RentACodecTest, RedResetsSpeechEncoder) {
   TestCngAndRedResetSpeechEncoder(false, true);
 }
 
-TEST_F(CodecOwnerTest, CngAndRedResetsSpeechEncoder) {
+TEST(RentACodecTest, CngAndRedResetsSpeechEncoder) {
   TestCngAndRedResetSpeechEncoder(true, true);
 }
 
-TEST_F(CodecOwnerTest, NoCngAndRedNoSpeechEncoderReset) {
+TEST(RentACodecTest, NoCngAndRedNoSpeechEncoderReset) {
   TestCngAndRedResetSpeechEncoder(false, false);
 }
 
-TEST_F(CodecOwnerTest, SetEncodersError) {
-  CodecInst codec_inst = kDefaultCodecInst;
-  static const char bad_name[] = "Robert'); DROP TABLE Students;";
-  std::memcpy(codec_inst.plname, bad_name, sizeof bad_name);
-  EXPECT_FALSE(codec_owner_.SetEncoders(codec_inst, -1, VADNormal, -1));
+TEST(RentACodecTest, RentEncoderError) {
+  const CodecInst codec_inst = {
+      0, "Robert'); DROP TABLE Students;", 8000, 160, 1, 64000};
+  RentACodec rent_a_codec;
+  EXPECT_FALSE(rent_a_codec.RentEncoder(codec_inst));
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(RentACodecTest, RentEncoderStackWithoutSpeechEncoder) {
+  RentACodec::StackParameters sp;
+  EXPECT_EQ(nullptr, sp.speech_encoder);
+  EXPECT_DEATH(RentACodec().RentEncoderStack(&sp), "");
 }
+#endif
 
 }  // namespace acm2
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_coding.gypi b/webrtc/modules/audio_coding/audio_coding.gypi
index bc3c48d075..abdb1915c3 100644
--- a/webrtc/modules/audio_coding/audio_coding.gypi
+++ b/webrtc/modules/audio_coding/audio_coding.gypi
@@ -19,12 +19,195 @@
     'codecs/isac/isacfix.gypi',
     'codecs/pcm16b/pcm16b.gypi',
     'codecs/red/red.gypi',
-    'main/audio_coding_module.gypi',
     'neteq/neteq.gypi',
   ],
+  'variables': {
+    'audio_coding_dependencies': [
+      'cng',
+      'g711',
+      'pcm16b',
+      '<(webrtc_root)/common.gyp:webrtc_common',
+      '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+      '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+    ],
+    'audio_coding_defines': [],
+    'conditions': [
+      ['include_opus==1', {
+        'audio_coding_dependencies': ['webrtc_opus',],
+        'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
+      }],
+      ['build_with_mozilla==0', {
+        'conditions': [
+          ['target_arch=="arm"', {
+            'audio_coding_dependencies': ['isac_fix',],
+            'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
+          }, {
+            'audio_coding_dependencies': ['isac',],
+            'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
+          }],
+        ],
+        'audio_coding_dependencies': ['g722',],
+        'audio_coding_defines': ['WEBRTC_CODEC_G722',],
+      }],
+      ['build_with_mozilla==0 and build_with_chromium==0', {
+        'audio_coding_dependencies': ['ilbc', 'red',],
+        'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
+      }],
+    ],
+  },
+  'targets': [
+    {
+      'target_name': 'rent_a_codec',
+      'type': 'static_library',
+      'defines': [
+        '<@(audio_coding_defines)',
+      ],
+      'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
+      ],
+      'include_dirs': [
+        '<(webrtc_root)',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '<(webrtc_root)',
+        ],
+      },
+      'sources': [
+        'acm2/acm_codec_database.cc',
+        'acm2/acm_codec_database.h',
+        'acm2/rent_a_codec.cc',
+        'acm2/rent_a_codec.h',
+      ],
+    },
+    {
+      'target_name': 'audio_coding_module',
+      'type': 'static_library',
+      'defines': [
+        '<@(audio_coding_defines)',
+      ],
+      'dependencies': [
+        '<@(audio_coding_dependencies)',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+        'neteq',
+        'rent_a_codec',
+      ],
+      'include_dirs': [
+        'include',
+        '../include',
+        '<(webrtc_root)',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          'include',
+          '../include',
+          '<(webrtc_root)',
+        ],
+      },
+      'conditions': [
+        ['include_opus==1', {
+          'export_dependent_settings': ['webrtc_opus'],
+        }],
+      ],
+      'sources': [
+        'acm2/acm_common_defs.h',
+        'acm2/acm_receiver.cc',
+        'acm2/acm_receiver.h',
+        'acm2/acm_resampler.cc',
+        'acm2/acm_resampler.h',
+        'acm2/audio_coding_module.cc',
+        'acm2/audio_coding_module_impl.cc',
+        'acm2/audio_coding_module_impl.h',
+        'acm2/call_statistics.cc',
+        'acm2/call_statistics.h',
+        'acm2/codec_manager.cc',
+        'acm2/codec_manager.h',
+        'acm2/initial_delay_manager.cc',
+        'acm2/initial_delay_manager.h',
+        'include/audio_coding_module.h',
+        'include/audio_coding_module_typedefs.h',
+      ],
+    },
+  ],
   'conditions': [
     ['include_opus==1', {
       'includes': ['codecs/opus/opus.gypi',],
     }],
+    ['include_tests==1', {
+      'targets': [
+        {
+          'target_name': 'acm_receive_test',
+          'type': 'static_library',
+          'defines': [
+            '<@(audio_coding_defines)',
+          ],
+          'dependencies': [
+            '<@(audio_coding_dependencies)',
+            'audio_coding_module',
+            'neteq_unittest_tools',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'acm2/acm_receive_test_oldapi.cc',
+            'acm2/acm_receive_test_oldapi.h',
+          ],
+        }, # acm_receive_test
+        {
+          'target_name': 'acm_send_test',
+          'type': 'static_library',
+          'defines': [
+            '<@(audio_coding_defines)',
+          ],
+          'dependencies': [
+            '<@(audio_coding_dependencies)',
+            'audio_coding_module',
+            'neteq_unittest_tools',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'acm2/acm_send_test_oldapi.cc',
+            'acm2/acm_send_test_oldapi.h',
+          ],
+        }, # acm_send_test
+        {
+          'target_name': 'delay_test',
+          'type': 'executable',
+          'dependencies': [
+            'audio_coding_module',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(webrtc_root)/common.gyp:webrtc_common',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+             'test/delay_test.cc',
+             'test/Channel.cc',
+             'test/PCMFile.cc',
+             'test/utility.cc',
+           ],
+        }, # delay_test
+        {
+          'target_name': 'insert_packet_with_timing',
+          'type': 'executable',
+          'dependencies': [
+            'audio_coding_module',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(webrtc_root)/common.gyp:webrtc_common',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+             'test/insert_packet_with_timing.cc',
+             'test/Channel.cc',
+             'test/PCMFile.cc',
+           ],
+        }, # delay_test
+      ],
+    }],
   ],
 }
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
index 08d101c5ae..d2984b97b0 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
@@ -13,12 +13,14 @@
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
                          int sample_rate_hz, size_t max_decoded_bytes,
                          int16_t* decoded, SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
   int duration = PacketDuration(encoded, encoded_len);
   if (duration >= 0 &&
       duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
@@ -31,6 +33,7 @@ int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
 int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
                                   int sample_rate_hz, size_t max_decoded_bytes,
                                   int16_t* decoded, SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
   int duration = PacketDurationRedundant(encoded, encoded_len);
   if (duration >= 0 &&
       duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
@@ -40,12 +43,6 @@ int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
                                  speech_type);
 }
 
-int AudioDecoder::DecodeInternal(const uint8_t* encoded, size_t encoded_len,
-                                 int sample_rate_hz, int16_t* decoded,
-                                 SpeechType* speech_type) {
-  return kNotImplemented;
-}
-
 int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
                                           size_t encoded_len,
                                           int sample_rate_hz, int16_t* decoded,
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h
index 6189be098d..81ac873183 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -14,7 +14,7 @@
 #include <stdlib.h>  // NULL
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -41,21 +41,21 @@ class AudioDecoder {
   // is set to kComfortNoise, otherwise it is kSpeech. The desired output
   // sample rate is provided in |sample_rate_hz|, which must be valid for the
   // codec at hand.
-  virtual int Decode(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     size_t max_decoded_bytes,
-                     int16_t* decoded,
-                     SpeechType* speech_type);
+  int Decode(const uint8_t* encoded,
+             size_t encoded_len,
+             int sample_rate_hz,
+             size_t max_decoded_bytes,
+             int16_t* decoded,
+             SpeechType* speech_type);
 
   // Same as Decode(), but interfaces to the decoders redundant decode function.
   // The default implementation simply calls the regular Decode() method.
-  virtual int DecodeRedundant(const uint8_t* encoded,
-                              size_t encoded_len,
-                              int sample_rate_hz,
-                              size_t max_decoded_bytes,
-                              int16_t* decoded,
-                              SpeechType* speech_type);
+  int DecodeRedundant(const uint8_t* encoded,
+                      size_t encoded_len,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded,
+                      SpeechType* speech_type);
 
   // Indicates if the decoder implements the DecodePlc method.
   virtual bool HasDecodePlc() const;
@@ -107,7 +107,7 @@ class AudioDecoder {
                              size_t encoded_len,
                              int sample_rate_hz,
                              int16_t* decoded,
-                             SpeechType* speech_type);
+                             SpeechType* speech_type) = 0;
 
   virtual int DecodeRedundantInternal(const uint8_t* encoded,
                                       size_t encoded_len,
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 6d763005ac..e99fc30995 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -9,7 +9,9 @@
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
@@ -21,13 +23,14 @@ int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
 }
 
-AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
-                                               const int16_t* audio,
-                                               size_t num_samples_per_channel,
-                                               size_t max_encoded_bytes,
-                                               uint8_t* encoded) {
-  RTC_CHECK_EQ(num_samples_per_channel,
-               static_cast<size_t>(SampleRateHz() / 100));
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded) {
+  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
+  RTC_CHECK_EQ(audio.size(),
+               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
   EncodedInfo info =
       EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
   RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index cda9d86f2e..a46b0e86a7 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -14,6 +14,7 @@
 #include <algorithm>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -60,7 +61,7 @@ class AudioEncoder {
   // Returns the input sample rate in Hz and the number of input channels.
   // These are constants set at instantiation time.
   virtual int SampleRateHz() const = 0;
-  virtual int NumChannels() const = 0;
+  virtual size_t NumChannels() const = 0;
 
   // Returns the rate at which the RTP timestamps are updated. The default
   // implementation returns SampleRateHz().
@@ -91,13 +92,12 @@ class AudioEncoder {
   // Encode() checks some preconditions, calls EncodeInternal() which does the
   // actual work, and then checks some postconditions.
   EncodedInfo Encode(uint32_t rtp_timestamp,
-                     const int16_t* audio,
-                     size_t num_samples_per_channel,
+                     rtc::ArrayView<const int16_t> audio,
                      size_t max_encoded_bytes,
                      uint8_t* encoded);
 
   virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                                     const int16_t* audio,
+                                     rtc::ArrayView<const int16_t> audio,
                                      size_t max_encoded_bytes,
                                      uint8_t* encoded) = 0;
 
@@ -125,7 +125,7 @@ class AudioEncoder {
   // Tells the encoder about the highest sample rate the decoder is expected to
   // use when decoding the bitstream. The encoder would typically use this
   // information to adjust the quality of the encoding. The default
-  // implementation just returns true.
+  // implementation does nothing.
   virtual void SetMaxPlaybackRate(int frequency_hz);
 
   // Tells the encoder what the projected packet loss rate is. The rate is in
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 121524633c..180166c40c 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 
 #include <algorithm>
 #include <limits>
@@ -75,7 +75,7 @@ int AudioEncoderCng::SampleRateHz() const {
   return speech_encoder_->SampleRateHz();
 }
 
-int AudioEncoderCng::NumChannels() const {
+size_t AudioEncoderCng::NumChannels() const {
   return 1;
 }
 
@@ -97,7 +97,7 @@ int AudioEncoderCng::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   RTC_CHECK_GE(max_encoded_bytes,
@@ -106,9 +106,8 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
   RTC_CHECK_EQ(speech_buffer_.size(),
                rtp_timestamps_.size() * samples_per_10ms_frame);
   rtp_timestamps_.push_back(rtp_timestamp);
-  for (size_t i = 0; i < samples_per_10ms_frame; ++i) {
-    speech_buffer_.push_back(audio[i]);
-  }
+  RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
+  speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
   const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
   if (rtp_timestamps_.size() < frames_to_encode) {
     return EncodedInfo();
@@ -242,9 +241,12 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   AudioEncoder::EncodedInfo info;
   for (size_t i = 0; i < frames_to_encode; ++i) {
-    info = speech_encoder_->Encode(
-        rtp_timestamps_.front(), &speech_buffer_[i * samples_per_10ms_frame],
-        samples_per_10ms_frame, max_encoded_bytes, encoded);
+    info =
+        speech_encoder_->Encode(rtp_timestamps_.front(),
+                                rtc::ArrayView<const int16_t>(
+                                    &speech_buffer_[i * samples_per_10ms_frame],
+                                    samples_per_10ms_frame),
+                                max_encoded_bytes, encoded);
     if (i + 1 == frames_to_encode) {
       RTC_CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
     } else {
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index 3ca9eb60f3..87383e2ac5 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_INCLUDE_AUDIO_ENCODER_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_INCLUDE_AUDIO_ENCODER_CNG_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
 
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/vad/include/vad.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 
 namespace webrtc {
 
@@ -32,7 +32,7 @@ class AudioEncoderCng final : public AudioEncoder {
   struct Config {
     bool IsOk() const;
 
-    int num_channels = 1;
+    size_t num_channels = 1;
     int payload_type = 13;
     // Caller keeps ownership of the AudioEncoder object.
     AudioEncoder* speech_encoder = nullptr;
@@ -51,13 +51,13 @@ class AudioEncoderCng final : public AudioEncoder {
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
@@ -92,4 +92,4 @@ class AudioEncoderCng final : public AudioEncoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_INCLUDE_AUDIO_ENCODER_CNG_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 0b837a0f12..feb3ed1f0a 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -13,7 +13,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/vad/mock/mock_vad.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 
 using ::testing::Return;
@@ -75,8 +75,10 @@ class AudioEncoderCngTest : public ::testing::Test {
 
   void Encode() {
     ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
-    encoded_info_ = cng_->Encode(timestamp_, audio_, num_audio_samples_10ms_,
-                                 encoded_.size(), &encoded_[0]);
+    encoded_info_ = cng_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
+        encoded_.size(), &encoded_[0]);
     timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
   }
 
diff --git a/webrtc/modules/audio_coding/codecs/cng/cng.gypi b/webrtc/modules/audio_coding/codecs/cng/cng.gypi
index 78dc41a94f..c020f4740d 100644
--- a/webrtc/modules/audio_coding/codecs/cng/cng.gypi
+++ b/webrtc/modules/audio_coding/codecs/cng/cng.gypi
@@ -15,23 +15,13 @@
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'include/audio_encoder_cng.h',
-        'include/webrtc_cng.h',
         'audio_encoder_cng.cc',
-        'webrtc_cng.c',
+        'audio_encoder_cng.h',
         'cng_helpfuns.c',
         'cng_helpfuns.h',
+        'webrtc_cng.c',
+        'webrtc_cng.h',
       ],
     },
   ], # targets
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
index 35660c4c3c..64bea1e26f 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -9,8 +9,8 @@
  */
 
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INCLUDE_WEBRTC_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INCLUDE_WEBRTC_CNG_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
 
 #include <stddef.h>
 #include "webrtc/typedefs.h"
@@ -144,7 +144,7 @@ int16_t WebRtcCng_Generate(CNG_dec_inst* cng_inst, int16_t* outData,
  * WebRtcCng_GetErrorCodeEnc/Dec(...)
  *
  * This functions can be used to check the error code of a CNG instance. When
- * a function returns -1 a error code will be set for that instance. The 
+ * a function returns -1 a error code will be set for that instance. The
  * function below extract the code of the last error that occurred in the
  * specified instance.
  *
@@ -160,4 +160,4 @@ int16_t WebRtcCng_GetErrorCodeDec(CNG_dec_inst* cng_inst);
 }
 #endif
 
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INCLUDE_WEBRTC_CNG_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index 12306d9167..9757b4a010 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 
-#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 7bc37d3b7a..9dc3a6fd7a 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
@@ -60,4 +60,4 @@ class AudioDecoderPcmA final : public AudioDecoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index dde3cc6799..ff61db8e8d 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -8,27 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 namespace webrtc {
 
 namespace {
 
-int16_t NumSamplesPerFrame(int num_channels,
-                           int frame_size_ms,
-                           int sample_rate_hz) {
-  int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
-  RTC_CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
-      << "Frame size too large.";
-  return static_cast<int16_t>(samples_per_frame);
-}
-
 template <typename T>
 typename T::Config CreateConfig(const CodecInst& codec_inst) {
   typename T::Config config;
@@ -50,9 +41,8 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
       payload_type_(config.payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
-      full_frame_samples_(NumSamplesPerFrame(config.num_channels,
-                                             config.frame_size_ms,
-                                             sample_rate_hz_)),
+      full_frame_samples_(
+          config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
       first_timestamp_in_buffer_(0) {
   RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
   RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
@@ -70,7 +60,7 @@ int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
-int AudioEncoderPcm::NumChannels() const {
+size_t AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
@@ -83,21 +73,19 @@ size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
 }
 
 int AudioEncoderPcm::GetTargetBitrate() const {
-  return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
+  return static_cast<int>(
+      8 * BytesPerSample() * SampleRateHz() * NumChannels());
 }
 
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  const int num_samples = SampleRateHz() / 100 * NumChannels();
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
-  for (int i = 0; i < num_samples; ++i) {
-    speech_buffer_.push_back(audio[i]);
-  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
@@ -125,7 +113,7 @@ size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
   return WebRtcG711_EncodeA(audio, input_len, encoded);
 }
 
-int AudioEncoderPcmA::BytesPerSample() const {
+size_t AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
@@ -138,7 +126,7 @@ size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
-int AudioEncoderPcmU::BytesPerSample() const {
+size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index e532f9b1bc..b839488628 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
 
 #include <vector>
 
@@ -25,7 +25,7 @@ class AudioEncoderPcm : public AudioEncoder {
     bool IsOk() const;
 
     int frame_size_ms;
-    int num_channels;
+    size_t num_channels;
     int payload_type;
 
    protected:
@@ -37,12 +37,12 @@ class AudioEncoderPcm : public AudioEncoder {
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
@@ -54,11 +54,11 @@ class AudioEncoderPcm : public AudioEncoder {
                             size_t input_len,
                             uint8_t* encoded) = 0;
 
-  virtual int BytesPerSample() const = 0;
+  virtual size_t BytesPerSample() const = 0;
 
  private:
   const int sample_rate_hz_;
-  const int num_channels_;
+  const size_t num_channels_;
   const int payload_type_;
   const size_t num_10ms_frames_per_packet_;
   const size_t full_frame_samples_;
@@ -83,7 +83,7 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
                     size_t input_len,
                     uint8_t* encoded) override;
 
-  int BytesPerSample() const override;
+  size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
@@ -105,7 +105,7 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
                     size_t input_len,
                     uint8_t* encoded) override;
 
-  int BytesPerSample() const override;
+  size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
@@ -114,4 +114,4 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
diff --git a/webrtc/modules/audio_coding/codecs/g711/g711.gypi b/webrtc/modules/audio_coding/codecs/g711/g711.gypi
index d35d7874e7..4b902809ea 100644
--- a/webrtc/modules/audio_coding/codecs/g711/g711.gypi
+++ b/webrtc/modules/audio_coding/codecs/g711/g711.gypi
@@ -14,25 +14,15 @@
       'dependencies': [
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'include/g711_interface.h',
-        'include/audio_decoder_pcm.h',
-        'include/audio_encoder_pcm.h',
+        'audio_decoder_pcm.cc',
+        'audio_decoder_pcm.h',
+        'audio_encoder_pcm.cc',
+        'audio_encoder_pcm.h',
         'g711_interface.c',
+        'g711_interface.h',
         'g711.c',
         'g711.h',
-        'audio_decoder_pcm.cc',
-        'audio_encoder_pcm.cc',
       ],
     },
   ], # targets
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h b/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
index f9867f4504..00854bbb2c 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h
+++ b/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_AUDIO_CODING_CODECS_G711_MAIN_INCLUDE_G711_INTERFACE_H_
-#define MODULES_AUDIO_CODING_CODECS_G711_MAIN_INCLUDE_G711_INTERFACE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -132,4 +132,4 @@ int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
 }
 #endif
 
-#endif /* MODULES_AUDIO_CODING_CODECS_G711_MAIN_INCLUDE_G711_INCLUDE_H_ */
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
diff --git a/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
index 94248f7a66..5675b1f8b0 100644
--- a/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
@@ -17,7 +17,7 @@
 #include <string.h>
 
 /* include API */
-#include "g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 /* Runtime statistics */
 #include <time.h>
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
index 55ebe7a315..7676e90d9e 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_decoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
 
 #include <string.h>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_decoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
index b9fa68fc48..7cc2ea9877 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/audio_decoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_DECODER_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_DECODER_G722_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
 
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
@@ -69,4 +69,4 @@ class AudioDecoderG722Stereo final : public AudioDecoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_DECODER_G722_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index 43b097fa0e..d7203b9da3 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
 
 #include <limits>
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
@@ -48,7 +48,7 @@ AudioEncoderG722::AudioEncoderG722(const Config& config)
   RTC_CHECK(config.IsOk());
   const size_t samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
     encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
@@ -68,7 +68,7 @@ int AudioEncoderG722::SampleRateHz() const {
   return kSampleRateHz;
 }
 
-int AudioEncoderG722::NumChannels() const {
+size_t AudioEncoderG722::NumChannels() const {
   return num_channels_;
 }
 
@@ -88,12 +88,12 @@ size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
 
 int AudioEncoderG722::GetTargetBitrate() const {
   // 4 bits/sample, 16000 samples/s/channel.
-  return 64000 * NumChannels();
+  return static_cast<int>(64000 * NumChannels());
 }
 
 AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
@@ -104,7 +104,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   // Deinterleave samples and save them in each channel's buffer.
   const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
   for (size_t i = 0; i < kSampleRateHz / 100; ++i)
-    for (int j = 0; j < num_channels_; ++j)
+    for (size_t j = 0; j < num_channels_; ++j)
       encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
 
   // If we don't yet have enough samples for a packet, we're done for now.
@@ -116,7 +116,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
   const size_t samples_per_channel = SamplesPerChannel();
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     const size_t encoded = WebRtcG722_Encode(
         encoders_[i].encoder, encoders_[i].speech_buffer.get(),
         samples_per_channel, encoders_[i].encoded_buffer.data());
@@ -127,12 +127,12 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   // channel and the interleaved stream encodes two samples per byte, most
   // significant half first.
   for (size_t i = 0; i < samples_per_channel / 2; ++i) {
-    for (int j = 0; j < num_channels_; ++j) {
+    for (size_t j = 0; j < num_channels_; ++j) {
       uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
       interleave_buffer_.data()[j] = two_samples >> 4;
       interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
     }
-    for (int j = 0; j < num_channels_; ++j)
+    for (size_t j = 0; j < num_channels_; ++j)
       encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 |
                                        interleave_buffer_.data()[2 * j + 1];
   }
@@ -145,7 +145,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
 
 void AudioEncoderG722::Reset() {
   num_10ms_frames_buffered_ = 0;
-  for (int i = 0; i < num_channels_; ++i)
+  for (size_t i = 0; i < num_channels_; ++i)
     RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index 12495c5f48..07d767e778 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
@@ -27,7 +27,7 @@ class AudioEncoderG722 final : public AudioEncoder {
 
     int payload_type = 9;
     int frame_size_ms = 20;
-    int num_channels = 1;
+    size_t num_channels = 1;
   };
 
   explicit AudioEncoderG722(const Config& config);
@@ -36,13 +36,13 @@ class AudioEncoderG722 final : public AudioEncoder {
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
@@ -59,7 +59,7 @@ class AudioEncoderG722 final : public AudioEncoder {
 
   size_t SamplesPerChannel() const;
 
-  const int num_channels_;
+  const size_t num_channels_;
   const int payload_type_;
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
@@ -70,4 +70,4 @@ class AudioEncoderG722 final : public AudioEncoder {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
diff --git a/webrtc/modules/audio_coding/codecs/g722/g722.gypi b/webrtc/modules/audio_coding/codecs/g722/g722.gypi
index aad11e3685..756fabe345 100644
--- a/webrtc/modules/audio_coding/codecs/g722/g722.gypi
+++ b/webrtc/modules/audio_coding/codecs/g722/g722.gypi
@@ -13,26 +13,16 @@
       'dependencies': [
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
         'audio_decoder_g722.cc',
+        'audio_decoder_g722.h',
         'audio_encoder_g722.cc',
-        'include/audio_decoder_g722.h',
-        'include/audio_encoder_g722.h',
-        'include/g722_interface.h',
+        'audio_encoder_g722.h',
         'g722_interface.c',
-        'g722_encode.c',
+        'g722_interface.h',
         'g722_decode.c',
         'g722_enc_dec.h',
+        'g722_encode.c',
       ],
     },
   ], # targets
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h b/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
index 5a46ef2ad5..b411ef0e8e 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ b/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_AUDIO_CODING_CODECS_G722_MAIN_INCLUDE_G722_INTERFACE_H_
-#define MODULES_AUDIO_CODING_CODECS_G722_MAIN_INCLUDE_G722_INTERFACE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -179,4 +179,4 @@ int16_t WebRtcG722_Version(char *versionStr, short len);
 #endif
 
 
-#endif /* MODULES_AUDIO_CODING_CODECS_G722_MAIN_INCLUDE_G722_INCLUDE_H_ */
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
diff --git a/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc b/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
index b473c138c6..c55a2eb357 100644
--- a/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -18,7 +18,7 @@
 #include "webrtc/typedefs.h"
 
 /* include API */
-#include "g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 /* Runtime statistics */
 #include <time.h>
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
index ba6284f33d..9ae0e1a95e 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
index fd52da7986..e890635da0 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_DECODER_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_DECODER_ILBC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
 
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
@@ -39,4 +39,4 @@ class AudioDecoderIlbc final : public AudioDecoder {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_DECODER_ILBC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index 065dc06817..ddd6dde31c 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
 
-#include <cstring>
+#include <algorithm>
 #include <limits>
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 namespace webrtc {
 
@@ -64,7 +64,7 @@ int AudioEncoderIlbc::SampleRateHz() const {
   return kSampleRateHz;
 }
 
-int AudioEncoderIlbc::NumChannels() const {
+size_t AudioEncoderIlbc::NumChannels() const {
   return 1;
 }
 
@@ -91,7 +91,7 @@ int AudioEncoderIlbc::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
@@ -101,9 +101,9 @@ AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Buffer input.
-  std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_,
-              audio,
-              kSampleRateHz / 100 * sizeof(audio[0]));
+  RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
 
   // If we don't yet have enough buffered input for a whole packet, we're done
   // for now.
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index 2bb3101fd4..102a274642 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_ENCODER_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_ENCODER_ILBC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 namespace webrtc {
 
@@ -36,12 +36,12 @@ class AudioEncoderIlbc final : public AudioEncoder {
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
@@ -60,4 +60,4 @@ class AudioEncoderIlbc final : public AudioEncoder {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_ENCODER_ILBC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi
index ac9f2e7b39..ffb0574588 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi
@@ -15,24 +15,13 @@
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'include/audio_decoder_ilbc.h',
-        'include/audio_encoder_ilbc.h',
-        'include/ilbc.h',
         'abs_quant.c',
         'abs_quant_loop.c',
         'audio_decoder_ilbc.cc',
+        'audio_decoder_ilbc.h',
         'audio_encoder_ilbc.cc',
+        'audio_encoder_ilbc.h',
         'augmented_cb_corr.c',
         'bw_expand.c',
         'cb_construct.c',
@@ -65,6 +54,7 @@
         'hp_input.c',
         'hp_output.c',
         'ilbc.c',
+        'ilbc.h',
         'index_conv_dec.c',
         'index_conv_enc.c',
         'init_decode.c',
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
index 3be9142c8c..c021f5be52 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
@@ -15,8 +15,8 @@
  *
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_ILBC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
 
 #include <stddef.h>
 
@@ -53,10 +53,10 @@ extern "C" {
    * memory location
    *
    * Input:
-   *      - XXX_xxxinst       : Pointer to created instance that should be
-   *                            assigned
-   *      - ILBCXXX_inst_Addr : Pointer to the desired memory space
-   *      - size              : The size that this structure occupies (in Word16)
+   *     - XXX_xxxinst       : Pointer to created instance that should be
+   *                           assigned
+   *     - ILBCXXX_inst_Addr : Pointer to the desired memory space
+   *     - size              : The size that this structure occupies (in Word16)
    *
    * Return value             :  0 - Ok
    *                            -1 - Error
@@ -76,10 +76,10 @@ extern "C" {
    * These functions create a instance to the specified structure
    *
    * Input:
-   *      - XXX_inst          : Pointer to created instance that should be created
+   *      - XXX_inst        : Pointer to created instance that should be created
    *
-   * Return value             :  0 - Ok
-   *                            -1 - Error
+   * Return value           :  0 - Ok
+   *                          -1 - Error
    */
 
   int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst);
@@ -255,4 +255,4 @@ extern "C" {
 }
 #endif
 
-#endif
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
index 1199c816d8..b440c7a45f 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -19,7 +19,7 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 /*---------------------------------------------------------------*
  *  Main program to test iLBC encoding and decoding
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
index f14192c2ae..7ffa4a7d0e 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -21,7 +21,7 @@ iLBC_test.c
 #include <stdio.h>
 #include <string.h>
 #include <time.h>
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 //#define JUNK_DATA
 #ifdef JUNK_DATA
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
index 303ede3e63..5454948287 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -21,13 +21,13 @@
 #include <stdio.h>
 #include <string.h>
 
-#include "defines.h"
-#include "nit_encode.h"
-#include "encode.h"
-#include "init_decode.h"
-#include "decode.h"
-#include "constants.h"
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/defines.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/nit_encode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/encode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/decode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/constants.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 #define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index b15ad942df..321dac3567 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -56,12 +56,12 @@ class AudioEncoderIsacT final : public AudioEncoder {
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 279f80d6fc..d4438cc775 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -88,7 +88,7 @@ int AudioEncoderIsacT<T>::SampleRateHz() const {
 }
 
 template <typename T>
-int AudioEncoderIsacT<T>::NumChannels() const {
+size_t AudioEncoderIsacT<T>::NumChannels() const {
   return 1;
 }
 
@@ -115,7 +115,7 @@ int AudioEncoderIsacT<T>::GetTargetBitrate() const {
 template <typename T>
 AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   if (!packet_in_progress_) {
@@ -127,7 +127,7 @@ AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
     IsacBandwidthInfo bwinfo = bwinfo_->Get();
     T::SetBandwidthInfo(isac_state_, &bwinfo);
   }
-  int r = T::Encode(isac_state_, audio, encoded);
+  int r = T::Encode(isac_state_, audio.data(), encoded);
   RTC_CHECK_GE(r, 0) << "Encode failed (error code "
                      << T::GetErrorCode(isac_state_) << ")";
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index 632a4fe825..32f36c5261 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -92,7 +92,7 @@ float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream,
   value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes,
                                out_data, &audio_type);
   clocks = clock() - clocks;
-  EXPECT_EQ(output_length_sample_, value);
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index b82af1c059..ac0fa350c9 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -112,12 +112,12 @@ int main(int argc, char* argv[]) {
   char version_number[20];
   int mode = -1, tmp, nbTest = 0; /*,sss;*/
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   FILE* fy;
   double kbps;
   size_t totalbits = 0;
   int totalsmpls = 0;
-#endif /* _DEBUG */
+#endif
 
   /* only one structure used for ISAC encoder */
   ISAC_MainStruct* ISAC_main_inst;
@@ -126,12 +126,12 @@ int main(int argc, char* argv[]) {
   BottleNeckModel BN_data;
   f_bn = NULL;
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   fy = fopen("bit_rate.dat", "w");
   fclose(fy);
   fy = fopen("bytes_frames.dat", "w");
   fclose(fy);
-#endif /* _DEBUG */
+#endif
 
   // histfile = fopen("histo.dat", "ab");
   // ratefile = fopen("rates.dat", "ab");
@@ -589,7 +589,7 @@ int main(int argc, char* argv[]) {
     fprintf(stderr, "  \rframe = %d", framecnt);
     framecnt++;
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 
     totalsmpls += declen;
     totalbits += 8 * stream_len;
@@ -598,15 +598,15 @@ int main(int argc, char* argv[]) {
     fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
     fclose(fy);
 
-#endif /* _DEBUG */
+#endif
   }
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   printf("\n\ntotal bits               = %" PRIuS " bits", totalbits);
   printf("\nmeasured average bitrate = %0.3f kbits/s",
          (double)totalbits * (FS / 1000) / totalsmpls);
   printf("\n");
-#endif /* _DEBUG */
+#endif
 
   /* Runtime statistics */
   runtime = (double)(clock() / (double)CLOCKS_PER_SEC - starttime);
diff --git a/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi b/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
index 47944b7f42..54cedb4e18 100644
--- a/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
+++ b/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
@@ -25,6 +25,19 @@
         './main/test/simpleKenny.c',
         './main/util/utility.c',
       ],
+      'conditions': [
+        ['OS=="win" and clang==1', {
+          'msvs_settings': {
+            'VCCLCompilerTool': {
+              'AdditionalOptions': [
+                # Disable warnings failing when compiling with Clang on Windows.
+                # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                '-Wno-format',
+              ],
+            },
+          },
+        }],
+      ],  # conditions.
     },
     # ReleaseTest-API
     {
@@ -63,6 +76,5 @@
         './main/util/utility.c',
       ],
     },
-
   ],
 }
diff --git a/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi b/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
index f10de56c5a..7730d16dc9 100644
--- a/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
+++ b/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
@@ -77,11 +77,6 @@
         'fix/source/structs.h',
       ],
       'conditions': [
-        ['OS!="win"', {
-          'defines': [
-            'WEBRTC_LINUX',
-          ],
-        }],
         ['target_arch=="arm" and arm_version>=7', {
           'sources': [
             'fix/source/lattice_armv7.S',
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 2e5badd82c..4cef8f7b3b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -73,10 +73,10 @@ int main(int argc, char* argv[]) {
   FILE* plFile;
   int32_t sendBN;
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   FILE* fy;
   double kbps;
-#endif /* _DEBUG */
+#endif
   size_t totalbits = 0;
   int totalsmpls = 0;
 
@@ -103,12 +103,12 @@ int main(int argc, char* argv[]) {
 
   BottleNeckModel BN_data;
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   fy = fopen("bit_rate.dat", "w");
   fclose(fy);
   fy = fopen("bytes_frames.dat", "w");
   fclose(fy);
-#endif /* _DEBUG */
+#endif
 
   /* Handling wrong input arguments in the command line */
   if ((argc < 3) || (argc > 17)) {
@@ -885,14 +885,14 @@ int main(int argc, char* argv[]) {
 
     totalsmpls += declen;
     totalbits += 8 * stream_len;
-#ifdef _DEBUG
+#if !defined(NDEBUG)
     kbps = ((double)sampFreqKHz * 1000.) / ((double)cur_framesmpls) * 8.0 *
            stream_len / 1000.0;  // kbits/s
     fy = fopen("bit_rate.dat", "a");
     fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
     fclose(fy);
 
-#endif /* _DEBUG */
+#endif
   }
   printf("\n");
   printf("total bits               = %" PRIuS " bits\n", totalbits);
diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
index 95426d89e1..66adde4be1 100644
--- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -24,7 +24,7 @@ class MockAudioEncoder final : public AudioEncoder {
   MOCK_METHOD1(Mark, void(std::string desc));
   MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
   MOCK_CONST_METHOD0(SampleRateHz, int());
-  MOCK_CONST_METHOD0(NumChannels, int());
+  MOCK_CONST_METHOD0(NumChannels, size_t());
   MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
   MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
   MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
@@ -32,7 +32,7 @@ class MockAudioEncoder final : public AudioEncoder {
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD4(EncodeInternal,
                EncodedInfo(uint32_t timestamp,
-                           const int16_t* audio,
+                           rtc::ArrayView<const int16_t> audio,
                            size_t max_encoded_bytes,
                            uint8_t* encoded));
   MOCK_METHOD0(Reset, void());
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index d1390e2ca4..f64e811afe 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/include/audio_decoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 
 #include "webrtc/base/checks.h"
 
@@ -17,7 +17,7 @@ namespace webrtc {
 AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
     : channels_(num_channels) {
   RTC_DCHECK(num_channels == 1 || num_channels == 2);
-  WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
+  WebRtcOpus_DecoderCreate(&dec_state_, channels_);
   WebRtcOpus_DecoderInit(dec_state_);
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/include/audio_decoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index 6b0a88ae97..af32a84512 100644
--- a/webrtc/modules/audio_coding/codecs/opus/include/audio_decoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_DECODER_OPUS_H
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_DECODER_OPUS_H
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
 
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 
 namespace webrtc {
 
@@ -48,4 +48,4 @@ class AudioDecoderOpus final : public AudioDecoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_DECODER_OPUS_H
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index eac7412178..707d6c2488 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 
 namespace webrtc {
 
@@ -114,7 +114,7 @@ int AudioEncoderOpus::SampleRateHz() const {
   return kSampleRateHz;
 }
 
-int AudioEncoderOpus::NumChannels() const {
+size_t AudioEncoderOpus::NumChannels() const {
   return config_.num_channels;
 }
 
@@ -132,24 +132,22 @@ int AudioEncoderOpus::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
-  input_buffer_.insert(input_buffer_.end(), audio,
-                       audio + SamplesPer10msFrame());
+  RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
-      (static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
-  RTC_CHECK_EQ(
-      input_buffer_.size(),
-      static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame());
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
-      rtc::CheckedDivExact(input_buffer_.size(),
-                           static_cast<size_t>(config_.num_channels)),
+      rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
       rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded);
   RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
@@ -214,11 +212,11 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps));
 }
 
-int AudioEncoderOpus::Num10msFramesPerPacket() const {
-  return rtc::CheckedDivExact(config_.frame_size_ms, 10);
+size_t AudioEncoderOpus::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
 }
 
-int AudioEncoderOpus::SamplesPer10msFrame() const {
+size_t AudioEncoderOpus::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 7f2b563fd9..59c8f796ee 100644
--- a/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
 
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
@@ -31,7 +31,7 @@ class AudioEncoderOpus final : public AudioEncoder {
   struct Config {
     bool IsOk() const;
     int frame_size_ms = 20;
-    int num_channels = 1;
+    size_t num_channels = 1;
     int payload_type = 120;
     ApplicationMode application = kVoip;
     int bitrate_bps = 64000;
@@ -56,13 +56,13 @@ class AudioEncoderOpus final : public AudioEncoder {
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
 
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
 
@@ -85,8 +85,8 @@ class AudioEncoderOpus final : public AudioEncoder {
   bool dtx_enabled() const { return config_.dtx_enabled; }
 
  private:
-  int Num10msFramesPerPacket() const;
-  int SamplesPer10msFrame() const;
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
   bool RecreateEncoderInstance(const Config& config);
 
   Config config_;
@@ -99,4 +99,4 @@ class AudioEncoderOpus final : public AudioEncoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index e69f259554..441e807b4f 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -12,7 +12,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus.gypi b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
index 05da3e5e47..d7454d632d 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus.gypi
+++ b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
@@ -39,17 +39,14 @@
       'dependencies': [
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        '<(webrtc_root)',
-      ],
       'sources': [
         'audio_decoder_opus.cc',
+        'audio_decoder_opus.h',
         'audio_encoder_opus.cc',
-        'include/audio_decoder_opus.h',
-        'include/audio_encoder_opus.h',
-        'include/opus_interface.h',
+        'audio_encoder_opus.h',
         'opus_inst.h',
         'opus_interface.c',
+        'opus_interface.h',
       ],
     },
   ],
@@ -65,9 +62,6 @@
             '<(webrtc_root)/test/test.gyp:test_support_main',
             '<(DEPTH)/testing/gtest.gyp:gtest',
           ],
-          'include_dirs': [
-            '<(webrtc_root)',
-          ],
           'sources': [
             'opus_fec_test.cc',
           ],
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index f257210431..4f9f7ff7bb 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -9,8 +9,9 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::string;
@@ -21,7 +22,7 @@ using ::testing::TestWithParam;
 namespace webrtc {
 
 // Define coding parameter as <channels, bit_rate, filename, extension>.
-typedef tuple<int, int, string, string> coding_param;
+typedef tuple<size_t, int, string, string> coding_param;
 typedef struct mode mode;
 
 struct mode {
@@ -47,7 +48,7 @@ class OpusFecTest : public TestWithParam<coding_param> {
   int sampling_khz_;
   size_t block_length_sample_;
 
-  int channels_;
+  size_t channels_;
   int bit_rate_;
 
   size_t data_pointer_;
@@ -68,7 +69,7 @@ class OpusFecTest : public TestWithParam<coding_param> {
 void OpusFecTest::SetUp() {
   channels_ = get<0>(GetParam());
   bit_rate_ = get<1>(GetParam());
-  printf("Coding %d channel signal at %d bps.\n", channels_, bit_rate_);
+  printf("Coding %" PRIuS " channel signal at %d bps.\n", channels_, bit_rate_);
 
   in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
index 373db392a6..8d032baf35 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -11,17 +11,26 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
 
+#include <stddef.h>
+
 #include "opus.h"
 
 struct WebRtcOpusEncInst {
   OpusEncoder* encoder;
+  size_t channels;
   int in_dtx_mode;
+  // When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros
+  // to break long zero segment so as to prevent DTX from going wrong. We use
+  // one counter for each channel. After each encoding, |zero_counts| contain
+  // the remaining zeros from the last frame.
+  // TODO(minyue): remove this when Opus gets an internal fix to DTX.
+  size_t* zero_counts;
 };
 
 struct WebRtcOpusDecInst {
   OpusDecoder* decoder;
   int prev_decoded_samples;
-  int channels;
+  size_t channels;
   int in_dtx_mode;
 };
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 1a632422c5..9dc7ef95fe 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 
+#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
@@ -29,48 +30,61 @@ enum {
 
   /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
   kWebRtcOpusDefaultFrameSize = 960,
+
+  // Maximum number of consecutive zeros, beyond or equal to which DTX can fail.
+  kZeroBreakCount = 157,
+
+#if defined(OPUS_FIXED_POINT)
+  kZeroBreakValue = 10,
+#else
+  kZeroBreakValue = 1,
+#endif
 };
 
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
-                                 int32_t channels,
+                                 size_t channels,
                                  int32_t application) {
-  OpusEncInst* state;
-  if (inst != NULL) {
-    state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
-    if (state) {
-      int opus_app;
-      switch (application) {
-        case 0: {
-          opus_app = OPUS_APPLICATION_VOIP;
-          break;
-        }
-        case 1: {
-          opus_app = OPUS_APPLICATION_AUDIO;
-          break;
-        }
-        default: {
-          free(state);
-          return -1;
-        }
-      }
+  int opus_app;
+  if (!inst)
+    return -1;
 
-      int error;
-      state->encoder = opus_encoder_create(48000, channels, opus_app,
-                                           &error);
-      state->in_dtx_mode = 0;
-      if (error == OPUS_OK && state->encoder != NULL) {
-        *inst = state;
-        return 0;
-      }
-      free(state);
-    }
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
   }
-  return -1;
+
+  OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
+  assert(state);
+
+  // Allocate zero counters.
+  state->zero_counts = calloc(channels, sizeof(size_t));
+  assert(state->zero_counts);
+
+  int error;
+  state->encoder = opus_encoder_create(48000, (int)channels, opus_app,
+                                       &error);
+  if (error != OPUS_OK || !state->encoder) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+
+  *inst = state;
+  return 0;
 }
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   if (inst) {
     opus_encoder_destroy(inst->encoder);
+    free(inst->zero_counts);
     free(inst);
     return 0;
   } else {
@@ -84,13 +98,42 @@ int WebRtcOpus_Encode(OpusEncInst* inst,
                       size_t length_encoded_buffer,
                       uint8_t* encoded) {
   int res;
+  size_t i;
+  size_t c;
+
+  int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs];
 
   if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
     return -1;
   }
 
+  const size_t channels = inst->channels;
+  int use_buffer = 0;
+
+  // Break long consecutive zeros by forcing a "1" every |kZeroBreakCount|
+  // samples.
+  if (inst->in_dtx_mode) {
+    for (i = 0; i < samples; ++i) {
+      for (c = 0; c < channels; ++c) {
+        if (audio_in[i * channels + c] == 0) {
+          ++inst->zero_counts[c];
+          if (inst->zero_counts[c] == kZeroBreakCount) {
+            if (!use_buffer) {
+              memcpy(buffer, audio_in, samples * channels * sizeof(int16_t));
+              use_buffer = 1;
+            }
+            buffer[i * channels + c] = kZeroBreakValue;
+            inst->zero_counts[c] = 0;
+          }
+        } else {
+          inst->zero_counts[c] = 0;
+        }
+      }
+    }
+  }
+
   res = opus_encode(inst->encoder,
-                    (const opus_int16*)audio_in,
+                    use_buffer ? buffer : audio_in,
                     (int)samples,
                     encoded,
                     (opus_int32)length_encoded_buffer);
@@ -205,7 +248,7 @@ int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
   }
 }
 
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
   int error;
   OpusDecInst* state;
 
@@ -217,7 +260,7 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
     }
 
     /* Create new memory, always at 48000 Hz. */
-    state->decoder = opus_decoder_create(48000, channels, &error);
+    state->decoder = opus_decoder_create(48000, (int)channels, &error);
     if (error == OPUS_OK && state->decoder != NULL) {
       /* Creation of memory all ok. */
       state->channels = channels;
@@ -246,7 +289,7 @@ int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
   }
 }
 
-int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
   return inst->channels;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
index 50b2338ab5..754b49c808 100644
--- a/webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_OPUS_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_OPUS_INTERFACE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
 
 #include <stddef.h>
 
@@ -43,7 +43,7 @@ typedef struct WebRtcOpusDecInst OpusDecInst;
  *                            -1 - Error
  */
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
-                                 int32_t channels,
+                                 size_t channels,
                                  int32_t application);
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
@@ -195,7 +195,7 @@ int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
  */
 int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
 
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels);
 int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
 
 /****************************************************************************
@@ -203,7 +203,7 @@ int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
  *
  * This function returns the number of channels created for Opus decoder.
  */
-int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst);
 
 /****************************************************************************
  * WebRtcOpus_DecoderInit(...)
@@ -346,4 +346,4 @@ int WebRtcOpus_PacketHasFec(const uint8_t* payload,
 }  // extern "C"
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_OPUS_INCLUDE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
index 29def14bf8..4d1aa42c89 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 using ::std::string;
@@ -77,7 +77,7 @@ float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
   value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
                             &audio_type);
   clocks = clock() - clocks;
-  EXPECT_EQ(output_length_sample_, value);
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index 4630e44807..c82b184b38 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -10,7 +10,8 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -35,17 +36,18 @@ class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
  protected:
   OpusTest();
 
-  void TestDtxEffect(bool dtx);
+  void TestDtxEffect(bool dtx, int block_length_ms);
 
   // Prepare |speech_data_| for encoding, read from a hard-coded file.
   // After preparation, |speech_data_.GetNextBlock()| returns a pointer to a
   // block of |block_length_ms| milliseconds. The data is looped every
   // |loop_length_ms| milliseconds.
-  void PrepareSpeechData(int channel, int block_length_ms, int loop_length_ms);
+  void PrepareSpeechData(size_t channel,
+                         int block_length_ms,
+                         int loop_length_ms);
 
   int EncodeDecode(WebRtcOpusEncInst* encoder,
-                   const int16_t* input_audio,
-                   size_t input_samples,
+                   rtc::ArrayView<const int16_t> input_audio,
                    WebRtcOpusDecInst* decoder,
                    int16_t* output_audio,
                    int16_t* audio_type);
@@ -53,13 +55,16 @@ class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
   void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
                           opus_int32 expect, int32_t set);
 
+  void CheckAudioBounded(const int16_t* audio, size_t samples, size_t channels,
+                         uint16_t bound) const;
+
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
 
   AudioLoop speech_data_;
   uint8_t bitstream_[kMaxBytes];
   size_t encoded_bytes_;
-  int channels_;
+  size_t channels_;
   int application_;
 };
 
@@ -67,11 +72,11 @@ OpusTest::OpusTest()
     : opus_encoder_(NULL),
       opus_decoder_(NULL),
       encoded_bytes_(0),
-      channels_(::testing::get<0>(GetParam())),
+      channels_(static_cast<size_t>(::testing::get<0>(GetParam()))),
       application_(::testing::get<1>(GetParam())) {
 }
 
-void OpusTest::PrepareSpeechData(int channel, int block_length_ms,
+void OpusTest::PrepareSpeechData(size_t channel, int block_length_ms,
                                  int loop_length_ms) {
   const std::string file_name =
         webrtc::test::ResourcePath((channel == 1) ?
@@ -95,14 +100,25 @@ void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
   EXPECT_EQ(expect, bandwidth);
 }
 
+void OpusTest::CheckAudioBounded(const int16_t* audio, size_t samples,
+                                 size_t channels, uint16_t bound) const {
+  for (size_t i = 0; i < samples; ++i) {
+    for (size_t c = 0; c < channels; ++c) {
+      ASSERT_GE(audio[i * channels + c], -bound);
+      ASSERT_LE(audio[i * channels + c], bound);
+    }
+  }
+}
+
 int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
-                           const int16_t* input_audio,
-                           size_t input_samples,
+                           rtc::ArrayView<const int16_t> input_audio,
                            WebRtcOpusDecInst* decoder,
                            int16_t* output_audio,
                            int16_t* audio_type) {
-  int encoded_bytes_int = WebRtcOpus_Encode(encoder, input_audio, input_samples,
-                                            kMaxBytes, bitstream_);
+  int encoded_bytes_int = WebRtcOpus_Encode(
+      encoder, input_audio.data(),
+      rtc::CheckedDivExact(input_audio.size(), channels_),
+      kMaxBytes, bitstream_);
   EXPECT_GE(encoded_bytes_int, 0);
   encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
   int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
@@ -115,8 +131,9 @@ int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
 
 // Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
 // they should not. This test is signal dependent.
-void OpusTest::TestDtxEffect(bool dtx) {
-  PrepareSpeechData(channels_, 20, 2000);
+void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
+  PrepareSpeechData(channels_, block_length_ms, 2000);
+  const size_t samples = kOpusRateKhz * block_length_ms;
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
@@ -129,22 +146,20 @@ void OpusTest::TestDtxEffect(bool dtx) {
                                      channels_ == 1 ? 32000 : 64000));
 
   // Set input audio as silence.
-  int16_t* silence = new int16_t[kOpus20msFrameSamples * channels_];
-  memset(silence, 0, sizeof(int16_t) * kOpus20msFrameSamples * channels_);
+  std::vector<int16_t> silence(samples * channels_, 0);
 
   // Setting DTX.
   EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) :
       WebRtcOpus_DisableDtx(opus_encoder_));
 
   int16_t audio_type;
-  int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
+  int16_t* output_data_decode = new int16_t[samples * channels_];
 
   for (int i = 0; i < 100; ++i) {
-    EXPECT_EQ(kOpus20msFrameSamples,
+    EXPECT_EQ(samples,
               static_cast<size_t>(EncodeDecode(
-                  opus_encoder_, speech_data_.GetNextBlock(),
-                  kOpus20msFrameSamples, opus_decoder_, output_data_decode,
-                  &audio_type)));
+                  opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+                  output_data_decode, &audio_type)));
     // If not DTX, it should never enter DTX mode. If DTX, we do not care since
     // whether it enters DTX depends on the signal type.
     if (!dtx) {
@@ -158,10 +173,10 @@ void OpusTest::TestDtxEffect(bool dtx) {
   // We input some silent segments. In DTX mode, the encoder will stop sending.
   // However, DTX may happen after a while.
   for (int i = 0; i < 30; ++i) {
-    EXPECT_EQ(kOpus20msFrameSamples,
+    EXPECT_EQ(samples,
               static_cast<size_t>(EncodeDecode(
-                  opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
-                  output_data_decode, &audio_type)));
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
     if (!dtx) {
       EXPECT_GT(encoded_bytes_, 1U);
       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -177,21 +192,47 @@ void OpusTest::TestDtxEffect(bool dtx) {
 
   // When Opus is in DTX, it wakes up in a regular basis. It sends two packets,
   // one with an arbitrary size and the other of 1-byte, then stops sending for
-  // 19 frames.
-  const int cycles = 5;
-  for (int j = 0; j < cycles; ++j) {
-    // DTX mode is maintained 19 frames.
-    for (int i = 0; i < 19; ++i) {
-      EXPECT_EQ(kOpus20msFrameSamples,
+  // a certain number of frames.
+
+  // |max_dtx_frames| is the maximum number of frames Opus can stay in DTX.
+  const int max_dtx_frames = 400 / block_length_ms + 1;
+
+  // We run |kRunTimeMs| milliseconds of pure silence.
+  const int kRunTimeMs = 2000;
+
+  // We check that, after a |kCheckTimeMs| milliseconds (given that the CNG in
+  // Opus needs time to adapt), the absolute values of DTX decoded signal are
+  // bounded by |kOutputValueBound|.
+  const int kCheckTimeMs = 1500;
+
+#if defined(OPUS_FIXED_POINT)
+  const uint16_t kOutputValueBound = 20;
+#else
+  const uint16_t kOutputValueBound = 2;
+#endif
+
+  int time = 0;
+  while (time < kRunTimeMs) {
+    // DTX mode is maintained for maximum |max_dtx_frames| frames.
+    int i = 0;
+    for (; i < max_dtx_frames; ++i) {
+      time += block_length_ms;
+      EXPECT_EQ(samples,
                 static_cast<size_t>(EncodeDecode(
-                    opus_encoder_, silence, kOpus20msFrameSamples,
-                    opus_decoder_, output_data_decode, &audio_type)));
+                    opus_encoder_, silence, opus_decoder_, output_data_decode,
+                    &audio_type)));
       if (dtx) {
+        if (encoded_bytes_ > 1)
+          break;
         EXPECT_EQ(0U, encoded_bytes_)  // Send 0 byte.
             << "Opus should have entered DTX mode.";
         EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
         EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
         EXPECT_EQ(2, audio_type);  // Comfort noise.
+        if (time >= kCheckTimeMs) {
+          CheckAudioBounded(output_data_decode, samples, channels_,
+                            kOutputValueBound);
+        }
       } else {
         EXPECT_GT(encoded_bytes_, 1U);
         EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -200,27 +241,31 @@ void OpusTest::TestDtxEffect(bool dtx) {
       }
     }
 
-    // Quit DTX after 19 frames.
-    EXPECT_EQ(kOpus20msFrameSamples,
-              static_cast<size_t>(EncodeDecode(
-                  opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
-                  output_data_decode, &audio_type)));
+    if (dtx) {
+      // With DTX, Opus must stop transmission for some time.
+      EXPECT_GT(i, 1);
+    }
 
-    EXPECT_GT(encoded_bytes_, 1U);
+    // We expect a normal payload.
     EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
     EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
     EXPECT_EQ(0, audio_type);  // Speech.
 
     // Enters DTX again immediately.
-    EXPECT_EQ(kOpus20msFrameSamples,
+    time += block_length_ms;
+    EXPECT_EQ(samples,
               static_cast<size_t>(EncodeDecode(
-                  opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
-                  output_data_decode, &audio_type)));
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
     if (dtx) {
       EXPECT_EQ(1U, encoded_bytes_);  // Send 1 byte.
       EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
       EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
       EXPECT_EQ(2, audio_type);  // Comfort noise.
+      if (time >= kCheckTimeMs) {
+        CheckAudioBounded(output_data_decode, samples, channels_,
+                          kOutputValueBound);
+      }
     } else {
       EXPECT_GT(encoded_bytes_, 1U);
       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -232,10 +277,10 @@ void OpusTest::TestDtxEffect(bool dtx) {
   silence[0] = 10000;
   if (dtx) {
     // Verify that encoder/decoder can jump out from DTX mode.
-    EXPECT_EQ(kOpus20msFrameSamples,
+    EXPECT_EQ(samples,
               static_cast<size_t>(EncodeDecode(
-                  opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
-                  output_data_decode, &audio_type)));
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
     EXPECT_GT(encoded_bytes_, 1U);
     EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
     EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
@@ -244,7 +289,6 @@ void OpusTest::TestDtxEffect(bool dtx) {
 
   // Free memory.
   delete[] output_data_decode;
-  delete[] silence;
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }
@@ -314,10 +358,9 @@ TEST_P(OpusTest, OpusEncodeDecode) {
   int16_t audio_type;
   int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
   EXPECT_EQ(kOpus20msFrameSamples,
-            static_cast<size_t>(EncodeDecode(
-                opus_encoder_, speech_data_.GetNextBlock(),
-                kOpus20msFrameSamples, opus_decoder_, output_data_decode,
-                &audio_type)));
+            static_cast<size_t>(
+                EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                             opus_decoder_, output_data_decode, &audio_type)));
 
   // Free memory.
   delete[] output_data_decode;
@@ -374,10 +417,9 @@ TEST_P(OpusTest, OpusDecodeInit) {
   int16_t audio_type;
   int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
   EXPECT_EQ(kOpus20msFrameSamples,
-            static_cast<size_t>(EncodeDecode(
-                opus_encoder_, speech_data_.GetNextBlock(),
-                kOpus20msFrameSamples, opus_decoder_, output_data_decode,
-                &audio_type)));
+            static_cast<size_t>(
+                EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                             opus_decoder_, output_data_decode, &audio_type)));
 
   WebRtcOpus_DecoderInit(opus_decoder_);
 
@@ -444,11 +486,15 @@ TEST_P(OpusTest, OpusEnableDisableDtx) {
 }
 
 TEST_P(OpusTest, OpusDtxOff) {
-  TestDtxEffect(false);
+  TestDtxEffect(false, 10);
+  TestDtxEffect(false, 20);
+  TestDtxEffect(false, 40);
 }
 
 TEST_P(OpusTest, OpusDtxOn) {
-  TestDtxEffect(true);
+  TestDtxEffect(true, 10);
+  TestDtxEffect(true, 20);
+  TestDtxEffect(true, 40);
 }
 
 TEST_P(OpusTest, OpusSetPacketLossRate) {
@@ -513,10 +559,9 @@ TEST_P(OpusTest, OpusDecodePlc) {
   int16_t audio_type;
   int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
   EXPECT_EQ(kOpus20msFrameSamples,
-            static_cast<size_t>(EncodeDecode(
-                opus_encoder_, speech_data_.GetNextBlock(),
-                kOpus20msFrameSamples, opus_decoder_, output_data_decode,
-                &audio_type)));
+            static_cast<size_t>(
+                EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                             opus_decoder_, output_data_decode, &audio_type)));
 
   // Call decoder PLC.
   int16_t* plc_buffer = new int16_t[kOpus20msFrameSamples * channels_];
@@ -542,10 +587,11 @@ TEST_P(OpusTest, OpusDurationEstimation) {
   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
 
   // 10 ms. We use only first 10 ms of a 20 ms block.
-  int encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
-                                            speech_data_.GetNextBlock(),
-                                            kOpus10msFrameSamples,
-                                            kMaxBytes, bitstream_);
+  auto speech_block = speech_data_.GetNextBlock();
+  int encoded_bytes_int = WebRtcOpus_Encode(
+      opus_encoder_, speech_block.data(),
+      rtc::CheckedDivExact(speech_block.size(), 2 * channels_),
+      kMaxBytes, bitstream_);
   EXPECT_GE(encoded_bytes_int, 0);
   EXPECT_EQ(kOpus10msFrameSamples,
             static_cast<size_t>(WebRtcOpus_DurationEst(
@@ -553,10 +599,11 @@ TEST_P(OpusTest, OpusDurationEstimation) {
                 static_cast<size_t>(encoded_bytes_int))));
 
   // 20 ms
-  encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
-                                        speech_data_.GetNextBlock(),
-                                        kOpus20msFrameSamples,
-                                        kMaxBytes, bitstream_);
+  speech_block = speech_data_.GetNextBlock();
+  encoded_bytes_int = WebRtcOpus_Encode(
+      opus_encoder_, speech_block.data(),
+      rtc::CheckedDivExact(speech_block.size(), channels_),
+      kMaxBytes, bitstream_);
   EXPECT_GE(encoded_bytes_int, 0);
   EXPECT_EQ(kOpus20msFrameSamples,
             static_cast<size_t>(WebRtcOpus_DurationEst(
@@ -594,10 +641,11 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
   OpusRepacketizer* rp = opus_repacketizer_create();
 
   for (int idx = 0; idx < kPackets; idx++) {
-    encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
-                                       speech_data_.GetNextBlock(),
-                                       kOpus20msFrameSamples, kMaxBytes,
-                                       bitstream_);
+    auto speech_block = speech_data_.GetNextBlock();
+    encoded_bytes_ =
+        WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                          rtc::CheckedDivExact(speech_block.size(), channels_),
+                          kMaxBytes, bitstream_);
     EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_));
   }
 
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index 7d07b23a3c..834c070073 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_decoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_decoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
index 96131c4d21..692cb94282 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_decoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_DECODER_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_DECODER_PCM16B_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
@@ -37,4 +37,4 @@ class AudioDecoderPcm16B final : public AudioDecoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_DECODER_PCM16B_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
index 6c30c7ff62..f4d4022302 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 
 namespace webrtc {
 
@@ -22,7 +22,7 @@ size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
   return WebRtcPcm16b_Encode(audio, input_len, encoded);
 }
 
-int AudioEncoderPcm16B::BytesPerSample() const {
+size_t AudioEncoderPcm16B::BytesPerSample() const {
   return 2;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index e03da213df..68ca2da77e 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_ENCODER_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_ENCODER_PCM16B_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 namespace webrtc {
 
@@ -37,12 +37,12 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm {
                     size_t input_len,
                     uint8_t* encoded) override;
 
-  int BytesPerSample() const override;
+  size_t BytesPerSample() const override;
 
-private:
+ private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_ENCODER_PCM16B_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi b/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
index 3dc2f772c1..d0dd21bb60 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
@@ -15,23 +15,13 @@
         'audio_encoder_interface',
         'g711',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'include/audio_decoder_pcm16b.h',
-        'include/audio_encoder_pcm16b.h',
-        'include/pcm16b.h',
         'audio_decoder_pcm16b.cc',
+        'audio_decoder_pcm16b.h',
         'audio_encoder_pcm16b.cc',
+        'audio_encoder_pcm16b.h',
         'pcm16b.c',
+        'pcm16b.h',
       ],
     },
   ], # targets
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
index d86a65db49..f96e741c46 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INCLUDE_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INCLUDE_PCM16B_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
 /*
  * Define the fixpoint numeric formats
  */
@@ -65,4 +65,4 @@ size_t WebRtcPcm16b_Decode(const uint8_t* encoded,
 }
 #endif
 
-#endif /* PCM16B */
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_ */
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index a19d194e59..7ef1ce096b 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -32,7 +32,7 @@ int AudioEncoderCopyRed::SampleRateHz() const {
   return speech_encoder_->SampleRateHz();
 }
 
-int AudioEncoderCopyRed::NumChannels() const {
+size_t AudioEncoderCopyRed::NumChannels() const {
   return speech_encoder_->NumChannels();
 }
 
@@ -54,12 +54,11 @@ int AudioEncoderCopyRed::GetTargetBitrate() const {
 
 AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  EncodedInfo info = speech_encoder_->Encode(
-      rtp_timestamp, audio, static_cast<size_t>(SampleRateHz() / 100),
-      max_encoded_bytes, encoded);
+  EncodedInfo info =
+      speech_encoder_->Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
   RTC_CHECK_GE(max_encoded_bytes,
                info.encoded_bytes + secondary_info_.encoded_bytes);
   RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index 7837010605..2f53765389 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -38,13 +38,13 @@ class AudioEncoderCopyRed final : public AudioEncoder {
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index cb50652183..22601b6597 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -42,7 +42,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
     config.speech_encoder = &mock_encoder_;
     red_.reset(new AudioEncoderCopyRed(config));
     memset(audio_, 0, sizeof(audio_));
-    EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
+    EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
     EXPECT_CALL(mock_encoder_, SampleRateHz())
         .WillRepeatedly(Return(sample_rate_hz_));
     EXPECT_CALL(mock_encoder_, MaxEncodedBytes())
@@ -60,8 +60,10 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
 
   void Encode() {
     ASSERT_TRUE(red_.get() != NULL);
-    encoded_info_ = red_->Encode(timestamp_, audio_, num_audio_samples_10ms,
-                                 encoded_.size(), &encoded_[0]);
+    encoded_info_ = red_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
+        encoded_.size(), &encoded_[0]);
     timestamp_ += num_audio_samples_10ms;
   }
 
@@ -83,7 +85,7 @@ class MockEncodeHelper {
   }
 
   AudioEncoder::EncodedInfo Encode(uint32_t timestamp,
-                                   const int16_t* audio,
+                                   rtc::ArrayView<const int16_t> audio,
                                    size_t max_encoded_bytes,
                                    uint8_t* encoded) {
     if (write_payload_) {
@@ -108,8 +110,8 @@ TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
-  EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17));
-  EXPECT_EQ(17, red_->NumChannels());
+  EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->NumChannels());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index 3395721f8b..3dc665482a 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::tr1::get;
@@ -23,8 +24,10 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
     : block_duration_ms_(block_duration_ms),
       input_sampling_khz_(input_sampling_khz),
       output_sampling_khz_(output_sampling_khz),
-      input_length_sample_(block_duration_ms_ * input_sampling_khz_),
-      output_length_sample_(block_duration_ms_ * output_sampling_khz_),
+      input_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
+      output_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
       data_pointer_(0),
       loop_length_samples_(0),
       max_bytes_(0),
@@ -65,8 +68,7 @@ void AudioCodecSpeedTest::SetUp() {
   memcpy(&in_data_[loop_length_samples_], &in_data_[0],
          input_length_sample_ * channels_ * sizeof(int16_t));
 
-  max_bytes_ =
-      static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
+  max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
   out_data_.reset(new int16_t[output_length_sample_ * channels_]);
   bit_stream_.reset(new uint8_t[max_bytes_]);
 
@@ -98,7 +100,7 @@ void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
   size_t time_now_ms = 0;
   float time_ms;
 
-  printf("Coding %d kHz-sampled %d-channel audio at %d bps ...\n",
+  printf("Coding %d kHz-sampled %" PRIuS "-channel audio at %d bps ...\n",
          input_sampling_khz_, channels_, bit_rate_);
 
   while (time_now_ms < audio_duration_sec * 1000) {
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index 2736c2912e..fb7b3e5b1e 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -20,7 +20,8 @@ namespace webrtc {
 
 // Define coding parameter as
 // <channels, bit_rate, file_name, extension, if_save_output>.
-typedef std::tr1::tuple<int, int, std::string, std::string, bool> coding_param;
+typedef std::tr1::tuple<size_t, int, std::string, std::string, bool>
+    coding_param;
 
 class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
  protected:
@@ -55,10 +56,10 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   int output_sampling_khz_;
 
   // Number of samples-per-channel in a frame.
-  int input_length_sample_;
+  size_t input_length_sample_;
 
   // Expected output number of samples-per-channel in a frame.
-  int output_length_sample_;
+  size_t output_length_sample_;
 
   rtc::scoped_ptr<int16_t[]> in_data_;
   rtc::scoped_ptr<int16_t[]> out_data_;
@@ -74,7 +75,7 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   float decoding_time_ms_;
   FILE* out_file_;
 
-  int channels_;
+  size_t channels_;
 
   // Bit rate is in bit-per-second.
   int bit_rate_;
diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index b145cf423e..9e7991f22f 100644
--- a/webrtc/modules/audio_coding/main/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -8,16 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
 
+#include <string>
 #include <vector>
 
+#include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
@@ -60,7 +61,11 @@ class AudioCodingModule {
 
  public:
   struct Config {
-    Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {}
+    Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
+      // Post-decode VAD is disabled by default in NetEq, however, Audio
+      // Conference Mixer relies on VAD decisions and fails without them.
+      neteq_config.enable_post_decode_vad = true;
+    }
 
     int id;
     NetEq::Config neteq_config;
@@ -129,7 +134,7 @@ class AudioCodingModule {
   //    0 if succeeded.
   //
   static int Codec(const char* payload_name, CodecInst* codec,
-                       int sampling_freq_hz, int channels);
+                   int sampling_freq_hz, size_t channels);
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t Codec()
@@ -148,7 +153,7 @@ class AudioCodingModule {
   //   -1 if the codec is not found.
   //
   static int Codec(const char* payload_name, int sampling_freq_hz,
-                             int channels);
+                   size_t channels);
 
   ///////////////////////////////////////////////////////////////////////////
   // bool IsCodecValid()
@@ -206,14 +211,10 @@ class AudioCodingModule {
   // int32_t SendCodec()
   // Get parameters for the codec currently registered as send codec.
   //
-  // Output:
-  //   -current_send_codec          : parameters of the send codec.
-  //
   // Return value:
-  //   -1 if failed to get send codec,
-  //    0 if succeeded.
+  //   The send codec, or nothing if we don't have one
   //
-  virtual int32_t SendCodec(CodecInst* current_send_codec) const = 0;
+  virtual rtc::Optional<CodecInst> SendCodec() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SendFrequency()
@@ -471,10 +472,14 @@ class AudioCodingModule {
   //
   virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
 
+  // Registers an external decoder. The name is only used to provide information
+  // back to the caller about the decoder. Hence, the name is arbitrary, and may
+  // be empty.
   virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
                                            AudioDecoder* external_decoder,
                                            int sample_rate_hz,
-                                           int num_channels) = 0;
+                                           int num_channels,
+                                           const std::string& name) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t UnregisterReceiveCodec()
@@ -705,23 +710,6 @@ class AudioCodingModule {
       NetworkStatistics* network_statistics) = 0;
 
   //
-  // Set an initial delay for playout.
-  // An initial delay yields ACM playout silence until equivalent of |delay_ms|
-  // audio payload is accumulated in NetEq jitter. Thereafter, ACM pulls audio
-  // from NetEq in its regular fashion, and the given delay is maintained
-  // through out the call, unless channel conditions yield to a higher jitter
-  // buffer delay.
-  //
-  // Input:
-  //   -delay_ms           : delay in milliseconds.
-  //
-  // Return values:
-  //   -1 if failed to set the delay.
-  //    0 if delay is set successfully.
-  //
-  virtual int SetInitialPlayoutDelay(int delay_ms) = 0;
-
-  //
   // Enable NACK and set the maximum size of the NACK list. If NACK is already
   // enable then the maximum NACK list size is modified accordingly.
   //
@@ -755,4 +743,4 @@ class AudioCodingModule {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h b/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
index 489df406f4..280d6bffa2 100644
--- a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
 
 #include <map>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -48,4 +48,4 @@ enum OpusApplicationMode {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/OWNERS b/webrtc/modules/audio_coding/main/acm2/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/webrtc/modules/audio_coding/main/acm2/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc b/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
deleted file mode 100644
index f9b77e8985..0000000000
--- a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
+++ /dev/null
@@ -1,465 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-namespace acm2 {
-
-namespace {
-bool IsCodecRED(const CodecInst& codec) {
-  return (STR_CASE_CMP(codec.plname, "RED") == 0);
-}
-
-bool IsCodecCN(const CodecInst& codec) {
-  return (STR_CASE_CMP(codec.plname, "CN") == 0);
-}
-
-// Check if the given codec is a valid to be registered as send codec.
-int IsValidSendCodec(const CodecInst& send_codec, bool is_primary_encoder) {
-  int dummy_id = 0;
-  if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "Wrong number of channels (%d, only mono and stereo are "
-                 "supported) for %s encoder",
-                 send_codec.channels,
-                 is_primary_encoder ? "primary" : "secondary");
-    return -1;
-  }
-
-  int codec_id = ACMCodecDB::CodecNumber(send_codec);
-  if (codec_id < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "Invalid codec setting for the send codec.");
-    return -1;
-  }
-
-  // TODO(tlegrand): Remove this check. Already taken care of in
-  // ACMCodecDB::CodecNumber().
-  // Check if the payload-type is valid
-  if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "Invalid payload-type %d for %s.", send_codec.pltype,
-                 send_codec.plname);
-    return -1;
-  }
-
-  // Telephone-event cannot be a send codec.
-  if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "telephone-event cannot be a send codec");
-    return -1;
-  }
-
-  if (ACMCodecDB::codec_settings_[codec_id].channel_support <
-      send_codec.channels) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "%d number of channels not supportedn for %s.",
-                 send_codec.channels, send_codec.plname);
-    return -1;
-  }
-
-  if (!is_primary_encoder) {
-    // If registering the secondary encoder, then RED and CN are not valid
-    // choices as encoder.
-    if (IsCodecRED(send_codec)) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "RED cannot be secondary codec");
-      return -1;
-    }
-
-    if (IsCodecCN(send_codec)) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "DTX cannot be secondary codec");
-      return -1;
-    }
-  }
-  return codec_id;
-}
-
-bool IsIsac(const CodecInst& codec) {
-  return
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-      !STR_CASE_CMP(codec.plname, "isac") ||
-#endif
-      false;
-}
-
-bool IsOpus(const CodecInst& codec) {
-  return
-#ifdef WEBRTC_CODEC_OPUS
-      !STR_CASE_CMP(codec.plname, "opus") ||
-#endif
-      false;
-}
-
-bool IsPcmU(const CodecInst& codec) {
-  return !STR_CASE_CMP(codec.plname, "pcmu");
-}
-
-bool IsPcmA(const CodecInst& codec) {
-  return !STR_CASE_CMP(codec.plname, "pcma");
-}
-
-bool IsPcm16B(const CodecInst& codec) {
-  return !STR_CASE_CMP(codec.plname, "l16");
-}
-
-bool IsIlbc(const CodecInst& codec) {
-  return
-#ifdef WEBRTC_CODEC_ILBC
-      !STR_CASE_CMP(codec.plname, "ilbc") ||
-#endif
-      false;
-}
-
-bool IsG722(const CodecInst& codec) {
-  return
-#ifdef WEBRTC_CODEC_G722
-      !STR_CASE_CMP(codec.plname, "g722") ||
-#endif
-      false;
-}
-
-bool CodecSupported(const CodecInst& codec) {
-  return IsOpus(codec) || IsPcmU(codec) || IsPcmA(codec) || IsPcm16B(codec) ||
-         IsIlbc(codec) || IsG722(codec) || IsIsac(codec);
-}
-
-const CodecInst kEmptyCodecInst = {-1, "noCodecRegistered", 0, 0, 0, 0};
-}  // namespace
-
-CodecManager::CodecManager()
-    : cng_nb_pltype_(255),
-      cng_wb_pltype_(255),
-      cng_swb_pltype_(255),
-      cng_fb_pltype_(255),
-      red_nb_pltype_(255),
-      stereo_send_(false),
-      dtx_enabled_(false),
-      vad_mode_(VADNormal),
-      send_codec_inst_(kEmptyCodecInst),
-      red_enabled_(false),
-      codec_fec_enabled_(false),
-      encoder_is_opus_(false) {
-  // Register the default payload type for RED and for CNG at sampling rates of
-  // 8, 16, 32 and 48 kHz.
-  for (const CodecInst& ci : RentACodec::Database()) {
-    if (IsCodecRED(ci) && ci.plfreq == 8000) {
-      red_nb_pltype_ = static_cast<uint8_t>(ci.pltype);
-    } else if (IsCodecCN(ci)) {
-      if (ci.plfreq == 8000) {
-        cng_nb_pltype_ = static_cast<uint8_t>(ci.pltype);
-      } else if (ci.plfreq == 16000) {
-        cng_wb_pltype_ = static_cast<uint8_t>(ci.pltype);
-      } else if (ci.plfreq == 32000) {
-        cng_swb_pltype_ = static_cast<uint8_t>(ci.pltype);
-      } else if (ci.plfreq == 48000) {
-        cng_fb_pltype_ = static_cast<uint8_t>(ci.pltype);
-      }
-    }
-  }
-  thread_checker_.DetachFromThread();
-}
-
-CodecManager::~CodecManager() = default;
-
-int CodecManager::RegisterEncoder(const CodecInst& send_codec) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  int codec_id = IsValidSendCodec(send_codec, true);
-
-  // Check for reported errors from function IsValidSendCodec().
-  if (codec_id < 0) {
-    return -1;
-  }
-
-  int dummy_id = 0;
-  // RED can be registered with other payload type. If not registered a default
-  // payload type is used.
-  if (IsCodecRED(send_codec)) {
-    // TODO(tlegrand): Remove this check. Already taken care of in
-    // ACMCodecDB::CodecNumber().
-    // Check if the payload-type is valid
-    if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "Invalid payload-type %d for %s.", send_codec.pltype,
-                   send_codec.plname);
-      return -1;
-    }
-    // Set RED payload type.
-    if (send_codec.plfreq == 8000) {
-      red_nb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-    } else {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "RegisterSendCodec() failed, invalid frequency for RED "
-                   "registration");
-      return -1;
-    }
-    return 0;
-  }
-
-  // CNG can be registered with other payload type. If not registered the
-  // default payload types from codec database will be used.
-  if (IsCodecCN(send_codec)) {
-    // CNG is registered.
-    switch (send_codec.plfreq) {
-      case 8000: {
-        cng_nb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        return 0;
-      }
-      case 16000: {
-        cng_wb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        return 0;
-      }
-      case 32000: {
-        cng_swb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        return 0;
-      }
-      case 48000: {
-        cng_fb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        return 0;
-      }
-      default: {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "RegisterSendCodec() failed, invalid frequency for CNG "
-                     "registration");
-        return -1;
-      }
-    }
-  }
-
-  // Set Stereo, and make sure VAD and DTX is turned off.
-  if (send_codec.channels == 2) {
-    stereo_send_ = true;
-    if (dtx_enabled_) {
-      WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, dummy_id,
-                   "VAD/DTX is turned off, not supported when sending stereo.");
-    }
-    dtx_enabled_ = false;
-  } else {
-    stereo_send_ = false;
-  }
-
-  // Check if the codec is already registered as send codec.
-  bool new_codec = true;
-  if (codec_owner_.Encoder()) {
-    int new_codec_id = ACMCodecDB::CodecNumber(send_codec_inst_);
-    RTC_DCHECK_GE(new_codec_id, 0);
-    new_codec = new_codec_id != codec_id;
-  }
-
-  if (RedPayloadType(send_codec.plfreq) == -1) {
-    red_enabled_ = false;
-  }
-
-  encoder_is_opus_ = IsOpus(send_codec);
-
-  if (new_codec) {
-    // This is a new codec. Register it and return.
-    RTC_DCHECK(CodecSupported(send_codec));
-    if (IsOpus(send_codec)) {
-      // VAD/DTX not supported.
-      dtx_enabled_ = false;
-    }
-    if (!codec_owner_.SetEncoders(
-            send_codec, dtx_enabled_ ? CngPayloadType(send_codec.plfreq) : -1,
-            vad_mode_, red_enabled_ ? RedPayloadType(send_codec.plfreq) : -1))
-      return -1;
-    RTC_DCHECK(codec_owner_.Encoder());
-
-    codec_fec_enabled_ = codec_fec_enabled_ &&
-                         codec_owner_.Encoder()->SetFec(codec_fec_enabled_);
-
-    send_codec_inst_ = send_codec;
-    return 0;
-  }
-
-  // This is an existing codec; re-create it if any parameters have changed.
-  if (send_codec_inst_.plfreq != send_codec.plfreq ||
-      send_codec_inst_.pacsize != send_codec.pacsize ||
-      send_codec_inst_.channels != send_codec.channels) {
-    if (!codec_owner_.SetEncoders(
-            send_codec, dtx_enabled_ ? CngPayloadType(send_codec.plfreq) : -1,
-            vad_mode_, red_enabled_ ? RedPayloadType(send_codec.plfreq) : -1))
-      return -1;
-    RTC_DCHECK(codec_owner_.Encoder());
-  }
-  send_codec_inst_.plfreq = send_codec.plfreq;
-  send_codec_inst_.pacsize = send_codec.pacsize;
-  send_codec_inst_.channels = send_codec.channels;
-  send_codec_inst_.pltype = send_codec.pltype;
-
-  // Check if a change in Rate is required.
-  if (send_codec.rate != send_codec_inst_.rate) {
-    codec_owner_.Encoder()->SetTargetBitrate(send_codec.rate);
-    send_codec_inst_.rate = send_codec.rate;
-  }
-
-  codec_fec_enabled_ =
-      codec_fec_enabled_ && codec_owner_.Encoder()->SetFec(codec_fec_enabled_);
-
-  return 0;
-}
-
-void CodecManager::RegisterEncoder(AudioEncoder* external_speech_encoder) {
-  // Make up a CodecInst.
-  send_codec_inst_.channels = external_speech_encoder->NumChannels();
-  send_codec_inst_.plfreq = external_speech_encoder->SampleRateHz();
-  send_codec_inst_.pacsize = rtc::CheckedDivExact(
-      static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
-                       send_codec_inst_.plfreq),
-      100);
-  send_codec_inst_.pltype = -1;  // Not valid.
-  send_codec_inst_.rate = -1;    // Not valid.
-  static const char kName[] = "external";
-  memcpy(send_codec_inst_.plname, kName, sizeof(kName));
-
-  if (stereo_send_)
-    dtx_enabled_ = false;
-  codec_fec_enabled_ =
-      codec_fec_enabled_ && codec_owner_.Encoder()->SetFec(codec_fec_enabled_);
-  int cng_pt = dtx_enabled_
-                   ? CngPayloadType(external_speech_encoder->SampleRateHz())
-                   : -1;
-  int red_pt = red_enabled_ ? RedPayloadType(send_codec_inst_.plfreq) : -1;
-  codec_owner_.SetEncoders(external_speech_encoder, cng_pt, vad_mode_, red_pt);
-}
-
-int CodecManager::GetCodecInst(CodecInst* current_codec) const {
-  int dummy_id = 0;
-  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, dummy_id,
-               "SendCodec()");
-
-  if (!codec_owner_.Encoder()) {
-    WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, dummy_id,
-                 "SendCodec Failed, no codec is registered");
-    return -1;
-  }
-  *current_codec = send_codec_inst_;
-  return 0;
-}
-
-bool CodecManager::SetCopyRed(bool enable) {
-  if (enable && codec_fec_enabled_) {
-    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
-                 "Codec internal FEC and RED cannot be co-enabled.");
-    return false;
-  }
-  if (enable && RedPayloadType(send_codec_inst_.plfreq) == -1) {
-    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
-                 "Cannot enable RED at %i Hz.", send_codec_inst_.plfreq);
-    return false;
-  }
-  if (red_enabled_ != enable) {
-    red_enabled_ = enable;
-    if (codec_owner_.Encoder()) {
-      int cng_pt = dtx_enabled_ ? CngPayloadType(send_codec_inst_.plfreq) : -1;
-      int red_pt = red_enabled_ ? RedPayloadType(send_codec_inst_.plfreq) : -1;
-      codec_owner_.ChangeCngAndRed(cng_pt, vad_mode_, red_pt);
-    }
-  }
-  return true;
-}
-
-int CodecManager::SetVAD(bool enable, ACMVADMode mode) {
-  // Sanity check of the mode.
-  RTC_DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
-             mode == VADVeryAggr);
-
-  // Check that the send codec is mono. We don't support VAD/DTX for stereo
-  // sending.
-  if (enable && stereo_send_) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "VAD/DTX not supported for stereo sending");
-    dtx_enabled_ = false;
-    return -1;
-  }
-
-  // If a send codec is registered, set VAD/DTX for the codec.
-  if (IsOpus(send_codec_inst_)) {
-    // VAD/DTX not supported.
-    dtx_enabled_ = false;
-    return 0;
-  }
-
-  if (dtx_enabled_ != enable || vad_mode_ != mode) {
-    dtx_enabled_ = enable;
-    vad_mode_ = mode;
-    if (codec_owner_.Encoder()) {
-      int cng_pt = dtx_enabled_ ? CngPayloadType(send_codec_inst_.plfreq) : -1;
-      int red_pt = red_enabled_ ? RedPayloadType(send_codec_inst_.plfreq) : -1;
-      codec_owner_.ChangeCngAndRed(cng_pt, vad_mode_, red_pt);
-    }
-  }
-  return 0;
-}
-
-void CodecManager::VAD(bool* dtx_enabled,
-                       bool* vad_enabled,
-                       ACMVADMode* mode) const {
-  *dtx_enabled = dtx_enabled_;
-  *vad_enabled = dtx_enabled_;
-  *mode = vad_mode_;
-}
-
-int CodecManager::SetCodecFEC(bool enable_codec_fec) {
-  if (enable_codec_fec == true && red_enabled_ == true) {
-    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
-                 "Codec internal FEC and RED cannot be co-enabled.");
-    return -1;
-  }
-
-  RTC_CHECK(codec_owner_.Encoder());
-  codec_fec_enabled_ =
-      codec_owner_.Encoder()->SetFec(enable_codec_fec) && enable_codec_fec;
-  return codec_fec_enabled_ == enable_codec_fec ? 0 : -1;
-}
-
-AudioDecoder* CodecManager::GetAudioDecoder(const CodecInst& codec) {
-  return IsIsac(codec) ? codec_owner_.GetIsacDecoder() : nullptr;
-}
-
-int CodecManager::CngPayloadType(int sample_rate_hz) const {
-  switch (sample_rate_hz) {
-    case 8000:
-      return cng_nb_pltype_;
-    case 16000:
-      return cng_wb_pltype_;
-    case 32000:
-      return cng_swb_pltype_;
-    case 48000:
-      return cng_fb_pltype_;
-    default:
-      FATAL() << sample_rate_hz << " Hz is not supported";
-      return -1;
-  }
-}
-
-int CodecManager::RedPayloadType(int sample_rate_hz) const {
-  switch (sample_rate_hz) {
-    case 8000:
-      return red_nb_pltype_;
-    case 16000:
-    case 32000:
-    case 48000:
-      return -1;
-    default:
-      FATAL() << sample_rate_hz << " Hz is not supported";
-      return -1;
-  }
-}
-
-}  // namespace acm2
-}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager.h b/webrtc/modules/audio_coding/main/acm2/codec_manager.h
deleted file mode 100644
index c6c262ea26..0000000000
--- a/webrtc/modules/audio_coding/main/acm2/codec_manager.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/audio_coding/main/acm2/codec_owner.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class AudioDecoder;
-class AudioEncoder;
-
-namespace acm2 {
-
-class CodecManager final {
- public:
-  CodecManager();
-  ~CodecManager();
-
-  int RegisterEncoder(const CodecInst& send_codec);
-
-  void RegisterEncoder(AudioEncoder* external_speech_encoder);
-
-  int GetCodecInst(CodecInst* current_codec) const;
-
-  bool SetCopyRed(bool enable);
-
-  int SetVAD(bool enable, ACMVADMode mode);
-
-  void VAD(bool* dtx_enabled, bool* vad_enabled, ACMVADMode* mode) const;
-
-  int SetCodecFEC(bool enable_codec_fec);
-
-  // Returns a pointer to AudioDecoder of the given codec. For iSAC, encoding
-  // and decoding have to be performed on a shared codec instance. By calling
-  // this method, we get the codec instance that ACM owns.
-  // If |codec| does not share an instance between encoder and decoder, returns
-  // null.
-  AudioDecoder* GetAudioDecoder(const CodecInst& codec);
-
-  bool stereo_send() const { return stereo_send_; }
-
-  bool red_enabled() const { return red_enabled_; }
-
-  bool codec_fec_enabled() const { return codec_fec_enabled_; }
-
-  AudioEncoder* CurrentEncoder() { return codec_owner_.Encoder(); }
-  const AudioEncoder* CurrentEncoder() const { return codec_owner_.Encoder(); }
-
-  bool CurrentEncoderIsOpus() const { return encoder_is_opus_; }
-
- private:
-  int CngPayloadType(int sample_rate_hz) const;
-
-  int RedPayloadType(int sample_rate_hz) const;
-
-  rtc::ThreadChecker thread_checker_;
-  uint8_t cng_nb_pltype_;
-  uint8_t cng_wb_pltype_;
-  uint8_t cng_swb_pltype_;
-  uint8_t cng_fb_pltype_;
-  uint8_t red_nb_pltype_;
-  bool stereo_send_;
-  bool dtx_enabled_;
-  ACMVADMode vad_mode_;
-  CodecInst send_codec_inst_;
-  bool red_enabled_;
-  bool codec_fec_enabled_;
-  CodecOwner codec_owner_;
-  bool encoder_is_opus_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(CodecManager);
-};
-
-}  // namespace acm2
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc b/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
deleted file mode 100644
index df9a992dac..0000000000
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
+++ /dev/null
@@ -1,213 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/codec_owner.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
-#ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
-#endif
-#ifdef WEBRTC_CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h"
-#endif
-#ifdef WEBRTC_CODEC_ISACFX
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
-#endif
-#ifdef WEBRTC_CODEC_ISAC
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h"
-#endif
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
-#ifdef WEBRTC_CODEC_RED
-#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
-#endif
-
-namespace webrtc {
-namespace acm2 {
-
-CodecOwner::CodecOwner() : external_speech_encoder_(nullptr) {
-}
-
-CodecOwner::~CodecOwner() = default;
-
-namespace {
-
-rtc::scoped_ptr<AudioDecoder> CreateIsacDecoder(
-    LockedIsacBandwidthInfo* bwinfo) {
-#if defined(WEBRTC_CODEC_ISACFX)
-  return rtc_make_scoped_ptr(new AudioDecoderIsacFix(bwinfo));
-#elif defined(WEBRTC_CODEC_ISAC)
-  return rtc_make_scoped_ptr(new AudioDecoderIsac(bwinfo));
-#else
-  FATAL() << "iSAC is not supported.";
-  return rtc::scoped_ptr<AudioDecoder>();
-#endif
-}
-
-// Returns a new speech encoder, or null on error.
-// TODO(kwiberg): Don't handle errors here (bug 5033)
-rtc::scoped_ptr<AudioEncoder> CreateSpeechEncoder(
-    const CodecInst& speech_inst,
-    LockedIsacBandwidthInfo* bwinfo) {
-#if defined(WEBRTC_CODEC_ISACFX)
-  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderIsacFix(speech_inst, bwinfo));
-#endif
-#if defined(WEBRTC_CODEC_ISAC)
-  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderIsac(speech_inst, bwinfo));
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-  if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderOpus(speech_inst));
-#endif
-  if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderPcmU(speech_inst));
-  if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderPcmA(speech_inst));
-  if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderPcm16B(speech_inst));
-#ifdef WEBRTC_CODEC_ILBC
-  if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderIlbc(speech_inst));
-#endif
-#ifdef WEBRTC_CODEC_G722
-  if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
-    return rtc_make_scoped_ptr(new AudioEncoderG722(speech_inst));
-#endif
-  LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
-  return rtc::scoped_ptr<AudioEncoder>();
-}
-
-AudioEncoder* CreateRedEncoder(int red_payload_type,
-                               AudioEncoder* encoder,
-                               rtc::scoped_ptr<AudioEncoder>* red_encoder) {
-#ifdef WEBRTC_CODEC_RED
-  if (red_payload_type != -1) {
-    AudioEncoderCopyRed::Config config;
-    config.payload_type = red_payload_type;
-    config.speech_encoder = encoder;
-    red_encoder->reset(new AudioEncoderCopyRed(config));
-    return red_encoder->get();
-  }
-#endif
-
-  red_encoder->reset();
-  return encoder;
-}
-
-void CreateCngEncoder(int cng_payload_type,
-                      ACMVADMode vad_mode,
-                      AudioEncoder* encoder,
-                      rtc::scoped_ptr<AudioEncoder>* cng_encoder) {
-  if (cng_payload_type == -1) {
-    cng_encoder->reset();
-    return;
-  }
-  AudioEncoderCng::Config config;
-  config.num_channels = encoder->NumChannels();
-  config.payload_type = cng_payload_type;
-  config.speech_encoder = encoder;
-  switch (vad_mode) {
-    case VADNormal:
-      config.vad_mode = Vad::kVadNormal;
-      break;
-    case VADLowBitrate:
-      config.vad_mode = Vad::kVadLowBitrate;
-      break;
-    case VADAggr:
-      config.vad_mode = Vad::kVadAggressive;
-      break;
-    case VADVeryAggr:
-      config.vad_mode = Vad::kVadVeryAggressive;
-      break;
-    default:
-      FATAL();
-  }
-  cng_encoder->reset(new AudioEncoderCng(config));
-}
-}  // namespace
-
-bool CodecOwner::SetEncoders(const CodecInst& speech_inst,
-                             int cng_payload_type,
-                             ACMVADMode vad_mode,
-                             int red_payload_type) {
-  speech_encoder_ = CreateSpeechEncoder(speech_inst, &isac_bandwidth_info_);
-  if (!speech_encoder_)
-    return false;
-  external_speech_encoder_ = nullptr;
-  ChangeCngAndRed(cng_payload_type, vad_mode, red_payload_type);
-  return true;
-}
-
-void CodecOwner::SetEncoders(AudioEncoder* external_speech_encoder,
-                             int cng_payload_type,
-                             ACMVADMode vad_mode,
-                             int red_payload_type) {
-  external_speech_encoder_ = external_speech_encoder;
-  speech_encoder_.reset();
-  ChangeCngAndRed(cng_payload_type, vad_mode, red_payload_type);
-}
-
-void CodecOwner::ChangeCngAndRed(int cng_payload_type,
-                                 ACMVADMode vad_mode,
-                                 int red_payload_type) {
-  AudioEncoder* speech_encoder = SpeechEncoder();
-  if (cng_payload_type != -1 || red_payload_type != -1) {
-    // The RED and CNG encoders need to be in sync with the speech encoder, so
-    // reset the latter to ensure its buffer is empty.
-    speech_encoder->Reset();
-  }
-  AudioEncoder* encoder =
-      CreateRedEncoder(red_payload_type, speech_encoder, &red_encoder_);
-  CreateCngEncoder(cng_payload_type, vad_mode, encoder, &cng_encoder_);
-  RTC_DCHECK_EQ(!!speech_encoder_ + !!external_speech_encoder_, 1);
-}
-
-AudioDecoder* CodecOwner::GetIsacDecoder() {
-  if (!isac_decoder_)
-    isac_decoder_ = CreateIsacDecoder(&isac_bandwidth_info_);
-  return isac_decoder_.get();
-}
-
-AudioEncoder* CodecOwner::Encoder() {
-  const auto& const_this = *this;
-  return const_cast<AudioEncoder*>(const_this.Encoder());
-}
-
-const AudioEncoder* CodecOwner::Encoder() const {
-  if (cng_encoder_)
-    return cng_encoder_.get();
-  if (red_encoder_)
-    return red_encoder_.get();
-  return SpeechEncoder();
-}
-
-AudioEncoder* CodecOwner::SpeechEncoder() {
-  const auto* const_this = this;
-  return const_cast<AudioEncoder*>(const_this->SpeechEncoder());
-}
-
-const AudioEncoder* CodecOwner::SpeechEncoder() const {
-  RTC_DCHECK(!speech_encoder_ || !external_speech_encoder_);
-  return external_speech_encoder_ ? external_speech_encoder_
-                                  : speech_encoder_.get();
-}
-
-}  // namespace acm2
-}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner.h b/webrtc/modules/audio_coding/main/acm2/codec_owner.h
deleted file mode 100644
index d0fb4f760e..0000000000
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_OWNER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_OWNER_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
-#else
-// Dummy implementation, for when we don't have iSAC.
-namespace webrtc {
-class LockedIsacBandwidthInfo {};
-}
-#endif
-
-namespace webrtc {
-namespace acm2 {
-
-class CodecOwner {
- public:
-  CodecOwner();
-  ~CodecOwner();
-
-  // Start using the specified encoder. Returns false on error.
-  // TODO(kwiberg): Don't handle errors here (bug 5033)
-  bool SetEncoders(const CodecInst& speech_inst,
-                   int cng_payload_type,
-                   ACMVADMode vad_mode,
-                   int red_payload_type) WARN_UNUSED_RESULT;
-
-  void SetEncoders(AudioEncoder* external_speech_encoder,
-                   int cng_payload_type,
-                   ACMVADMode vad_mode,
-                   int red_payload_type);
-
-  void ChangeCngAndRed(int cng_payload_type,
-                       ACMVADMode vad_mode,
-                       int red_payload_type);
-
-  // Returns a pointer to an iSAC decoder owned by the CodecOwner. The decoder
-  // will live as long as the CodecOwner exists.
-  AudioDecoder* GetIsacDecoder();
-
-  AudioEncoder* Encoder();
-  const AudioEncoder* Encoder() const;
-
- private:
-  AudioEncoder* SpeechEncoder();
-  const AudioEncoder* SpeechEncoder() const;
-
-  // At most one of these is non-null:
-  rtc::scoped_ptr<AudioEncoder> speech_encoder_;
-  AudioEncoder* external_speech_encoder_;
-
-  // If we've created an iSAC decoder because someone called GetIsacDecoder,
-  // store it here.
-  rtc::scoped_ptr<AudioDecoder> isac_decoder_;
-
-  // iSAC bandwidth estimation info, for use with iSAC encoders and decoders.
-  LockedIsacBandwidthInfo isac_bandwidth_info_;
-
-  // |cng_encoder_| and |red_encoder_| are valid iff CNG or RED, respectively,
-  // are active.
-  rtc::scoped_ptr<AudioEncoder> cng_encoder_;
-  rtc::scoped_ptr<AudioEncoder> red_encoder_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(CodecOwner);
-};
-
-}  // namespace acm2
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_OWNER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/main/acm2/rent_a_codec.cc
deleted file mode 100644
index 42f0a4c7db..0000000000
--- a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.cc
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
-
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-
-namespace webrtc {
-namespace acm2 {
-
-rtc::Maybe<RentACodec::CodecId> RentACodec::CodecIdByParams(
-    const char* payload_name,
-    int sampling_freq_hz,
-    int channels) {
-  return CodecIdFromIndex(
-      ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
-}
-
-rtc::Maybe<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
-  rtc::Maybe<int> mi = CodecIndexFromId(codec_id);
-  return mi ? rtc::Maybe<CodecInst>(Database()[*mi]) : rtc::Maybe<CodecInst>();
-}
-
-rtc::Maybe<CodecInst> RentACodec::CodecInstByParams(const char* payload_name,
-                                                    int sampling_freq_hz,
-                                                    int channels) {
-  rtc::Maybe<CodecId> codec_id =
-      CodecIdByParams(payload_name, sampling_freq_hz, channels);
-  if (!codec_id)
-    return rtc::Maybe<CodecInst>();
-  rtc::Maybe<CodecInst> ci = CodecInstById(*codec_id);
-  RTC_DCHECK(ci);
-
-  // Keep the number of channels from the function call. For most codecs it
-  // will be the same value as in default codec settings, but not for all.
-  ci->channels = channels;
-
-  return ci;
-}
-
-bool RentACodec::IsCodecValid(const CodecInst& codec_inst) {
-  return ACMCodecDB::CodecNumber(codec_inst) >= 0;
-}
-
-rtc::ArrayView<const CodecInst> RentACodec::Database() {
-  return rtc::ArrayView<const CodecInst>(ACMCodecDB::database_,
-                                         NumberOfCodecs());
-}
-
-rtc::Maybe<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(CodecId codec_id,
-                                                             int num_channels) {
-  rtc::Maybe<int> i = CodecIndexFromId(codec_id);
-  if (!i)
-    return rtc::Maybe<NetEqDecoder>();
-  const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
-  return (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
-             ? NetEqDecoder::kDecoderOpus_2ch
-             : ned;
-}
-
-}  // namespace acm2
-}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.h b/webrtc/modules/audio_coding/main/acm2/rent_a_codec.h
deleted file mode 100644
index 55a5d0361a..0000000000
--- a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.h
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_
-
-#include <stddef.h>
-
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/maybe.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-struct CodecInst;
-
-namespace acm2 {
-
-class RentACodec {
- public:
-  enum class CodecId {
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-    kISAC,
-#endif
-#ifdef WEBRTC_CODEC_ISAC
-    kISACSWB,
-#endif
-    // Mono
-    kPCM16B,
-    kPCM16Bwb,
-    kPCM16Bswb32kHz,
-    // Stereo
-    kPCM16B_2ch,
-    kPCM16Bwb_2ch,
-    kPCM16Bswb32kHz_2ch,
-    // Mono
-    kPCMU,
-    kPCMA,
-    // Stereo
-    kPCMU_2ch,
-    kPCMA_2ch,
-#ifdef WEBRTC_CODEC_ILBC
-    kILBC,
-#endif
-#ifdef WEBRTC_CODEC_G722
-    kG722,      // Mono
-    kG722_2ch,  // Stereo
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-    kOpus,  // Mono and stereo
-#endif
-    kCNNB,
-    kCNWB,
-    kCNSWB,
-#ifdef ENABLE_48000_HZ
-    kCNFB,
-#endif
-    kAVT,
-#ifdef WEBRTC_CODEC_RED
-    kRED,
-#endif
-    kNumCodecs,  // Implementation detail. Don't use.
-
-// Set unsupported codecs to -1.
-#if !defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX)
-    kISAC = -1,
-#endif
-#ifndef WEBRTC_CODEC_ISAC
-    kISACSWB = -1,
-#endif
-    // 48 kHz not supported, always set to -1.
-    kPCM16Bswb48kHz = -1,
-#ifndef WEBRTC_CODEC_ILBC
-    kILBC = -1,
-#endif
-#ifndef WEBRTC_CODEC_G722
-    kG722 = -1,      // Mono
-    kG722_2ch = -1,  // Stereo
-#endif
-#ifndef WEBRTC_CODEC_OPUS
-    kOpus = -1,  // Mono and stereo
-#endif
-#ifndef WEBRTC_CODEC_RED
-    kRED = -1,
-#endif
-#ifndef ENABLE_48000_HZ
-    kCNFB = -1,
-#endif
-
-    kNone = -1
-  };
-
-  enum class NetEqDecoder {
-    kDecoderPCMu,
-    kDecoderPCMa,
-    kDecoderPCMu_2ch,
-    kDecoderPCMa_2ch,
-    kDecoderILBC,
-    kDecoderISAC,
-    kDecoderISACswb,
-    kDecoderPCM16B,
-    kDecoderPCM16Bwb,
-    kDecoderPCM16Bswb32kHz,
-    kDecoderPCM16Bswb48kHz,
-    kDecoderPCM16B_2ch,
-    kDecoderPCM16Bwb_2ch,
-    kDecoderPCM16Bswb32kHz_2ch,
-    kDecoderPCM16Bswb48kHz_2ch,
-    kDecoderPCM16B_5ch,
-    kDecoderG722,
-    kDecoderG722_2ch,
-    kDecoderRED,
-    kDecoderAVT,
-    kDecoderCNGnb,
-    kDecoderCNGwb,
-    kDecoderCNGswb32kHz,
-    kDecoderCNGswb48kHz,
-    kDecoderArbitrary,
-    kDecoderOpus,
-    kDecoderOpus_2ch,
-  };
-
-  static inline size_t NumberOfCodecs() {
-    return static_cast<size_t>(CodecId::kNumCodecs);
-  }
-
-  static inline rtc::Maybe<int> CodecIndexFromId(CodecId codec_id) {
-    const int i = static_cast<int>(codec_id);
-    return i < static_cast<int>(NumberOfCodecs()) ? i : rtc::Maybe<int>();
-  }
-
-  static inline rtc::Maybe<CodecId> CodecIdFromIndex(int codec_index) {
-    return static_cast<size_t>(codec_index) < NumberOfCodecs()
-               ? static_cast<RentACodec::CodecId>(codec_index)
-               : rtc::Maybe<RentACodec::CodecId>();
-  }
-
-  static rtc::Maybe<CodecId> CodecIdByParams(const char* payload_name,
-                                             int sampling_freq_hz,
-                                             int channels);
-  static rtc::Maybe<CodecInst> CodecInstById(CodecId codec_id);
-  static rtc::Maybe<CodecInst> CodecInstByParams(const char* payload_name,
-                                                 int sampling_freq_hz,
-                                                 int channels);
-  static bool IsCodecValid(const CodecInst& codec_inst);
-  static rtc::ArrayView<const CodecInst> Database();
-
-  static rtc::Maybe<NetEqDecoder> NetEqDecoderFromCodecId(CodecId codec_id,
-                                                          int num_channels);
-};
-
-}  // namespace acm2
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_
diff --git a/webrtc/modules/audio_coding/main/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/audio_coding_module.gypi
deleted file mode 100644
index 6fb37d25fa..0000000000
--- a/webrtc/modules/audio_coding/main/audio_coding_module.gypi
+++ /dev/null
@@ -1,193 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'variables': {
-    'audio_coding_dependencies': [
-      'cng',
-      'g711',
-      'pcm16b',
-      '<(webrtc_root)/common.gyp:webrtc_common',
-      '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
-      '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-    ],
-    'audio_coding_defines': [],
-    'conditions': [
-      ['include_opus==1', {
-        'audio_coding_dependencies': ['webrtc_opus',],
-        'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
-      }],
-      ['build_with_mozilla==0', {
-        'conditions': [
-          ['target_arch=="arm"', {
-            'audio_coding_dependencies': ['isac_fix',],
-            'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
-          }, {
-            'audio_coding_dependencies': ['isac',],
-            'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
-          }],
-        ],
-        'audio_coding_dependencies': ['g722',],
-        'audio_coding_defines': ['WEBRTC_CODEC_G722',],
-      }],
-      ['build_with_mozilla==0 and build_with_chromium==0', {
-        'audio_coding_dependencies': ['ilbc', 'red',],
-        'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
-      }],
-    ],
-  },
-  'targets': [
-    {
-      'target_name': 'rent_a_codec',
-      'type': 'static_library',
-      'defines': [
-        '<@(audio_coding_defines)',
-      ],
-      'dependencies': [
-        '<(webrtc_root)/common.gyp:webrtc_common',
-      ],
-      'include_dirs': [
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '<(webrtc_root)',
-        ],
-      },
-      'sources': [
-        'acm2/acm_codec_database.cc',
-        'acm2/acm_codec_database.h',
-        'acm2/rent_a_codec.cc',
-        'acm2/rent_a_codec.h',
-      ],
-    },
-    {
-      'target_name': 'audio_coding_module',
-      'type': 'static_library',
-      'defines': [
-        '<@(audio_coding_defines)',
-      ],
-      'dependencies': [
-        '<@(audio_coding_dependencies)',
-        '<(webrtc_root)/common.gyp:webrtc_common',
-        '<(webrtc_root)/webrtc.gyp:rtc_event_log',
-        'neteq',
-        'rent_a_codec',
-      ],
-      'include_dirs': [
-        'include',
-        '../../interface',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '../../interface',
-          '<(webrtc_root)',
-        ],
-      },
-      'sources': [
-        'acm2/acm_common_defs.h',
-        'acm2/acm_receiver.cc',
-        'acm2/acm_receiver.h',
-        'acm2/acm_resampler.cc',
-        'acm2/acm_resampler.h',
-        'acm2/audio_coding_module.cc',
-        'acm2/audio_coding_module_impl.cc',
-        'acm2/audio_coding_module_impl.h',
-        'acm2/call_statistics.cc',
-        'acm2/call_statistics.h',
-        'acm2/codec_manager.cc',
-        'acm2/codec_manager.h',
-        'acm2/codec_owner.cc',
-        'acm2/codec_owner.h',
-        'acm2/initial_delay_manager.cc',
-        'acm2/initial_delay_manager.h',
-        'include/audio_coding_module.h',
-        'include/audio_coding_module_typedefs.h',
-      ],
-    },
-  ],
-  'conditions': [
-    ['include_tests==1', {
-      'targets': [
-        {
-          'target_name': 'acm_receive_test',
-          'type': 'static_library',
-          'defines': [
-            '<@(audio_coding_defines)',
-          ],
-          'dependencies': [
-            '<@(audio_coding_dependencies)',
-            'audio_coding_module',
-            'neteq_unittest_tools',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-          ],
-          'sources': [
-            'acm2/acm_receive_test_oldapi.cc',
-            'acm2/acm_receive_test_oldapi.h',
-          ],
-        }, # acm_receive_test
-        {
-          'target_name': 'acm_send_test',
-          'type': 'static_library',
-          'defines': [
-            '<@(audio_coding_defines)',
-          ],
-          'dependencies': [
-            '<@(audio_coding_dependencies)',
-            'audio_coding_module',
-            'neteq_unittest_tools',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-          ],
-          'sources': [
-            'acm2/acm_send_test_oldapi.cc',
-            'acm2/acm_send_test_oldapi.h',
-          ],
-        }, # acm_send_test
-        {
-          'target_name': 'delay_test',
-          'type': 'executable',
-          'dependencies': [
-            'audio_coding_module',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/test/test.gyp:test_support',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-          ],
-          'sources': [
-             'test/delay_test.cc',
-             'test/Channel.cc',
-             'test/PCMFile.cc',
-             'test/utility.cc',
-           ],
-        }, # delay_test
-        {
-          'target_name': 'insert_packet_with_timing',
-          'type': 'executable',
-          'dependencies': [
-            'audio_coding_module',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/test/test.gyp:test_support',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-          ],
-          'sources': [
-             'test/insert_packet_with_timing.cc',
-             'test/Channel.cc',
-             'test/PCMFile.cc',
-           ],
-        }, # delay_test
-      ],
-    }],
-  ],
-}
diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
deleted file mode 100644
index 8495e0e596..0000000000
--- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ /dev/null
@@ -1,175 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-
-#include <assert.h>
-#include <math.h>
-
-#include <iostream>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-namespace {
-
-double FrameRms(AudioFrame& frame) {
-  size_t samples = frame.num_channels_ * frame.samples_per_channel_;
-  double rms = 0;
-  for (size_t n = 0; n < samples; ++n)
-    rms += frame.data_[n] * frame.data_[n];
-  rms /= samples;
-  rms = sqrt(rms);
-  return rms;
-}
-
-}
-
-class InitialPlayoutDelayTest : public ::testing::Test {
- protected:
-  InitialPlayoutDelayTest()
-      : acm_a_(AudioCodingModule::Create(0)),
-        acm_b_(AudioCodingModule::Create(1)),
-        channel_a2b_(NULL) {}
-
-  ~InitialPlayoutDelayTest() {
-    if (channel_a2b_ != NULL) {
-      delete channel_a2b_;
-      channel_a2b_ = NULL;
-    }
-  }
-
-  void SetUp() {
-    ASSERT_TRUE(acm_a_.get() != NULL);
-    ASSERT_TRUE(acm_b_.get() != NULL);
-
-    EXPECT_EQ(0, acm_b_->InitializeReceiver());
-    EXPECT_EQ(0, acm_a_->InitializeReceiver());
-
-    // Register all L16 codecs in receiver.
-    CodecInst codec;
-    const int kFsHz[3] = { 8000, 16000, 32000 };
-    const int kChannels[2] = { 1, 2 };
-    for (int n = 0; n < 3; ++n) {
-      for (int k = 0; k < 2; ++k) {
-        AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
-        acm_b_->RegisterReceiveCodec(codec);
-      }
-    }
-
-    // Create and connect the channel
-    channel_a2b_ = new Channel;
-    acm_a_->RegisterTransportCallback(channel_a2b_);
-    channel_a2b_->RegisterReceiverACM(acm_b_.get());
-  }
-
-  void NbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 1);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
-  void NbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 2);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
- private:
-  void Run(CodecInst codec, int initial_delay_ms) {
-    AudioFrame in_audio_frame;
-    AudioFrame out_audio_frame;
-    int num_frames = 0;
-    const int kAmp = 10000;
-    in_audio_frame.sample_rate_hz_ = codec.plfreq;
-    in_audio_frame.num_channels_ = codec.channels;
-    in_audio_frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
-    size_t samples = in_audio_frame.num_channels_ *
-        in_audio_frame.samples_per_channel_;
-    for (size_t n = 0; n < samples; ++n) {
-      in_audio_frame.data_[n] = kAmp;
-    }
-
-    uint32_t timestamp = 0;
-    double rms = 0;
-    ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
-    acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
-    while (rms < kAmp / 2) {
-      in_audio_frame.timestamp_ = timestamp;
-      timestamp += static_cast<uint32_t>(in_audio_frame.samples_per_channel_);
-      ASSERT_GE(acm_a_->Add10MsData(in_audio_frame), 0);
-      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
-      rms = FrameRms(out_audio_frame);
-      ++num_frames;
-    }
-
-    ASSERT_GE(num_frames * 10, initial_delay_ms);
-    ASSERT_LE(num_frames * 10, initial_delay_ms + 100);
-  }
-
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
-  Channel* channel_a2b_;
-};
-
-TEST_F(InitialPlayoutDelayTest, NbMono) { NbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, WbMono) { WbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
-
-}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index a9ea44d6a6..d800cc7dbe 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -13,13 +13,13 @@
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_decoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
 #endif
 #ifdef WEBRTC_CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 #endif
 #ifdef WEBRTC_CODEC_ISACFX
 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
@@ -30,9 +30,9 @@
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/include/audio_decoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 #endif
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_decoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index 3229033d92..bc8bdd9626 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -16,11 +16,11 @@
 #include "webrtc/engine_configurations.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 #endif
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 8f82fb11a4..599929e78d 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -18,20 +18,20 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_decoder_g722.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/audio_decoder_opus.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_decoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/system_wrappers/include/data_log.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -158,7 +158,10 @@ class AudioDecoderTest : public ::testing::Test {
                                                  interleaved_input.get());
 
       encoded_info_ = audio_encoder_->Encode(
-          0, interleaved_input.get(), audio_encoder_->SampleRateHz() / 100,
+          0, rtc::ArrayView<const int16_t>(interleaved_input.get(),
+                                           audio_encoder_->NumChannels() *
+                                               audio_encoder_->SampleRateHz() /
+                                               100),
           data_length_ * 2, output);
     }
     EXPECT_EQ(payload_type_, encoded_info_.payload_type);
@@ -563,18 +566,14 @@ TEST_F(AudioDecoderIsacSwbTest, SetTargetBitrate) {
   TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 32000);
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4198
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_EncodeDecode DISABLED_EncodeDecode
-#else
-#define MAYBE_EncodeDecode EncodeDecode
-#endif
-TEST_F(AudioDecoderIsacFixTest, MAYBE_EncodeDecode) {
+TEST_F(AudioDecoderIsacFixTest, EncodeDecode) {
   int tolerance = 11034;
   double mse = 3.46e6;
   int delay = 54;  // Delay from input to output.
-#ifdef WEBRTC_ANDROID
+#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
   static const int kEncodedBytes = 685;
+#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
+  static const int kEncodedBytes = 673;
 #else
   static const int kEncodedBytes = 671;
 #endif
diff --git a/webrtc/modules/audio_coding/neteq/comfort_noise.cc b/webrtc/modules/audio_coding/neteq/comfort_noise.cc
index 3fe6607778..a5b08469be 100644
--- a/webrtc/modules/audio_coding/neteq/comfort_noise.cc
+++ b/webrtc/modules/audio_coding/neteq/comfort_noise.cc
@@ -14,7 +14,7 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic.cc b/webrtc/modules/audio_coding/neteq/decision_logic.cc
index 14e0426d7d..39bb4662c7 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -128,9 +128,6 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
   const size_t cur_size_samples =
       samples_left + packet_buffer_.NumSamplesInBuffer(decoder_database_,
                                                        decoder_frame_length);
-  LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
-      " packets * " << decoder_frame_length << " samples/packet + " <<
-      samples_left << " samples in sync buffer = " << cur_size_samples;
 
   prev_time_scale_ = prev_time_scale_ &&
       (prev_mode == kModeAccelerateSuccess ||
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
index d3f6fa6dd4..0252d1cdfa 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -20,7 +20,7 @@
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.cc b/webrtc/modules/audio_coding/neteq/decoder_database.cc
index 41803f754a..92d4bab1e4 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -13,6 +13,7 @@
 #include <assert.h>
 #include <utility>  // pair
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
@@ -38,17 +39,17 @@ void DecoderDatabase::Reset() {
 }
 
 int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
-                                     NetEqDecoder codec_type) {
+                                     NetEqDecoder codec_type,
+                                     const std::string& name) {
   if (rtp_payload_type > 0x7F) {
     return kInvalidRtpPayloadType;
   }
   if (!CodecSupported(codec_type)) {
     return kCodecNotSupported;
   }
-  int fs_hz = CodecSampleRateHz(codec_type);
-  std::pair<DecoderMap::iterator, bool> ret;
-  DecoderInfo info(codec_type, fs_hz, NULL, false);
-  ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
+  const int fs_hz = CodecSampleRateHz(codec_type);
+  DecoderInfo info(codec_type, name, fs_hz, NULL, false);
+  auto ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
   if (ret.second == false) {
     // Database already contains a decoder with type |rtp_payload_type|.
     return kDecoderExists;
@@ -58,6 +59,7 @@ int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
 
 int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
                                     NetEqDecoder codec_type,
+                                    const std::string& codec_name,
                                     int fs_hz,
                                     AudioDecoder* decoder) {
   if (rtp_payload_type > 0x7F) {
@@ -73,7 +75,7 @@ int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
     return kInvalidPointer;
   }
   std::pair<DecoderMap::iterator, bool> ret;
-  DecoderInfo info(codec_type, fs_hz, decoder, true);
+  DecoderInfo info(codec_type, codec_name, fs_hz, decoder, true);
   ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
   if (ret.second == false) {
     // Database already contains a decoder with type |rtp_payload_type|.
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.h b/webrtc/modules/audio_coding/neteq/decoder_database.h
index ea70997c14..f34904fda8 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -12,8 +12,10 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
 
 #include <map>
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"  // NULL
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
@@ -35,25 +37,28 @@ class DecoderDatabase {
 
   // Struct used to store decoder info in the database.
   struct DecoderInfo {
-    // Constructors.
-    DecoderInfo()
-        : codec_type(NetEqDecoder::kDecoderArbitrary),
-          fs_hz(8000),
-          decoder(NULL),
-          external(false) {}
+    DecoderInfo() = default;
     DecoderInfo(NetEqDecoder ct, int fs, AudioDecoder* dec, bool ext)
+        : DecoderInfo(ct, "", fs, dec, ext) {}
+    DecoderInfo(NetEqDecoder ct,
+                const std::string& nm,
+                int fs,
+                AudioDecoder* dec,
+                bool ext)
         : codec_type(ct),
+          name(nm),
           fs_hz(fs),
+          rtp_sample_rate_hz(fs),
           decoder(dec),
-          external(ext) {
-    }
-    // Destructor. (Defined in decoder_database.cc.)
+          external(ext) {}
     ~DecoderInfo();
 
-    NetEqDecoder codec_type;
-    int fs_hz;
-    AudioDecoder* decoder;
-    bool external;
+    NetEqDecoder codec_type = NetEqDecoder::kDecoderArbitrary;
+    std::string name;
+    int fs_hz = 8000;
+    int rtp_sample_rate_hz = 8000;
+    AudioDecoder* decoder = nullptr;
+    bool external = false;
   };
 
   // Maximum value for 8 bits, and an invalid RTP payload type (since it is
@@ -75,16 +80,21 @@ class DecoderDatabase {
   // using InsertExternal().
   virtual void Reset();
 
-  // Registers |rtp_payload_type| as a decoder of type |codec_type|. Returns
-  // kOK on success; otherwise an error code.
+  // Registers |rtp_payload_type| as a decoder of type |codec_type|. The |name|
+  // is only used to populate the name field in the DecoderInfo struct in the
+  // database, and can be arbitrary (including empty). Returns kOK on success;
+  // otherwise an error code.
   virtual int RegisterPayload(uint8_t rtp_payload_type,
-                              NetEqDecoder codec_type);
+                              NetEqDecoder codec_type,
+                              const std::string& name);
 
   // Registers an externally created AudioDecoder object, and associates it
   // as a decoder of type |codec_type| with |rtp_payload_type|.
   virtual int InsertExternal(uint8_t rtp_payload_type,
                              NetEqDecoder codec_type,
-                             int fs_hz, AudioDecoder* decoder);
+                             const std::string& codec_name,
+                             int fs_hz,
+                             AudioDecoder* decoder);
 
   // Removes the entry for |rtp_payload_type| from the database.
   // Returns kDecoderNotFound or kOK depending on the outcome of the operation.
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc b/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
index e85d8d32fb..85aaef1143 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -19,7 +19,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -32,8 +31,10 @@ TEST(DecoderDatabase, CreateAndDestroy) {
 TEST(DecoderDatabase, InsertAndRemove) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu));
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu, kCodecName));
   EXPECT_EQ(1, db.Size());
   EXPECT_FALSE(db.Empty());
   EXPECT_EQ(DecoderDatabase::kOK, db.Remove(kPayloadType));
@@ -44,14 +45,17 @@ TEST(DecoderDatabase, InsertAndRemove) {
 TEST(DecoderDatabase, GetDecoderInfo) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu));
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu, kCodecName));
   const DecoderDatabase::DecoderInfo* info;
   info = db.GetDecoderInfo(kPayloadType);
   ASSERT_TRUE(info != NULL);
   EXPECT_EQ(NetEqDecoder::kDecoderPCMu, info->codec_type);
   EXPECT_EQ(NULL, info->decoder);
   EXPECT_EQ(8000, info->fs_hz);
+  EXPECT_EQ(kCodecName, info->name);
   EXPECT_FALSE(info->external);
   info = db.GetDecoderInfo(kPayloadType + 1);  // Other payload type.
   EXPECT_TRUE(info == NULL);  // Should not be found.
@@ -60,8 +64,10 @@ TEST(DecoderDatabase, GetDecoderInfo) {
 TEST(DecoderDatabase, GetRtpPayloadType) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu));
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu, kCodecName));
   EXPECT_EQ(kPayloadType, db.GetRtpPayloadType(NetEqDecoder::kDecoderPCMu));
   const uint8_t expected_value = DecoderDatabase::kRtpPayloadTypeError;
   EXPECT_EQ(expected_value,
@@ -72,8 +78,10 @@ TEST(DecoderDatabase, GetRtpPayloadType) {
 TEST(DecoderDatabase, GetDecoder) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCM16B));
+            db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCM16B,
+                               kCodecName));
   AudioDecoder* dec = db.GetDecoder(kPayloadType);
   ASSERT_TRUE(dec != NULL);
 }
@@ -86,14 +94,18 @@ TEST(DecoderDatabase, TypeTests) {
   const uint8_t kPayloadTypeRed = 101;
   const uint8_t kPayloadNotUsed = 102;
   // Load into database.
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypePcmU, NetEqDecoder::kDecoderPCMu, "pcmu"));
   EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypePcmU, NetEqDecoder::kDecoderPCMu));
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypeCng, NetEqDecoder::kDecoderCNGnb));
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypeDtmf, NetEqDecoder::kDecoderAVT));
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypeRed, NetEqDecoder::kDecoderRED));
+            db.RegisterPayload(kPayloadTypeCng, NetEqDecoder::kDecoderCNGnb,
+                               "cng-nb"));
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypeDtmf, NetEqDecoder::kDecoderAVT, "avt"));
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypeRed, NetEqDecoder::kDecoderRED, "red"));
   EXPECT_EQ(4, db.Size());
   // Test.
   EXPECT_FALSE(db.IsComfortNoise(kPayloadNotUsed));
@@ -112,11 +124,12 @@ TEST(DecoderDatabase, TypeTests) {
 TEST(DecoderDatabase, ExternalDecoder) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   MockAudioDecoder decoder;
   // Load into database.
   EXPECT_EQ(DecoderDatabase::kOK,
-            db.InsertExternal(kPayloadType, NetEqDecoder::kDecoderPCMu, 8000,
-                              &decoder));
+            db.InsertExternal(kPayloadType, NetEqDecoder::kDecoderPCMu,
+                              kCodecName, 8000, &decoder));
   EXPECT_EQ(1, db.Size());
   // Get decoder and make sure we get the external one.
   EXPECT_EQ(&decoder, db.GetDecoder(kPayloadType));
@@ -125,6 +138,7 @@ TEST(DecoderDatabase, ExternalDecoder) {
   info = db.GetDecoderInfo(kPayloadType);
   ASSERT_TRUE(info != NULL);
   EXPECT_EQ(NetEqDecoder::kDecoderPCMu, info->codec_type);
+  EXPECT_EQ(kCodecName, info->name);
   EXPECT_EQ(&decoder, info->decoder);
   EXPECT_EQ(8000, info->fs_hz);
   EXPECT_TRUE(info->external);
@@ -146,7 +160,7 @@ TEST(DecoderDatabase, CheckPayloadTypes) {
   for (uint8_t payload_type = 0; payload_type < kNumPayloads; ++payload_type) {
     EXPECT_EQ(
         DecoderDatabase::kOK,
-        db.RegisterPayload(payload_type, NetEqDecoder::kDecoderArbitrary));
+        db.RegisterPayload(payload_type, NetEqDecoder::kDecoderArbitrary, ""));
   }
   PacketList packet_list;
   for (int i = 0; i < kNumPayloads + 1; ++i) {
@@ -185,11 +199,11 @@ TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
   DecoderDatabase db;
   // Load payload types.
   ASSERT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(0, NetEqDecoder::kDecoderPCMu));
+            db.RegisterPayload(0, NetEqDecoder::kDecoderPCMu, "pcmu"));
   ASSERT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(103, NetEqDecoder::kDecoderISAC));
+            db.RegisterPayload(103, NetEqDecoder::kDecoderISAC, "isac"));
   ASSERT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(13, NetEqDecoder::kDecoderCNGnb));
+            db.RegisterPayload(13, NetEqDecoder::kDecoderCNGnb, "cng-nb"));
   // Verify that no decoders are active from the start.
   EXPECT_EQ(NULL, db.GetActiveDecoder());
   EXPECT_EQ(NULL, db.GetActiveCngDecoder());
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc
index 5140c0620f..806d02b8de 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -17,7 +17,7 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index 2aa9fb0a8d..ef7af46597 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -519,7 +519,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     energy2 = WebRtcSpl_DotProductWithScale(vector2, vector2, expansion_length,
                                             correlation_scale);
     // Confirm that amplitude ratio sqrt(energy1 / energy2) is within 0.5 - 2.0,
-    // i.e., energy1 / energy1 is within 0.25 - 4.
+    // i.e., energy1 / energy2 is within 0.25 - 4.
     int16_t amplitude_ratio;
     if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
       // Energy constraint fulfilled. Use both vectors and scale them
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 205a0dfe80..1322223970 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -81,6 +81,7 @@ class NetEq {
     Config()
         : sample_rate_hz(16000),
           enable_audio_classifier(false),
+          enable_post_decode_vad(false),
           max_packets_in_buffer(50),
           // |max_delay_ms| has the same effect as calling SetMaximumDelay().
           max_delay_ms(2000),
@@ -92,6 +93,7 @@ class NetEq {
 
     int sample_rate_hz;  // Initial value. Will change with input data.
     bool enable_audio_classifier;
+    bool enable_post_decode_vad;
     size_t max_packets_in_buffer;
     int max_delay_ms;
     BackgroundNoiseMode background_noise_mode;
@@ -145,8 +147,7 @@ class NetEq {
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                           const uint8_t* payload,
-                           size_t length_bytes,
+                           rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp) = 0;
 
   // Inserts a sync-packet into packet queue. Sync-packets are decoded to
@@ -170,20 +171,27 @@ class NetEq {
   // The speech type is written to |type|, if |type| is not NULL.
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(size_t max_length, int16_t* output_audio,
-                       size_t* samples_per_channel, int* num_channels,
+                       size_t* samples_per_channel, size_t* num_channels,
                        NetEqOutputType* type) = 0;
 
-  // Associates |rtp_payload_type| with |codec| and stores the information in
-  // the codec database. Returns 0 on success, -1 on failure.
+  // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
+  // information in the codec database. Returns 0 on success, -1 on failure.
+  // The name is only used to provide information back to the caller about the
+  // decoders. Hence, the name is arbitrary, and may be empty.
   virtual int RegisterPayloadType(NetEqDecoder codec,
+                                  const std::string& codec_name,
                                   uint8_t rtp_payload_type) = 0;
 
   // Provides an externally created decoder object |decoder| to insert in the
   // decoder database. The decoder implements a decoder of type |codec| and
-  // associates it with |rtp_payload_type|. The decoder will produce samples
-  // at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure.
+  // associates it with |rtp_payload_type| and |codec_name|. The decoder will
+  // produce samples at the rate |sample_rate_hz|. Returns kOK on success, kFail
+  // on failure.
+  // The name is only used to provide information back to the caller about the
+  // decoders. Hence, the name is arbitrary, and may be empty.
   virtual int RegisterExternalDecoder(AudioDecoder* decoder,
                                       NetEqDecoder codec,
+                                      const std::string& codec_name,
                                       uint8_t rtp_payload_type,
                                       int sample_rate_hz) = 0;
 
@@ -250,6 +258,11 @@ class NetEq {
   // Returns true if the RTP timestamp is valid, otherwise false.
   virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0;
 
+  // Returns the sample rate in Hz of the audio produced in the last GetAudio
+  // call. If GetAudio has not been called yet, the configured sample rate
+  // (Config::sample_rate_hz) is returned.
+  virtual int last_output_sample_rate_hz() const = 0;
+
   // Not implemented.
   virtual int SetTargetNumberOfChannels() = 0;
 
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
index 8debcbbb1e..c1cc09cb5e 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
@@ -22,9 +22,8 @@ class MockAudioDecoder : public AudioDecoder {
   MockAudioDecoder() {}
   virtual ~MockAudioDecoder() { Die(); }
   MOCK_METHOD0(Die, void());
-  MOCK_METHOD6(
-      Decode,
-      int(const uint8_t*, size_t, int, size_t, int16_t*, SpeechType*));
+  MOCK_METHOD5(DecodeInternal,
+               int(const uint8_t*, size_t, int, int16_t*, SpeechType*));
   MOCK_CONST_METHOD0(HasDecodePlc, bool());
   MOCK_METHOD2(DecodePlc, size_t(size_t, int16_t*));
   MOCK_METHOD0(Reset, void());
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h b/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
index d127c5d810..1b4a3c9da5 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
 
+#include <string>
+
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
@@ -27,10 +29,12 @@ class MockDecoderDatabase : public DecoderDatabase {
       int());
   MOCK_METHOD0(Reset,
       void());
-  MOCK_METHOD2(RegisterPayload,
-      int(uint8_t rtp_payload_type, NetEqDecoder codec_type));
-  MOCK_METHOD4(InsertExternal,
-      int(uint8_t rtp_payload_type, NetEqDecoder codec_type, int fs_hz,
+  MOCK_METHOD3(RegisterPayload,
+      int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
+          const std::string& name));
+  MOCK_METHOD5(InsertExternal,
+      int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
+          const std::string& codec_name, int fs_hz,
           AudioDecoder* decoder));
   MOCK_METHOD1(Remove,
       int(uint8_t rtp_payload_type));
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h b/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
index 8cf89c083d..42c17ae054 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
@@ -15,7 +15,7 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -30,7 +30,6 @@ class ExternalPcm16B : public AudioDecoder {
   ExternalPcm16B() {}
   void Reset() override {}
 
- protected:
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
                      int sample_rate_hz,
@@ -52,8 +51,8 @@ class MockExternalPcm16B : public ExternalPcm16B {
  public:
   MockExternalPcm16B() {
     // By default, all calls are delegated to the real object.
-    ON_CALL(*this, Decode(_, _, _, _, _, _))
-        .WillByDefault(Invoke(&real_, &ExternalPcm16B::Decode));
+    ON_CALL(*this, DecodeInternal(_, _, _, _, _))
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::DecodeInternal));
     ON_CALL(*this, HasDecodePlc())
         .WillByDefault(Invoke(&real_, &ExternalPcm16B::HasDecodePlc));
     ON_CALL(*this, DecodePlc(_, _))
@@ -68,11 +67,10 @@ class MockExternalPcm16B : public ExternalPcm16B {
   virtual ~MockExternalPcm16B() { Die(); }
 
   MOCK_METHOD0(Die, void());
-  MOCK_METHOD6(Decode,
+  MOCK_METHOD5(DecodeInternal,
                int(const uint8_t* encoded,
                    size_t encoded_len,
                    int sample_rate_hz,
-                   size_t max_decoded_bytes,
                    int16_t* decoded,
                    SpeechType* speech_type));
   MOCK_CONST_METHOD0(HasDecodePlc,
diff --git a/webrtc/modules/audio_coding/neteq/nack.cc b/webrtc/modules/audio_coding/neteq/nack.cc
index fd3d762605..011914b3d9 100644
--- a/webrtc/modules/audio_coding/neteq/nack.cc
+++ b/webrtc/modules/audio_coding/neteq/nack.cc
@@ -15,7 +15,7 @@
 #include <algorithm>  // For std::max.
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/nack.h b/webrtc/modules/audio_coding/neteq/nack.h
index 116b7e2192..17fef46464 100644
--- a/webrtc/modules/audio_coding/neteq/nack.h
+++ b/webrtc/modules/audio_coding/neteq/nack.h
@@ -15,7 +15,7 @@
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 
 //
diff --git a/webrtc/modules/audio_coding/neteq/nack_unittest.cc b/webrtc/modules/audio_coding/neteq/nack_unittest.cc
index 853af94ede..53b19dc50f 100644
--- a/webrtc/modules/audio_coding/neteq/nack_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/nack_unittest.cc
@@ -17,7 +17,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/audio_coding/neteq/neteq.cc b/webrtc/modules/audio_coding/neteq/neteq.cc
index ca51c9602d..c31dbdc1a3 100644
--- a/webrtc/modules/audio_coding/neteq/neteq.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq.cc
@@ -32,6 +32,8 @@ std::string NetEq::Config::ToString() const {
   std::stringstream ss;
   ss << "sample_rate_hz=" << sample_rate_hz << ", enable_audio_classifier="
      << (enable_audio_classifier ? "true" : "false")
+     << ", enable_post_decode_vad="
+     << (enable_post_decode_vad ? "true" : "false")
      << ", max_packets_in_buffer=" << max_packets_in_buffer
      << ", background_noise_mode=" << background_noise_mode
      << ", playout_mode=" << playout_mode
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 09eb5614fe..c03fbb7347 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -98,14 +98,16 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
       next_arrival_time = GetArrivalTime(next_send_time);
     } while (Lost());  // If lost, immediately read the next packet.
 
-    EXPECT_CALL(*external_decoder_,
-                Decode(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _, _))
+    EXPECT_CALL(
+        *external_decoder_,
+        DecodeInternal(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _))
         .Times(NumExpectedDecodeCalls(num_loops));
 
     uint32_t time_now = 0;
     for (int k = 0; k < num_loops; ++k) {
       while (time_now >= next_arrival_time) {
-        InsertPacket(rtp_header_, encoded_, payload_size_bytes_,
+        InsertPacket(rtp_header_, rtc::ArrayView<const uint8_t>(
+                                      encoded_, payload_size_bytes_),
                      next_arrival_time);
         // Get next input packet.
         do {
@@ -124,17 +126,14 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
     }
   }
 
-  void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
-                    size_t payload_size_bytes,
+  void InsertPacket(WebRtcRTPHeader rtp_header,
+                    rtc::ArrayView<const uint8_t> payload,
                     uint32_t receive_timestamp) override {
-    EXPECT_CALL(*external_decoder_,
-                IncomingPacket(_,
-                               payload_size_bytes,
-                               rtp_header.header.sequenceNumber,
-                               rtp_header.header.timestamp,
-                               receive_timestamp));
+    EXPECT_CALL(
+        *external_decoder_,
+        IncomingPacket(_, payload.size(), rtp_header.header.sequenceNumber,
+                       rtp_header.header.timestamp, receive_timestamp));
     NetEqExternalDecoderTest::InsertPacket(rtp_header, payload,
-                                           payload_size_bytes,
                                            receive_timestamp);
   }
 
@@ -181,15 +180,15 @@ class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest,
   }
 
   void SetUp() override {
-    ASSERT_EQ(NetEq::kOK,
-              neteq_internal_->RegisterPayloadType(
-                  NetEqDecoder::kDecoderPCM16Bswb32kHz, kPayloadType));
+    ASSERT_EQ(NetEq::kOK, neteq_internal_->RegisterPayloadType(
+                              NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                              "pcm16-swb32", kPayloadType));
   }
 
   void GetAndVerifyOutput() override {
     NetEqOutputType output_type;
     size_t samples_per_channel;
-    int num_channels;
+    size_t num_channels;
     // Get audio from internal decoder instance.
     EXPECT_EQ(NetEq::kOK,
               neteq_internal_->GetAudio(kMaxBlockSize,
@@ -197,7 +196,7 @@ class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest,
                                         &samples_per_channel,
                                         &num_channels,
                                         &output_type));
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
               samples_per_channel);
 
@@ -210,18 +209,15 @@ class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest,
     }
   }
 
-  void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
-                    size_t payload_size_bytes,
+  void InsertPacket(WebRtcRTPHeader rtp_header,
+                    rtc::ArrayView<const uint8_t> payload,
                     uint32_t receive_timestamp) override {
     // Insert packet in internal decoder.
-    ASSERT_EQ(
-        NetEq::kOK,
-        neteq_internal_->InsertPacket(
-            rtp_header, payload, payload_size_bytes, receive_timestamp));
+    ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(rtp_header, payload,
+                                                        receive_timestamp));
 
     // Insert packet in external decoder instance.
     NetEqExternalDecoderUnitTest::InsertPacket(rtp_header, payload,
-                                               payload_size_bytes,
                                                receive_timestamp);
   }
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 92ce41e2ea..6c07da46f0 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -18,6 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
@@ -42,7 +43,7 @@
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
@@ -106,28 +107,28 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
   }
   fs_hz_ = fs;
   fs_mult_ = fs / 8000;
+  last_output_sample_rate_hz_ = fs;
   output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
   decoder_frame_length_ = 3 * output_size_samples_;
   WebRtcSpl_Init();
   if (create_components) {
     SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
   }
+  RTC_DCHECK(!vad_->enabled());
+  if (config.enable_post_decode_vad) {
+    vad_->Enable();
+  }
 }
 
 NetEqImpl::~NetEqImpl() = default;
 
 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
-                            const uint8_t* payload,
-                            size_t length_bytes,
+                            rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp) {
+  TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
-      ", sn=" << rtp_header.header.sequenceNumber <<
-      ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
-      ", ssrc=" << rtp_header.header.ssrc <<
-      ", len=" << length_bytes;
-  int error = InsertPacketInternal(rtp_header, payload, length_bytes,
-                                   receive_timestamp, false);
+  int error =
+      InsertPacketInternal(rtp_header, payload, receive_timestamp, false);
   if (error != 0) {
     error_code_ = error;
     return kFail;
@@ -138,15 +139,9 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                 uint32_t receive_timestamp) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "InsertPacket-Sync: ts="
-      << rtp_header.header.timestamp <<
-      ", sn=" << rtp_header.header.sequenceNumber <<
-      ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
-      ", ssrc=" << rtp_header.header.ssrc;
-
   const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' };
-  int error = InsertPacketInternal(
-      rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true);
+  int error =
+      InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
 
   if (error != 0) {
     error_code_ = error;
@@ -156,14 +151,12 @@ int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
 }
 
 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
-                        size_t* samples_per_channel, int* num_channels,
+                        size_t* samples_per_channel, size_t* num_channels,
                         NetEqOutputType* type) {
+  TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "GetAudio";
   int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
                                num_channels);
-  LOG(LS_VERBOSE) << "Produced " << *samples_per_channel <<
-      " samples/channel for " << *num_channels << " channel(s)";
   if (error != 0) {
     error_code_ = error;
     return kFail;
@@ -171,16 +164,24 @@ int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
   if (type) {
     *type = LastOutputType();
   }
+  last_output_sample_rate_hz_ =
+      rtc::checked_cast<int>(*samples_per_channel * 100);
+  RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
+             last_output_sample_rate_hz_ == 16000 ||
+             last_output_sample_rate_hz_ == 32000 ||
+             last_output_sample_rate_hz_ == 48000)
+      << "Unexpected sample rate " << last_output_sample_rate_hz_;
   return kOK;
 }
 
 int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
+                                   const std::string& name,
                                    uint8_t rtp_payload_type) {
   CriticalSectionScoped lock(crit_sect_.get());
   LOG(LS_VERBOSE) << "RegisterPayloadType "
                   << static_cast<int>(rtp_payload_type) << " "
                   << static_cast<int>(codec);
-  int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec);
+  int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec, name);
   if (ret != DecoderDatabase::kOK) {
     switch (ret) {
       case DecoderDatabase::kInvalidRtpPayloadType:
@@ -202,6 +203,7 @@ int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
 
 int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
                                        NetEqDecoder codec,
+                                       const std::string& codec_name,
                                        uint8_t rtp_payload_type,
                                        int sample_rate_hz) {
   CriticalSectionScoped lock(crit_sect_.get());
@@ -213,8 +215,8 @@ int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
     assert(false);
     return kFail;
   }
-  int ret = decoder_database_->InsertExternal(rtp_payload_type, codec,
-                                              sample_rate_hz, decoder);
+  int ret = decoder_database_->InsertExternal(
+      rtp_payload_type, codec, codec_name, sample_rate_hz, decoder);
   if (ret != DecoderDatabase::kOK) {
     switch (ret) {
       case DecoderDatabase::kInvalidRtpPayloadType:
@@ -370,6 +372,11 @@ bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) {
   return true;
 }
 
+int NetEqImpl::last_output_sample_rate_hz() const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  return last_output_sample_rate_hz_;
+}
+
 int NetEqImpl::SetTargetNumberOfChannels() {
   return kNotImplemented;
 }
@@ -441,12 +448,11 @@ const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
 // Methods below this line are private.
 
 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
-                                    const uint8_t* payload,
-                                    size_t length_bytes,
+                                    rtc::ArrayView<const uint8_t> payload,
                                     uint32_t receive_timestamp,
                                     bool is_sync_packet) {
-  if (!payload) {
-    LOG_F(LS_ERROR) << "payload == NULL";
+  if (payload.empty()) {
+    LOG_F(LS_ERROR) << "payload is empty";
     return kInvalidPointer;
   }
   // Sanity checks for sync-packets.
@@ -482,7 +488,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     packet->header.timestamp = rtp_header.header.timestamp;
     packet->header.ssrc = rtp_header.header.ssrc;
     packet->header.numCSRCs = 0;
-    packet->payload_length = length_bytes;
+    packet->payload_length = payload.size();
     packet->primary = true;
     packet->waiting_time = 0;
     packet->payload = new uint8_t[packet->payload_length];
@@ -490,8 +496,8 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     if (!packet->payload) {
       LOG_F(LS_ERROR) << "Payload pointer is NULL.";
     }
-    assert(payload);  // Already checked above.
-    memcpy(packet->payload, payload, packet->payload_length);
+    assert(!payload.empty());  // Already checked above.
+    memcpy(packet->payload, payload.data(), packet->payload_length);
     // Insert packet in a packet list.
     packet_list.push_back(packet);
     // Save main payloads header for later.
@@ -738,7 +744,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
 int NetEqImpl::GetAudioInternal(size_t max_length,
                                 int16_t* output,
                                 size_t* samples_per_channel,
-                                int* num_channels) {
+                                size_t* num_channels) {
   PacketList packet_list;
   DtmfEvent dtmf_event;
   Operations operation;
@@ -749,8 +755,6 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
     last_mode_ = kModeError;
     return return_value;
   }
-  LOG(LS_VERBOSE) << "GetDecision returned operation=" << operation <<
-      " and " << packet_list.size() << " packet(s)";
 
   AudioDecoder::SpeechType speech_type;
   int length = 0;
@@ -864,10 +868,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
   const size_t samples_from_sync =
       sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
                                             output);
-  *num_channels = static_cast<int>(sync_buffer_->Channels());
-  LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
-      " insert " << algorithm_buffer_->Size() << " samples, extract " <<
-      samples_from_sync << " samples";
+  *num_channels = sync_buffer_->Channels();
   if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
     // The sync buffer should always contain |overlap_length| samples, but now
     // too many samples have been extracted. Reinstall the |overlap_length|
@@ -1325,7 +1326,6 @@ int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
             &decoded_buffer_[*decoded_length], speech_type);
     if (length > 0) {
       *decoded_length += length;
-      LOG(LS_VERBOSE) << "Decoded " << length << " CNG samples";
     } else {
       // Error.
       LOG(LS_WARNING) << "Failed to decode CNG";
@@ -1365,34 +1365,17 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
     int decode_length;
     if (packet->sync_packet) {
       // Decode to silence with the same frame size as the last decode.
-      LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
-          " ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       memset(&decoded_buffer_[*decoded_length], 0,
              decoder_frame_length_ * decoder->Channels() *
                  sizeof(decoded_buffer_[0]));
       decode_length = rtc::checked_cast<int>(decoder_frame_length_);
     } else if (!packet->primary) {
       // This is a redundant payload; call the special decoder method.
-      LOG(LS_VERBOSE) << "Decoding packet (redundant):" <<
-          " ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       decode_length = decoder->DecodeRedundant(
           packet->payload, packet->payload_length, fs_hz_,
           (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
           &decoded_buffer_[*decoded_length], speech_type);
     } else {
-      LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       decode_length =
           decoder->Decode(
               packet->payload, packet->payload_length, fs_hz_,
@@ -1408,9 +1391,6 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
       // Update |decoder_frame_length_| with number of samples per channel.
       decoder_frame_length_ =
           static_cast<size_t>(decode_length) / decoder->Channels();
-      LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples ("
-                      << decoder->Channels() << " channel(s) -> "
-                      << decoder_frame_length_ << " samples per channel)";
     } else if (decode_length < 0) {
       // Error.
       LOG(LS_WARNING) << "Decode " << decode_length << " " << payload_length;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index c001e53b81..940deadd2f 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
 
+#include <string>
+
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
@@ -79,8 +81,7 @@ class NetEqImpl : public webrtc::NetEq {
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                   const uint8_t* payload,
-                   size_t length_bytes,
+                   rtc::ArrayView<const uint8_t> payload,
                    uint32_t receive_timestamp) override;
 
   // Inserts a sync-packet into packet queue. Sync-packets are decoded to
@@ -106,20 +107,16 @@ class NetEqImpl : public webrtc::NetEq {
   int GetAudio(size_t max_length,
                int16_t* output_audio,
                size_t* samples_per_channel,
-               int* num_channels,
+               size_t* num_channels,
                NetEqOutputType* type) override;
 
-  // Associates |rtp_payload_type| with |codec| and stores the information in
-  // the codec database. Returns kOK on success, kFail on failure.
   int RegisterPayloadType(NetEqDecoder codec,
+                          const std::string& codec_name,
                           uint8_t rtp_payload_type) override;
 
-  // Provides an externally created decoder object |decoder| to insert in the
-  // decoder database. The decoder implements a decoder of type |codec| and
-  // associates it with |rtp_payload_type|. The decoder will produce samples
-  // at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure.
   int RegisterExternalDecoder(AudioDecoder* decoder,
                               NetEqDecoder codec,
+                              const std::string& codec_name,
                               uint8_t rtp_payload_type,
                               int sample_rate_hz) override;
 
@@ -169,6 +166,8 @@ class NetEqImpl : public webrtc::NetEq {
 
   bool GetPlayoutTimestamp(uint32_t* timestamp) override;
 
+  int last_output_sample_rate_hz() const override;
+
   int SetTargetNumberOfChannels() override;
 
   int SetTargetSampleRate() override;
@@ -207,8 +206,7 @@ class NetEqImpl : public webrtc::NetEq {
   // above. Returns 0 on success, otherwise an error code.
   // TODO(hlundin): Merge this with InsertPacket above?
   int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
-                           const uint8_t* payload,
-                           size_t length_bytes,
+                           rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp,
                            bool is_sync_packet)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
@@ -222,7 +220,8 @@ class NetEqImpl : public webrtc::NetEq {
   int GetAudioInternal(size_t max_length,
                        int16_t* output,
                        size_t* samples_per_channel,
-                       int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+                       size_t* num_channels)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Provides a decision to the GetAudioInternal method. The decision what to
   // do is written to |operation|. Packets to decode are written to
@@ -377,6 +376,7 @@ class NetEqImpl : public webrtc::NetEq {
   StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
   int fs_hz_ GUARDED_BY(crit_sect_);
   int fs_mult_ GUARDED_BY(crit_sect_);
+  int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
   size_t output_size_samples_ GUARDED_BY(crit_sect_);
   size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
   Modes last_mode_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 90640ca1d2..f734883635 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -240,9 +240,10 @@ TEST_F(NetEqImplTest, RegisterPayloadType) {
   CreateInstance();
   uint8_t rtp_payload_type = 0;
   NetEqDecoder codec_type = NetEqDecoder::kDecoderPCMu;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_CALL(*mock_decoder_database_,
-              RegisterPayload(rtp_payload_type, codec_type));
-  neteq_->RegisterPayloadType(codec_type, rtp_payload_type);
+              RegisterPayload(rtp_payload_type, codec_type, kCodecName));
+  neteq_->RegisterPayloadType(codec_type, kCodecName, rtp_payload_type);
 }
 
 TEST_F(NetEqImplTest, RemovePayloadType) {
@@ -359,13 +360,12 @@ TEST_F(NetEqImplTest, InsertPacket) {
       .WillRepeatedly(Return(PayloadSplitter::kOK));
 
   // Insert first packet.
-  neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime);
+  neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime);
 
   // Insert second packet.
   rtp_header.header.timestamp += 160;
   rtp_header.header.sequenceNumber += 1;
-  neteq_->InsertPacket(rtp_header, payload, kPayloadLength,
-                       kFirstReceiveTime + 155);
+  neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155);
 }
 
 TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
@@ -384,13 +384,12 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
   rtp_header.header.ssrc = 0x87654321;
 
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
-                            NetEqDecoder::kDecoderPCM16B, kPayloadType));
+                            NetEqDecoder::kDecoderPCM16B, "", kPayloadType));
 
   // Insert packets. The buffer should not flush.
   for (size_t i = 1; i <= config_.max_packets_in_buffer; ++i) {
     EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(
-                  rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
     rtp_header.header.timestamp += kPayloadLengthSamples;
     rtp_header.header.sequenceNumber += 1;
     EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
@@ -399,8 +398,7 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
   // Insert one more packet and make sure the buffer got flushed. That is, it
   // should only hold one single packet.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
   const RTPHeader* test_header = packet_buffer_->NextRtpHeader();
   EXPECT_EQ(rtp_header.header.timestamp, test_header->timestamp);
@@ -434,12 +432,11 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
     CountingSamplesDecoder() : next_value_(1) {}
 
     // Produce as many samples as input bytes (|encoded_len|).
-    int Decode(const uint8_t* encoded,
-               size_t encoded_len,
-               int /* sample_rate_hz */,
-               size_t /* max_decoded_bytes */,
-               int16_t* decoded,
-               SpeechType* speech_type) override {
+    int DecodeInternal(const uint8_t* encoded,
+                       size_t encoded_len,
+                       int /* sample_rate_hz */,
+                       int16_t* decoded,
+                       SpeechType* speech_type) override {
       for (size_t i = 0; i < encoded_len; ++i) {
         decoded[i] = next_value_++;
       }
@@ -459,25 +456,24 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
 
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
                             &decoder_, NetEqDecoder::kDecoderPCM16B,
-                            kPayloadType, kSampleRateHz));
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert one packet.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(
       NetEq::kOK,
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Start with a simple check that the fake decoder is behaving as expected.
@@ -531,33 +527,32 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   int16_t dummy_output[kPayloadLengthSamples] = {0};
   // The below expectation will make the mock decoder write
   // |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
-  EXPECT_CALL(mock_decoder,
-              Decode(Pointee(0), kPayloadLengthBytes, kSampleRateHz, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
                             &mock_decoder, NetEqDecoder::kDecoderPCM16B,
-                            kPayloadType, kSampleRateHz));
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert one packet.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(
       NetEq::kOK,
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Insert two more packets. The first one is out of order, and is already too
@@ -566,22 +561,20 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   rtp_header.header.timestamp -= kPayloadLengthSamples;
   payload[0] = 1;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   rtp_header.header.sequenceNumber += 2;
   rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
   payload[0] = 2;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Expect only the second packet to be decoded (the one with "2" as the first
   // payload byte).
-  EXPECT_CALL(mock_decoder,
-              Decode(Pointee(2), kPayloadLengthBytes, kSampleRateHz, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
 
   // Pull audio once.
@@ -590,7 +583,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Now check the packet buffer, and make sure it is empty, since the
@@ -622,35 +615,33 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
   // Insert one packet. Note that we have not registered any payload type, so
   // this packet will be rejected.
   EXPECT_EQ(NetEq::kFail,
-            neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                 kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   ASSERT_LE(samples_per_channel, kMaxOutputSize);
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputPLC, type);
 
   // Register the payload type.
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
-                            NetEqDecoder::kDecoderPCM16B, kPayloadType));
+                            NetEqDecoder::kDecoderPCM16B, "", kPayloadType));
 
   // Insert 10 packets.
   for (size_t i = 0; i < 10; ++i) {
     rtp_header.header.sequenceNumber++;
     rtp_header.header.timestamp += kPayloadLengthSamples;
     EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                   kReceiveTime));
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
     EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
   }
 
@@ -661,7 +652,7 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
                                &num_channels, &type));
     ASSERT_LE(samples_per_channel, kMaxOutputSize);
     EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(kOutputNormal, type)
         << "NetEq did not decode the packets as expected.";
   }
@@ -697,54 +688,53 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
 
   // Pointee(x) verifies that first byte of the payload equals x, this makes it
   // possible to verify that the correct payload is fed to Decode().
-  EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes,
-                                   kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
+                                           kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
 
-  EXPECT_CALL(mock_decoder, Decode(Pointee(1), kPayloadLengthBytes,
-                                   kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(1), kPayloadLengthBytes,
+                                           kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kComfortNoise),
+                      SetArgPointee<4>(AudioDecoder::kComfortNoise),
                       Return(kPayloadLengthSamples)));
 
-  EXPECT_CALL(mock_decoder, Decode(IsNull(), 0, kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder,
+              DecodeInternal(IsNull(), 0, kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kComfortNoise),
+                      SetArgPointee<4>(AudioDecoder::kComfortNoise),
                       Return(kPayloadLengthSamples)));
 
-  EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes,
-                                   kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
+                                           kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
 
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
                             &mock_decoder, NetEqDecoder::kDecoderOpus,
-                            kPayloadType, kSampleRateKhz * 1000));
+                            "dummy name", kPayloadType, kSampleRateKhz * 1000));
 
   // Insert one packet (decoder will return speech).
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Insert second packet (decoder will return CNG).
   payload[0] = 1;
   rtp_header.header.sequenceNumber++;
   rtp_header.header.timestamp += kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   uint32_t timestamp;
   uint32_t last_timestamp;
   NetEqOutputType type;
@@ -769,7 +759,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
 
   for (size_t i = 1; i < 6; ++i) {
     ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(expected_type[i - 1], type);
     EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&timestamp));
     EXPECT_EQ(NetEq::kOK,
@@ -785,12 +775,11 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
   rtp_header.header.sequenceNumber += 2;
   rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   for (size_t i = 6; i < 8; ++i) {
     ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(expected_type[i - 1], type);
     EXPECT_EQ(NetEq::kOK,
               neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
@@ -810,7 +799,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
   UseNoMocks();
   CreateInstance();
   static const size_t kNetEqMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
-  static const int kChannels = 2;
+  static const size_t kChannels = 2;
 
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
@@ -866,13 +855,12 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
 
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
                             &decoder_, NetEqDecoder::kDecoderPCM16B,
-                            kPayloadType, kSampleRateHz));
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert one packet.
   payload[0] = kFirstPayloadValue;  // This will make Decode() fail.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Insert another packet.
   payload[0] = kSecondPayloadValue;  // This will make Decode() successful.
@@ -881,14 +869,12 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
   // the second packet get decoded.
   rtp_header.header.timestamp += 3 * kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
-  const size_t kMaxOutputSize =
-      static_cast<size_t>(10 * kSampleRateHz / 1000 * kChannels);
+  const size_t kMaxOutputSize = 10 * kSampleRateHz / 1000 * kChannels;
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
 
   EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(kMaxOutputSize, output,
@@ -926,14 +912,13 @@ TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
   rtp_header.header.ssrc = 0x87654321;
 
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
-                            NetEqDecoder::kDecoderPCM16B, kPayloadType));
+                            NetEqDecoder::kDecoderPCM16B, "", kPayloadType));
 
   // Insert packets until the buffer flushes.
   for (size_t i = 0; i <= config_.max_packets_in_buffer; ++i) {
     EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
     EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                   kReceiveTime));
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
     rtp_header.header.timestamp +=
         rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
     ++rtp_header.header.sequenceNumber;
@@ -975,20 +960,19 @@ TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
   // |kPayloadLengthSamples| - 5 zeros to the output array, and mark it as
   // speech. That is, the decoded length is 5 samples shorter than the expected.
   EXPECT_CALL(mock_decoder,
-              Decode(_, kPayloadLengthBytes, kSampleRateHz, _, _, _))
+              DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
       .WillOnce(
-          DoAll(SetArrayArgument<4>(dummy_output,
+          DoAll(SetArrayArgument<3>(dummy_output,
                                     dummy_output + kPayloadLengthSamples - 5),
-                SetArgPointee<5>(AudioDecoder::kSpeech),
+                SetArgPointee<4>(AudioDecoder::kSpeech),
                 Return(kPayloadLengthSamples - 5)));
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
                             &mock_decoder, NetEqDecoder::kDecoderPCM16B,
-                            kPayloadType, kSampleRateHz));
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert one packet.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                 kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
 
@@ -996,13 +980,13 @@ TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   EXPECT_CALL(mock_decoder, Die());
@@ -1050,57 +1034,56 @@ TEST_F(NetEqImplTest, DecodingError) {
     InSequence sequence;  // Dummy variable.
     // Mock decoder works normally the first time.
     EXPECT_CALL(mock_decoder,
-                Decode(_, kPayloadLengthBytes, kSampleRateHz, _, _, _))
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
         .Times(3)
         .WillRepeatedly(
-            DoAll(SetArrayArgument<4>(dummy_output,
+            DoAll(SetArrayArgument<3>(dummy_output,
                                       dummy_output + kFrameLengthSamples),
-                  SetArgPointee<5>(AudioDecoder::kSpeech),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
                   Return(kFrameLengthSamples)))
         .RetiresOnSaturation();
 
     // Then mock decoder fails. A common reason for failure can be buffer being
     // too short
     EXPECT_CALL(mock_decoder,
-                Decode(_, kPayloadLengthBytes, kSampleRateHz, _, _, _))
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
         .WillOnce(Return(-1))
         .RetiresOnSaturation();
 
     // Mock decoder finally returns to normal.
     EXPECT_CALL(mock_decoder,
-                Decode(_, kPayloadLengthBytes, kSampleRateHz, _, _, _))
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
         .Times(2)
         .WillRepeatedly(
-            DoAll(SetArrayArgument<4>(dummy_output,
-                                  dummy_output + kFrameLengthSamples),
-                  SetArgPointee<5>(AudioDecoder::kSpeech),
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
                   Return(kFrameLengthSamples)));
   }
 
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
                             &mock_decoder, NetEqDecoder::kDecoderPCM16B,
-                            kPayloadType, kSampleRateHz));
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert packets.
   for (int i = 0; i < 6; ++i) {
     rtp_header.header.sequenceNumber += 1;
     rtp_header.header.timestamp += kFrameLengthSamples;
     EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                   kReceiveTime));
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   }
 
   // Pull audio.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Pull audio again. Decoder fails.
@@ -1110,7 +1093,7 @@ TEST_F(NetEqImplTest, DecodingError) {
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
   EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   // TODO(minyue): should NetEq better give kOutputPLC, since it is actually an
   // expansion.
   EXPECT_EQ(kOutputNormal, type);
@@ -1120,7 +1103,7 @@ TEST_F(NetEqImplTest, DecodingError) {
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputPLC, type);
 
   // Pull audio again, should behave normal.
@@ -1128,7 +1111,7 @@ TEST_F(NetEqImplTest, DecodingError) {
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   EXPECT_CALL(mock_decoder, Die());
@@ -1174,55 +1157,54 @@ TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
     InSequence sequence;  // Dummy variable.
     // Mock decoder works normally the first 2 times.
     EXPECT_CALL(mock_decoder,
-                Decode(_, kPayloadLengthBytes, kSampleRateHz, _, _, _))
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
         .Times(2)
         .WillRepeatedly(
-            DoAll(SetArrayArgument<4>(dummy_output,
+            DoAll(SetArrayArgument<3>(dummy_output,
                                       dummy_output + kFrameLengthSamples),
-                  SetArgPointee<5>(AudioDecoder::kComfortNoise),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
                   Return(kFrameLengthSamples)))
         .RetiresOnSaturation();
 
     // Then mock decoder fails. A common reason for failure can be buffer being
     // too short
-    EXPECT_CALL(mock_decoder, Decode(nullptr, 0, kSampleRateHz, _, _, _))
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
         .WillOnce(Return(-1))
         .RetiresOnSaturation();
 
     // Mock decoder finally returns to normal.
-    EXPECT_CALL(mock_decoder, Decode(nullptr, 0, kSampleRateHz, _, _, _))
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
         .Times(2)
         .WillRepeatedly(
-            DoAll(SetArrayArgument<4>(dummy_output,
-                                  dummy_output + kFrameLengthSamples),
-                  SetArgPointee<5>(AudioDecoder::kComfortNoise),
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
                   Return(kFrameLengthSamples)));
   }
 
   EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
                             &mock_decoder, NetEqDecoder::kDecoderPCM16B,
-                            kPayloadType, kSampleRateHz));
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert 2 packets. This will make netEq into codec internal CNG mode.
   for (int i = 0; i < 2; ++i) {
     rtp_header.header.sequenceNumber += 1;
     rtp_header.header.timestamp += kFrameLengthSamples;
     EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                   kReceiveTime));
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   }
 
   // Pull audio.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputCNG, type);
 
   // Pull audio again. Decoder fails.
@@ -1232,7 +1214,7 @@ TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
   EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   // TODO(minyue): should NetEq better give kOutputPLC, since it is actually an
   // expansion.
   EXPECT_EQ(kOutputCNG, type);
@@ -1242,10 +1224,19 @@ TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputCNG, type);
 
   EXPECT_CALL(mock_decoder, Die());
 }
 
+// Tests that the return value from last_output_sample_rate_hz() is equal to the
+// configured inital sample rate.
+TEST_F(NetEqImplTest, InitialLastOutputSampleRate) {
+  UseNoMocks();
+  config_.sample_rate_hz = 48000;
+  CreateInstance();
+  EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
+}
+
 }// namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 16fa04c234..34ca9ea856 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -191,8 +191,7 @@ struct NetEqNetworkStatsCheck {
                                                       frame_size_samples_,
                                                       &rtp_header_);
         if (!Lost(next_send_time)) {
-          InsertPacket(rtp_header_, payload_, kPayloadSizeByte,
-                       next_send_time);
+          InsertPacket(rtp_header_, payload_, next_send_time);
         }
       }
       GetOutputAudio(kMaxOutputSize, output_, &output_type);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 66874b8a50..d3f59ec668 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -16,19 +16,18 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
 struct TestParameters {
   int frame_size;
   int sample_rate;
-  int num_channels;
+  size_t num_channels;
 };
 
 // This is a parameterized test. The test parameters are supplied through a
@@ -127,11 +126,10 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
       default:
         FAIL() << "We shouldn't get here.";
     }
+    ASSERT_EQ(NetEq::kOK, neteq_mono_->RegisterPayloadType(mono_decoder, "mono",
+                                                           kPayloadTypeMono));
     ASSERT_EQ(NetEq::kOK,
-              neteq_mono_->RegisterPayloadType(mono_decoder,
-                                               kPayloadTypeMono));
-    ASSERT_EQ(NetEq::kOK,
-              neteq_->RegisterPayloadType(multi_decoder,
+              neteq_->RegisterPayloadType(multi_decoder, "multi-channel",
                                           kPayloadTypeMulti));
   }
 
@@ -165,7 +163,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
 
   void VerifyOutput(size_t num_samples) {
     for (size_t i = 0; i < num_samples; ++i) {
-      for (int j = 0; j < num_channels_; ++j) {
+      for (size_t j = 0; j < num_channels_; ++j) {
         ASSERT_EQ(output_[i], output_multi_channel_[i * num_channels_ + j]) <<
             "Diff in sample " << i << ", channel " << j << ".";
       }
@@ -196,14 +194,16 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
       while (time_now >= next_arrival_time) {
         // Insert packet in mono instance.
         ASSERT_EQ(NetEq::kOK,
-                  neteq_mono_->InsertPacket(rtp_header_mono_, encoded_,
-                                            payload_size_bytes_,
+                  neteq_mono_->InsertPacket(rtp_header_mono_,
+                                            rtc::ArrayView<const uint8_t>(
+                                                encoded_, payload_size_bytes_),
                                             next_arrival_time));
         // Insert packet in multi-channel instance.
-        ASSERT_EQ(NetEq::kOK,
-                  neteq_->InsertPacket(rtp_header_, encoded_multi_channel_,
-                                       multi_payload_size_bytes_,
-                                       next_arrival_time));
+        ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(
+                                  rtp_header_, rtc::ArrayView<const uint8_t>(
+                                                   encoded_multi_channel_,
+                                                   multi_payload_size_bytes_),
+                                  next_arrival_time));
         // Get next input packets (mono and multi-channel).
         do {
           next_send_time = GetNewPackets();
@@ -214,12 +214,12 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
       NetEqOutputType output_type;
       // Get audio from mono instance.
       size_t samples_per_channel;
-      int num_channels;
+      size_t num_channels;
       EXPECT_EQ(NetEq::kOK,
                 neteq_mono_->GetAudio(kMaxBlockSize, output_,
                                       &samples_per_channel, &num_channels,
                                       &output_type));
-      EXPECT_EQ(1, num_channels);
+      EXPECT_EQ(1u, num_channels);
       EXPECT_EQ(output_size_samples_, samples_per_channel);
       // Get audio from multi-channel instance.
       ASSERT_EQ(NetEq::kOK,
@@ -239,7 +239,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
     }
   }
 
-  const int num_channels_;
+  const size_t num_channels_;
   const int sample_rate_hz_;
   const int samples_per_ms_;
   const int frame_size_ms_;
@@ -275,7 +275,12 @@ class NetEqStereoTestNoJitter : public NetEqStereoTest {
   }
 };
 
-TEST_P(NetEqStereoTestNoJitter, DISABLED_ON_ANDROID(RunTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_RunTest DISABLED_RunTest
+#else
+#define MAYBE_RunTest RunTest
+#endif
+TEST_P(NetEqStereoTestNoJitter, MAYBE_RunTest) {
   RunTest(8);
 }
 
@@ -300,7 +305,7 @@ class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
   double drift_factor;
 };
 
-TEST_P(NetEqStereoTestPositiveDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestPositiveDrift, MAYBE_RunTest) {
   RunTest(100);
 }
 
@@ -313,7 +318,7 @@ class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
   }
 };
 
-TEST_P(NetEqStereoTestNegativeDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestNegativeDrift, MAYBE_RunTest) {
   RunTest(100);
 }
 
@@ -341,7 +346,7 @@ class NetEqStereoTestDelays : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestDelays, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestDelays, MAYBE_RunTest) {
   RunTest(1000);
 }
 
@@ -360,7 +365,10 @@ class NetEqStereoTestLosses : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestLosses, DISABLED_ON_ANDROID(RunTest)) {
+// TODO(pbos): Enable on non-Android, this went failing while being accidentally
+// disabled on all platforms and not just Android.
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5387
+TEST_P(NetEqStereoTestLosses, DISABLED_RunTest) {
   RunTest(100);
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_tests.gypi b/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
index ee9583ab85..f02d3deee9 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
+++ b/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
@@ -39,6 +39,21 @@
           'defines': [
           ],
         }, # neteq_rtpplay
+        {
+          'target_name': 'neteq_unittest_proto',
+          'type': 'static_library',
+          'sources': [
+            'neteq_unittest.proto',
+          ],
+          'variables': {
+            'proto_in_dir': '.',
+            # Workaround to protect against gyp's pathname relativization when
+            # this file is included by modules.gyp.
+            'proto_out_protected': 'webrtc/audio_coding/neteq',
+            'proto_out_dir': '<(proto_out_protected)',
+          },
+          'includes': ['../../../build/protoc.gypi',],
+        },
       ],
     }],
   ],
@@ -56,6 +71,7 @@
         'isac',
         'neteq_test_tools',  # Test helpers
         'pcm16b',
+        'webrtc_opus',
       ],
       'defines': [
         'CODEC_ILBC',
@@ -72,6 +88,7 @@
         'CODEC_CNGCODEC32',
         'CODEC_ATEVENT_DECODE',
         'CODEC_RED',
+        'CODEC_OPUS',
       ],
       'include_dirs': [
         'include',
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 4340f54975..8d52c615da 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -28,29 +28,91 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "webrtc/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+#endif
+
 DEFINE_bool(gen_ref, false, "Generate reference files.");
 
-namespace webrtc {
+namespace {
 
-static bool IsAllZero(const int16_t* buf, size_t buf_length) {
+bool IsAllZero(const int16_t* buf, size_t buf_length) {
   bool all_zero = true;
   for (size_t n = 0; n < buf_length && all_zero; ++n)
     all_zero = buf[n] == 0;
   return all_zero;
 }
 
-static bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
+bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
   bool all_non_zero = true;
   for (size_t n = 0; n < buf_length && all_non_zero; ++n)
     all_non_zero = buf[n] != 0;
   return all_non_zero;
 }
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
+             webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
+  stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
+  stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
+  stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
+  stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
+  stats->set_packet_discard_rate(stats_raw.packet_discard_rate);
+  stats->set_expand_rate(stats_raw.expand_rate);
+  stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
+  stats->set_preemptive_rate(stats_raw.preemptive_rate);
+  stats->set_accelerate_rate(stats_raw.accelerate_rate);
+  stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
+  stats->set_clockdrift_ppm(stats_raw.clockdrift_ppm);
+  stats->set_added_zero_samples(stats_raw.added_zero_samples);
+  stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
+  stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
+  stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
+  stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
+}
+
+void Convert(const webrtc::RtcpStatistics& stats_raw,
+             webrtc::neteq_unittest::RtcpStatistics* stats) {
+  stats->set_fraction_lost(stats_raw.fraction_lost);
+  stats->set_cumulative_lost(stats_raw.cumulative_lost);
+  stats->set_extended_max_sequence_number(
+      stats_raw.extended_max_sequence_number);
+  stats->set_jitter(stats_raw.jitter);
+}
+
+void WriteMessage(FILE* file, const std::string& message) {
+  int32_t size = message.length();
+  ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  if (size <= 0)
+    return;
+  ASSERT_EQ(static_cast<size_t>(size),
+            fwrite(message.data(), sizeof(char), size, file));
+}
+
+void ReadMessage(FILE* file, std::string* message) {
+  int32_t size;
+  ASSERT_EQ(1u, fread(&size, sizeof(size), 1, file));
+  if (size <= 0)
+    return;
+  rtc::scoped_ptr<char[]> buffer(new char[size]);
+  ASSERT_EQ(static_cast<size_t>(size),
+            fread(buffer.get(), sizeof(char), size, file));
+  message->assign(buffer.get(), size);
+}
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+
+}  // namespace
+
+namespace webrtc {
+
 class RefFiles {
  public:
   RefFiles(const std::string& input_file, const std::string& output_file);
@@ -128,92 +190,84 @@ void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
   }
 }
 
-void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats) {
-  if (output_fp_) {
-    ASSERT_EQ(1u, fwrite(&stats, sizeof(NetEqNetworkStatistics), 1,
-                         output_fp_));
-  }
+void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats_raw) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!output_fp_)
+    return;
+  neteq_unittest::NetEqNetworkStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  WriteMessage(output_fp_, stats_string);
+#else
+  FAIL() << "Writing to reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
 void RefFiles::ReadFromFileAndCompare(
     const NetEqNetworkStatistics& stats) {
-  // TODO(minyue): Update resource/audio_coding/neteq_network_stats.dat and
-  // resource/audio_coding/neteq_network_stats_win32.dat.
-  struct NetEqNetworkStatisticsOld {
-    uint16_t current_buffer_size_ms;  // Current jitter buffer size in ms.
-    uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
-    uint16_t jitter_peaks_found;  // 1 if adding extra delay due to peaky
-                                  // jitter; 0 otherwise.
-    uint16_t packet_loss_rate;  // Loss rate (network + late) in Q14.
-    uint16_t packet_discard_rate;  // Late loss rate in Q14.
-    uint16_t expand_rate;  // Fraction (of original stream) of synthesized
-                           // audio inserted through expansion (in Q14).
-    uint16_t preemptive_rate;  // Fraction of data inserted through pre-emptive
-                               // expansion (in Q14).
-    uint16_t accelerate_rate;  // Fraction of data removed through acceleration
-                               // (in Q14).
-    int32_t clockdrift_ppm;  // Average clock-drift in parts-per-million
-                             // (positive or negative).
-    int added_zero_samples;  // Number of zero samples added in "off" mode.
-  };
-  if (input_fp_) {
-    // Read from ref file.
-    size_t stat_size = sizeof(NetEqNetworkStatisticsOld);
-    NetEqNetworkStatisticsOld ref_stats;
-    ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
-    // Compare
-    ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms);
-    ASSERT_EQ(stats.preferred_buffer_size_ms,
-              ref_stats.preferred_buffer_size_ms);
-    ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found);
-    ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate);
-    ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate);
-    ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate);
-    ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate);
-    ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate);
-    ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm);
-    ASSERT_EQ(stats.added_zero_samples,
-              static_cast<size_t>(ref_stats.added_zero_samples));
-    ASSERT_EQ(stats.secondary_decoded_rate, 0);
-    ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate);
-  }
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!input_fp_)
+    return;
+
+  std::string stats_string;
+  ReadMessage(input_fp_, &stats_string);
+  neteq_unittest::NetEqNetworkStatistics ref_stats;
+  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
+
+  // Compare
+  ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms());
+  ASSERT_EQ(stats.preferred_buffer_size_ms,
+            ref_stats.preferred_buffer_size_ms());
+  ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found());
+  ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate());
+  ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate());
+  ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate());
+  ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate());
+  ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate());
+  ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm());
+  ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples());
+  ASSERT_EQ(stats.secondary_decoded_rate, ref_stats.secondary_decoded_rate());
+  ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate());
+#else
+  FAIL() << "Reading from reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
-void RefFiles::WriteToFile(const RtcpStatistics& stats) {
-  if (output_fp_) {
-    ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
-                         output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.cumulative_lost),
-                         sizeof(stats.cumulative_lost), 1, output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.extended_max_sequence_number),
-                         sizeof(stats.extended_max_sequence_number), 1,
-                         output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
-                         output_fp_));
-  }
+void RefFiles::WriteToFile(const RtcpStatistics& stats_raw) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!output_fp_)
+    return;
+  neteq_unittest::RtcpStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  WriteMessage(output_fp_, stats_string);
+#else
+  FAIL() << "Writing to reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
-void RefFiles::ReadFromFileAndCompare(
-    const RtcpStatistics& stats) {
-  if (input_fp_) {
-    // Read from ref file.
-    RtcpStatistics ref_stats;
-    ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
-                        sizeof(ref_stats.fraction_lost), 1, input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.cumulative_lost),
-                        sizeof(ref_stats.cumulative_lost), 1, input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.extended_max_sequence_number),
-                        sizeof(ref_stats.extended_max_sequence_number), 1,
-                        input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
-                        input_fp_));
-    // Compare
-    ASSERT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
-    ASSERT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost);
-    ASSERT_EQ(ref_stats.extended_max_sequence_number,
-              stats.extended_max_sequence_number);
-    ASSERT_EQ(ref_stats.jitter, stats.jitter);
-  }
+void RefFiles::ReadFromFileAndCompare(const RtcpStatistics& stats) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!input_fp_)
+    return;
+  std::string stats_string;
+  ReadMessage(input_fp_, &stats_string);
+  neteq_unittest::RtcpStatistics ref_stats;
+  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
+
+  // Compare
+  ASSERT_EQ(stats.fraction_lost, ref_stats.fraction_lost());
+  ASSERT_EQ(stats.cumulative_lost, ref_stats.cumulative_lost());
+  ASSERT_EQ(stats.extended_max_sequence_number,
+            ref_stats.extended_max_sequence_number());
+  ASSERT_EQ(stats.jitter, ref_stats.jitter());
+#else
+  FAIL() << "Reading from reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
 class NetEqDecodingTest : public ::testing::Test {
@@ -224,7 +278,8 @@ class NetEqDecodingTest : public ::testing::Test {
   static const size_t kBlockSize8kHz = kTimeStepMs * 8;
   static const size_t kBlockSize16kHz = kTimeStepMs * 16;
   static const size_t kBlockSize32kHz = kTimeStepMs * 32;
-  static const size_t kMaxBlockSize = kBlockSize32kHz;
+  static const size_t kBlockSize48kHz = kTimeStepMs * 48;
+  static const size_t kMaxBlockSize = kBlockSize48kHz;
   static const int kInitSampleRateHz = 8000;
 
   NetEqDecodingTest();
@@ -234,10 +289,12 @@ class NetEqDecodingTest : public ::testing::Test {
   void LoadDecoders();
   void OpenInputFile(const std::string &rtp_file);
   void Process(size_t* out_len);
+
   void DecodeAndCompare(const std::string& rtp_file,
                         const std::string& ref_file,
                         const std::string& stat_ref_file,
                         const std::string& rtcp_ref_file);
+
   static void PopulateRtpInfo(int frame_index,
                               int timestamp,
                               WebRtcRTPHeader* rtp_info);
@@ -304,32 +361,45 @@ void NetEqDecodingTest::TearDown() {
 
 void NetEqDecodingTest::LoadDecoders() {
   // Load PCMu.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMu, 0));
+  ASSERT_EQ(0,
+            neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMu, "pcmu", 0));
   // Load PCMa.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMa, 8));
+  ASSERT_EQ(0,
+            neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMa, "pcma", 8));
 #ifdef WEBRTC_CODEC_ILBC
   // Load iLBC.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderILBC, 102));
+  ASSERT_EQ(
+      0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderILBC, "ilbc", 102));
 #endif
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
   // Load iSAC.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, 103));
+  ASSERT_EQ(
+      0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac", 103));
 #endif
 #ifdef WEBRTC_CODEC_ISAC
   // Load iSAC SWB.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISACswb, 104));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISACswb,
+                                           "isac-swb", 104));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderOpus,
+                                           "opus", 111));
 #endif
   // Load PCM16B nb.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16B, 93));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16B,
+                                           "pcm16-nb", 93));
   // Load PCM16B wb.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb, 94));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb,
+                                           "pcm16-wb", 94));
   // Load PCM16B swb32.
-  ASSERT_EQ(
-      0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bswb32kHz, 95));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                                           "pcm16-swb32", 95));
   // Load CNG 8 kHz.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb, 13));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
+                                           "cng-nb", 13));
   // Load CNG 16 kHz.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb, 98));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
+                                           "cng-wb", 98));
 }
 
 void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
@@ -343,10 +413,11 @@ void NetEqDecodingTest::Process(size_t* out_len) {
       WebRtcRTPHeader rtp_header;
       packet_->ConvertHeader(&rtp_header);
       ASSERT_EQ(0, neteq_->InsertPacket(
-                       rtp_header, packet_->payload(),
-                       packet_->payload_length_bytes(),
-                       static_cast<uint32_t>(
-                           packet_->time_ms() * (output_sample_rate_ / 1000))));
+                       rtp_header,
+                       rtc::ArrayView<const uint8_t>(
+                           packet_->payload(), packet_->payload_length_bytes()),
+                       static_cast<uint32_t>(packet_->time_ms() *
+                                             (output_sample_rate_ / 1000))));
     }
     // Get next packet.
     packet_.reset(rtp_source_->NextPacket());
@@ -354,13 +425,15 @@ void NetEqDecodingTest::Process(size_t* out_len) {
 
   // Get audio from NetEq.
   NetEqOutputType type;
-  int num_channels;
+  size_t num_channels;
   ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
                                 &num_channels, &type));
   ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
               (*out_len == kBlockSize16kHz) ||
-              (*out_len == kBlockSize32kHz));
+              (*out_len == kBlockSize32kHz) ||
+              (*out_len == kBlockSize48kHz));
   output_sample_rate_ = static_cast<int>(*out_len / 10 * 1000);
+  EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
 
   // Increase time.
   sim_clock_ += kTimeStepMs;
@@ -442,17 +515,17 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISAC)) && \
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
+    (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
-#define IF_ALL_CODECS(x) x
+#define MAYBE_TestBitExactness TestBitExactness
 #else
-#define IF_ALL_CODECS(x) DISABLED_##x
+#define MAYBE_TestBitExactness DISABLED_TestBitExactness
 #endif
-
-TEST_F(NetEqDecodingTest,
-       DISABLED_ON_IOS(DISABLED_ON_ANDROID(IF_ALL_CODECS(TestBitExactness)))) {
-  const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
-      "resources/audio_coding/neteq_universal_new.rtp";
+TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
   // are identical. The latter could have been removed, but if clients still
   // have a copy of the file, the test will fail.
@@ -480,6 +553,34 @@ TEST_F(NetEqDecodingTest,
   }
 }
 
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
+    defined(WEBRTC_CODEC_OPUS)
+#define MAYBE_TestOpusBitExactness TestOpusBitExactness
+#else
+#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
+  const std::string input_ref_file =
+      webrtc::test::ResourcePath("audio_coding/neteq4_opus_ref", "pcm");
+  const std::string network_stat_ref_file =
+      webrtc::test::ResourcePath("audio_coding/neteq4_opus_network_stats",
+                                 "dat");
+  const std::string rtcp_stat_ref_file =
+      webrtc::test::ResourcePath("audio_coding/neteq4_opus_rtcp_stats", "dat");
+
+  if (FLAGS_gen_ref) {
+    DecodeAndCompare(input_rtp_file, "", "", "");
+  } else {
+    DecodeAndCompare(input_rtp_file,
+                     input_ref_file,
+                     network_stat_ref_file,
+                     rtcp_stat_ref_file);
+  }
+}
+
 // Use fax mode to avoid time-scaling. This is to simplify the testing of
 // packet waiting times in the packet buffer.
 class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
@@ -495,22 +596,19 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   for (size_t i = 0; i < num_frames; ++i) {
-    uint16_t payload[kSamples] = {0};
+    const uint8_t payload[kPayloadBytes] = {0};
     WebRtcRTPHeader rtp_info;
     rtp_info.header.sequenceNumber = i;
     rtp_info.header.timestamp = i * kSamples;
     rtp_info.header.ssrc = 0x1234;  // Just an arbitrary SSRC.
     rtp_info.header.payloadType = 94;  // PCM16b WB codec.
     rtp_info.header.markerBit = 0;
-    ASSERT_EQ(0, neteq_->InsertPacket(
-        rtp_info,
-        reinterpret_cast<uint8_t*>(payload),
-        kPayloadBytes, 0));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   }
   // Pull out all data.
   for (size_t i = 0; i < num_frames; ++i) {
     size_t out_len;
-    int num_channels;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
@@ -549,13 +647,13 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
     size_t out_len;
-    int num_channels;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
@@ -580,13 +678,13 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
     size_t out_len;
-    int num_channels;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
@@ -611,7 +709,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
   double next_input_time_ms = 0.0;
   double t_ms;
   size_t out_len;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
 
   // Insert speech for 5 seconds.
@@ -623,7 +721,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
@@ -649,7 +747,9 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(
+                       rtp_info,
+                       rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
@@ -696,7 +796,9 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(
+                       rtp_info,
+                       rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += kCngPeriodMs * drift_factor;
@@ -712,7 +814,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += kFrameSizeMs * drift_factor;
@@ -823,31 +925,30 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) {
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 1;  // Not registered as a decoder.
-  EXPECT_EQ(NetEq::kFail,
-            neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload, 0));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_DecoderError DISABLED_DecoderError
 #else
-#define IF_ISAC(x) DISABLED_##x
+#define MAYBE_DecoderError DecoderError
 #endif
-
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 103;  // iSAC, but the payload is invalid.
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < kMaxBlockSize; ++i) {
     out_data_[i] = 1;
   }
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   EXPECT_EQ(NetEq::kFail,
             neteq_->GetAudio(kMaxBlockSize, out_data_,
@@ -872,6 +973,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
     EXPECT_EQ(1, out_data_[i]);
   }
 }
+#endif
 
 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   NetEqOutputType type;
@@ -880,7 +982,7 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   for (size_t i = 0; i < kMaxBlockSize; ++i) {
     out_data_[i] = 1;
   }
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
                                 &samples_per_channel,
@@ -894,6 +996,8 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(0, out_data_[i]);
   }
+  // Verify that the sample rate did not change from the initial configuration.
+  EXPECT_EQ(config_.sample_rate_hz, neteq_->last_output_sample_rate_hz());
 }
 
 class NetEqBgnTest : public NetEqDecodingTest {
@@ -934,27 +1038,29 @@ class NetEqBgnTest : public NetEqDecodingTest {
     PopulateRtpInfo(0, 0, &rtp_info);
     rtp_info.header.payloadType = payload_type;
 
-    int number_channels = 0;
+    size_t number_channels = 0;
     size_t samples_per_channel = 0;
 
     uint32_t receive_timestamp = 0;
     for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
-      size_t enc_len_bytes = WebRtcPcm16b_Encode(
-          input.GetNextBlock(), expected_samples_per_channel, payload);
+      auto block = input.GetNextBlock();
+      ASSERT_EQ(expected_samples_per_channel, block.size());
+      size_t enc_len_bytes =
+          WebRtcPcm16b_Encode(block.data(), block.size(), payload);
       ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
 
       number_channels = 0;
       samples_per_channel = 0;
-      ASSERT_EQ(0,
-                neteq_->InsertPacket(rtp_info, payload, enc_len_bytes,
-                                     receive_timestamp));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, enc_len_bytes),
+                                        receive_timestamp));
       ASSERT_EQ(0,
                 neteq_->GetAudio(kBlockSize32kHz,
                                  output,
                                  &samples_per_channel,
                                  &number_channels,
                                  &type));
-      ASSERT_EQ(1, number_channels);
+      ASSERT_EQ(1u, number_channels);
       ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
       ASSERT_EQ(kOutputNormal, type);
 
@@ -976,7 +1082,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
                                &samples_per_channel,
                                &number_channels,
                                &type));
-    ASSERT_EQ(1, number_channels);
+    ASSERT_EQ(1u, number_channels);
     ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
 
     // To be able to test the fading of background noise we need at lease to
@@ -997,7 +1103,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
                                  &samples_per_channel,
                                  &number_channels,
                                  &type));
-      ASSERT_EQ(1, number_channels);
+      ASSERT_EQ(1u, number_channels);
       ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
       if (type == kOutputPLCtoCNG) {
         plc_to_cng = true;
@@ -1065,7 +1171,8 @@ TEST_F(NetEqBgnTestFade, RunTest) {
   CheckBgn(32000);
 }
 
-TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   WebRtcRTPHeader rtp_info;
   uint32_t receive_timestamp = 0;
   // For the readability use the following payloads instead of the defaults of
@@ -1081,20 +1188,20 @@ TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
 
   // Register decoders.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb,
-                                           kPcm16WbPayloadType));
+                                           "pcm16-wb", kPcm16WbPayloadType));
   ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
-                                           kCngNbPayloadType));
+                                           "cng-nb", kCngNbPayloadType));
   ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
-                                           kCngWbPayloadType));
+                                           "cng-wb", kCngWbPayloadType));
   ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb32kHz,
-                                           kCngSwb32PayloadType));
+                                           "cng-swb32", kCngSwb32PayloadType));
   ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb48kHz,
-                                           kCngSwb48PayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderAVT,
+                                           "cng-swb48", kCngSwb48PayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderAVT, "avt",
                                            kAvtPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderRED,
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderRED, "red",
                                            kRedPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC,
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac",
                                            kIsacPayloadType));
 
   PopulateRtpInfo(0, 0, &rtp_info);
@@ -1106,8 +1213,7 @@ TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
   // Payload length of 10 ms PCM16 16 kHz.
   const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
   uint8_t payload[kPayloadBytes] = {0};
-  ASSERT_EQ(0, neteq_->InsertPacket(
-      rtp_info, payload, kPayloadBytes, receive_timestamp));
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
 
   // Next packet. Last packet contained 10 ms audio.
   rtp_info.header.sequenceNumber++;
@@ -1145,6 +1251,7 @@ TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
   --rtp_info.header.ssrc;
   EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
 }
+#endif
 
 // First insert several noise like packets, then sync-packets. Decoding all
 // packets should not produce error, statistics should not show any packet loss
@@ -1165,17 +1272,16 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   // Insert some packets which decode to noise. We are not interested in
   // actual decoded values.
   NetEqOutputType output_type;
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   uint32_t receive_timestamp = 0;
   for (int n = 0; n < 100; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
 
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
@@ -1193,7 +1299,7 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     if (n > algorithmic_frame_delay) {
       EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
     }
@@ -1205,8 +1311,7 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   // We insert regular packets, if sync packet are not correctly buffered then
   // network statistics would show some packet loss.
   for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
@@ -1243,18 +1348,17 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
   // Insert some packets which decode to noise. We are not interested in
   // actual decoded values.
   NetEqOutputType output_type;
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   uint32_t receive_timestamp = 0;
   int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
   for (int n = 0; n < algorithmic_frame_delay; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
@@ -1281,8 +1385,7 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
 
   // Insert.
   for (int n = 0; n < kNumSyncPackets; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
@@ -1294,7 +1397,7 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
   }
 }
@@ -1312,7 +1415,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
   const size_t kPayloadBytes = kSamples * sizeof(int16_t);
   double next_input_time_ms = 0.0;
   int16_t decoded[kBlockSize16kHz];
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   NetEqOutputType output_type;
   uint32_t receive_timestamp = 0;
@@ -1334,8 +1437,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
       if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
         // This sequence number was not in the set to drop. Insert it.
         ASSERT_EQ(0,
-                  neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                       receive_timestamp));
+                  neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
         ++packets_inserted;
       }
       NetEqNetworkStatistics network_stats;
@@ -1366,7 +1468,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
 
     // Expect delay (in samples) to be less than 2 packets.
     EXPECT_LE(timestamp - PlayoutTimestamp(),
@@ -1417,13 +1519,13 @@ void NetEqDecodingTest::DuplicateCng() {
   // Insert three speech packets. Three are needed to get the frame length
   // correct.
   size_t out_len;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
@@ -1442,7 +1544,9 @@ void NetEqDecodingTest::DuplicateCng() {
   size_t payload_len;
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   // This is the first time this CNG packet is inserted.
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+  ASSERT_EQ(
+      0, neteq_->InsertPacket(
+             rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio once and make sure CNG is played.
   ASSERT_EQ(0,
@@ -1454,7 +1558,9 @@ void NetEqDecodingTest::DuplicateCng() {
 
   // Insert the same CNG packet again. Note that at this point it is old, since
   // we have already decoded the first copy of it.
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+  ASSERT_EQ(
+      0, neteq_->InsertPacket(
+             rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
   // we have already pulled out CNG once.
@@ -1472,7 +1578,7 @@ void NetEqDecodingTest::DuplicateCng() {
   ++seq_no;
   timestamp += kCngPeriodSamples;
   PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
 
   // Pull audio once and verify that the output is speech again.
   ASSERT_EQ(0,
@@ -1507,14 +1613,16 @@ TEST_F(NetEqDecodingTest, CngFirst) {
   WebRtcRTPHeader rtp_info;
 
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
-  ASSERT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+  ASSERT_EQ(
+      NetEq::kOK,
+      neteq_->InsertPacket(
+          rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
   ++seq_no;
   timestamp += kCngPeriodSamples;
 
   // Pull audio once and make sure CNG is played.
   size_t out_len;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                 &num_channels, &type));
@@ -1524,7 +1632,7 @@ TEST_F(NetEqDecodingTest, CngFirst) {
   // Insert some speech packets.
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.proto b/webrtc/modules/audio_coding/neteq/neteq_unittest.proto
new file mode 100644
index 0000000000..4b59848eb2
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.proto
@@ -0,0 +1,29 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.neteq_unittest;
+
+message NetEqNetworkStatistics {
+  optional uint32 current_buffer_size_ms = 1;
+  optional uint32 preferred_buffer_size_ms = 2;
+  optional uint32 jitter_peaks_found = 3;
+  optional uint32 packet_loss_rate = 4;
+  optional uint32 packet_discard_rate = 5;
+  optional uint32 expand_rate = 6;
+  optional uint32 speech_expand_rate = 7;
+  optional uint32 preemptive_rate = 8;
+  optional uint32 accelerate_rate = 9;
+  optional uint32 secondary_decoded_rate = 10;
+  optional int32 clockdrift_ppm = 11;
+  optional uint64 added_zero_samples = 12;
+  optional int32 mean_waiting_time_ms = 13;
+  optional int32 median_waiting_time_ms = 14;
+  optional int32 min_waiting_time_ms = 15;
+  optional int32 max_waiting_time_ms = 16;
+}
+
+message RtcpStatistics {
+  optional uint32 fraction_lost = 1;
+  optional uint32 cumulative_lost = 2;
+  optional uint32 extended_max_sequence_number = 3;
+  optional uint32 jitter = 4;
+}
\ No newline at end of file
diff --git a/webrtc/modules/audio_coding/neteq/normal.cc b/webrtc/modules/audio_coding/neteq/normal.cc
index ebecbf94bd..1b888f70d1 100644
--- a/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/webrtc/modules/audio_coding/neteq/normal.cc
@@ -16,7 +16,7 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
diff --git a/webrtc/modules/audio_coding/neteq/packet.h b/webrtc/modules/audio_coding/neteq/packet.h
index 723ed8b0a3..64b325e027 100644
--- a/webrtc/modules/audio_coding/neteq/packet.h
+++ b/webrtc/modules/audio_coding/neteq/packet.h
@@ -13,7 +13,7 @@
 
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc b/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
index bf26a8f517..07c4bac0b6 100644
--- a/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
@@ -310,10 +310,10 @@ TEST(RedPayloadSplitter, CheckRedPayloads) {
   // easier to just register the payload types and let the actual implementation
   // do its job.
   DecoderDatabase decoder_database;
-  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderCNGnb);
-  decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu);
-  decoder_database.RegisterPayload(2, NetEqDecoder::kDecoderAVT);
-  decoder_database.RegisterPayload(3, NetEqDecoder::kDecoderILBC);
+  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderCNGnb, "cng-nb");
+  decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
+  decoder_database.RegisterPayload(2, NetEqDecoder::kDecoderAVT, "avt");
+  decoder_database.RegisterPayload(3, NetEqDecoder::kDecoderILBC, "ilbc");
 
   PayloadSplitter splitter;
   splitter.CheckRedPayloads(&packet_list, decoder_database);
@@ -745,8 +745,8 @@ TEST(FecPayloadSplitter, MixedPayload) {
   PacketList packet_list;
   DecoderDatabase decoder_database;
 
-  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderOpus);
-  decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu);
+  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderOpus, "opus");
+  decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
 
   Packet* packet = CreatePacket(0, 10, 0xFF, true);
   packet_list.push_back(packet);
@@ -802,7 +802,7 @@ TEST(FecPayloadSplitter, EmbedFecInRed) {
 
   const int kTimestampOffset = 20 * 48;  // 20 ms * 48 kHz.
   uint8_t payload_types[] = {0, 0};
-  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderOpus);
+  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderOpus, "opus");
   Packet* packet = CreateRedPayload(2, payload_types, kTimestampOffset, true);
   packet_list.push_back(packet);
 
diff --git a/webrtc/modules/audio_coding/neteq/rtcp.cc b/webrtc/modules/audio_coding/neteq/rtcp.cc
index cf8e0280bb..7ef40bc814 100644
--- a/webrtc/modules/audio_coding/neteq/rtcp.cc
+++ b/webrtc/modules/audio_coding/neteq/rtcp.cc
@@ -15,7 +15,7 @@
 #include <algorithm>
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index e6a6fbf705..8f873762c5 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -50,7 +50,7 @@ void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
 }
 
 void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
-  RTC_HISTOGRAM_COUNTS(uma_name_, value, 1, max_value_, 50);
+  RTC_HISTOGRAM_COUNTS_SPARSE(uma_name_, value, 1, max_value_, 50);
 }
 
 StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
@@ -187,9 +187,9 @@ void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
 }
 
 void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
-  RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
-                       outage_duration_ms, 1 /* min */, 2000 /* max */,
-                       100 /* bucket count */);
+  RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.DelayedPacketOutageEventMs",
+                              outage_duration_ms, 1 /* min */, 2000 /* max */,
+                              100 /* bucket count */);
   delayed_packet_outage_counter_.RegisterSample();
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 3fbce8be5c..56ed72fcee 100644
--- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -14,7 +14,7 @@
 #include <map>
 #include <stdio.h>
 #include "webrtc/typedefs.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 enum stereoModes {
     stereoModeMono,
diff --git a/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h b/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
index c46a3daece..aba525b162 100644
--- a/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
+++ b/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
@@ -39,7 +39,7 @@
 #define NETEQ_CODEC_G722_1_16_PT		108
 #define NETEQ_CODEC_G722_1_24_PT		109
 #define NETEQ_CODEC_G722_1_32_PT		110
-#define NETEQ_CODEC_SC3_PT				111
+#define NETEQ_CODEC_OPUS_PT             111
 #define NETEQ_CODEC_AMR_PT				112
 #define NETEQ_CODEC_GSMEFR_PT			113
 //#define NETEQ_CODEC_ILBCRCU_PT			114
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index cbb7436152..45586ee111 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -25,7 +25,9 @@
 
 #include <algorithm>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/typedefs.h"
+
 // needed for NetEqDecoder
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
@@ -36,6 +38,10 @@
 
 #include "PayloadTypes.h"
 
+namespace {
+const size_t kRtpDataSize = 8000;
+}
+
 /*********************/
 /* Misc. definitions */
 /*********************/
@@ -126,10 +132,10 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
 #include "webrtc_vad.h"
 
 #if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC))
-#include "pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #endif
 #ifdef CODEC_G711
-#include "g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 #endif
 #ifdef CODEC_G729
 #include "G729Interface.h"
@@ -146,19 +152,19 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
 #include "AMRWBCreation.h"
 #endif
 #ifdef CODEC_ILBC
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 #endif
 #if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
-#include "isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 #endif
 #ifdef NETEQ_ISACFIX_CODEC
-#include "isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
 #ifdef CODEC_ISAC
 #error Cannot have both ISAC and ISACfix defined. Please de-select one.
 #endif
 #endif
 #ifdef CODEC_G722
-#include "g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 #endif
 #ifdef CODEC_G722_1_24
 #include "G722_1Interface.h"
@@ -188,11 +194,14 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
 #endif
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-#include "webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #endif
 #if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
 #include "SpeexInterface.h"
 #endif
+#ifdef CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#endif
 
 /***********************************/
 /* Global codec instance variables */
@@ -264,6 +273,9 @@ SPEEX_encinst_t* SPEEX8enc_inst[2];
 #ifdef CODEC_SPEEX_16
 SPEEX_encinst_t* SPEEX16enc_inst[2];
 #endif
+#ifdef CODEC_OPUS
+OpusEncInst* opus_inst[2];
+#endif
 
 int main(int argc, char* argv[]) {
   size_t packet_size;
@@ -275,7 +287,7 @@ int main(int argc, char* argv[]) {
   int useRed = 0;
   size_t len, enc_len;
   int16_t org_data[4000];
-  unsigned char rtp_data[8000];
+  unsigned char rtp_data[kRtpDataSize];
   int16_t seqNo = 0xFFF;
   uint32_t ssrc = 1235412312;
   uint32_t timestamp = 0xAC1245;
@@ -286,12 +298,12 @@ int main(int argc, char* argv[]) {
   uint32_t red_TS[2] = {0};
   uint16_t red_len[2] = {0};
   size_t RTPheaderLen = 12;
-  uint8_t red_data[8000];
+  uint8_t red_data[kRtpDataSize];
 #ifdef INSERT_OLD_PACKETS
   uint16_t old_length, old_plen;
   size_t old_enc_len;
   int first_old_packet = 1;
-  unsigned char old_rtp_data[8000];
+  unsigned char old_rtp_data[kRtpDataSize];
   size_t packet_age = 0;
 #endif
 #ifdef INSERT_DTMF_PACKETS
@@ -429,6 +441,10 @@ int main(int argc, char* argv[]) {
     printf("             : red_isac     Redundancy RTP packet with 2*iSAC "
            "frames\n");
 #endif
+#endif  // CODEC_RED
+#ifdef CODEC_OPUS
+    printf("             : opus         Opus codec with FEC (48kHz, 32kbps, FEC"
+        " on and tuned for 5%% packet losses)\n");
 #endif
     printf("\n");
 
@@ -880,6 +896,10 @@ void NetEQTest_GetCodec_and_PT(char* name,
     *PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
     *fs = 16000;
     *useRed = 1;
+  } else if (!strcmp(name, "opus")) {
+    *codec = webrtc::NetEqDecoder::kDecoderOpus;
+    *PT = NETEQ_CODEC_OPUS_PT; /* this will be the PT for the sub-headers */
+    *fs = 48000;
   } else {
     printf("Error: Not a supported codec (%s)\n", name);
     exit(0);
@@ -1411,12 +1431,23 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
         }
         break;
 #endif
+#ifdef CODEC_OPUS
+      case webrtc::NetEqDecoder::kDecoderOpus:
+        ok = WebRtcOpus_EncoderCreate(&opus_inst[k], 1, 0);
+        if (ok != 0) {
+          printf("Error: Couldn't allocate memory for Opus encoding "
+                 "instance\n");
+          exit(0);
+        }
+        WebRtcOpus_EnableFec(opus_inst[k]);
+        WebRtcOpus_SetPacketLossRate(opus_inst[k], 5);
+        break;
+#endif
       default:
         printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
         exit(0);
         break;
     }
-
     if (ok != 0) {
       return (ok);
     }
@@ -1543,6 +1574,11 @@ int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
         WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
         break;
 #endif
+#ifdef CODEC_OPUS
+      case webrtc::NetEqDecoder::kDecoderOpus:
+        WebRtcOpus_EncoderFree(opus_inst[k]);
+        break;
+#endif
       default:
         printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
         exit(0);
@@ -1687,6 +1723,11 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
       cdlen = static_cast<size_t>(res);
     }
 #endif
+#ifdef CODEC_OPUS
+    cdlen = WebRtcOpus_Encode(opus_inst[k], indata, frameLen, kRtpDataSize - 12,
+                              encoded);
+    RTC_CHECK_GT(cdlen, 0u);
+#endif
     indata += frameLen;
     encoded += cdlen;
     totalLen += cdlen;
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
index 2042e0d2b8..0c09e92b4d 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -11,7 +11,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -50,7 +50,7 @@ class NetEqIlbcQualityTest : public NetEqQualityTest {
                          NetEqDecoder::kDecoderILBC) {}
 
   void SetUp() override {
-    ASSERT_EQ(1, channels_) << "iLBC supports only mono audio.";
+    ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
     AudioEncoderIlbc::Config config;
     config.frame_size_ms = FLAGS_frame_size_ms;
     encoder_.reset(new AudioEncoderIlbc(config));
@@ -66,8 +66,10 @@ class NetEqIlbcQualityTest : public NetEqQualityTest {
     uint32_t dummy_timestamp = 0;
     AudioEncoder::EncodedInfo info;
     do {
-      info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
-                              kFrameSizeSamples, max_bytes, payload);
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              max_bytes, payload);
       encoded_samples += kFrameSizeSamples;
     } while (info.encoded_bytes == 0);
     return rtc::checked_cast<int>(info.encoded_bytes);
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index 66b0903f66..4ccebb3e66 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -59,7 +59,7 @@ NetEqIsacQualityTest::NetEqIsacQualityTest()
       bit_rate_kbps_(FLAGS_bit_rate_kbps) {}
 
 void NetEqIsacQualityTest::SetUp() {
-  ASSERT_EQ(1, channels_) << "iSAC supports only mono audio.";
+  ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio.";
   // Create encoder memory.
   WebRtcIsacfix_Create(&isac_encoder_);
   ASSERT_TRUE(isac_encoder_ != NULL);
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
index 5e8b2297d4..5ab55ba9e8 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
 
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
index 422a9fa6eb..ac478ab5ac 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -11,7 +11,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -50,7 +50,7 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
                          NetEqDecoder::kDecoderPCMu) {}
 
   void SetUp() override {
-    ASSERT_EQ(1, channels_) << "PCMu supports only mono audio.";
+    ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
     AudioEncoderPcmU::Config config;
     config.frame_size_ms = FLAGS_frame_size_ms;
     encoder_.reset(new AudioEncoderPcmU(config));
@@ -66,8 +66,10 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
     uint32_t dummy_timestamp = 0;
     AudioEncoder::EncodedInfo info;
     do {
-      info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
-                              kFrameSizeSamples, max_bytes, payload);
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              max_bytes, payload);
       encoded_samples += kFrameSizeSamples;
     } while (info.encoded_bytes == 0);
     return rtc::checked_cast<int>(info.encoded_bytes);
diff --git a/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
index eb69ac7889..c1abdc30f5 100644
--- a/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
+++ b/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -52,19 +52,11 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
       denominator_ = 1;
       break;
     }
-    case NetEqDecoder::kDecoderCNGswb48kHz: {
-      // Use timestamp scaling with factor 2/3 (32 kHz sample rate, but RTP
-      // timestamps run on 48 kHz).
-      // TODO(tlegrand): Remove scaling for kDecoderCNGswb48kHz once ACM has
-      // full 48 kHz support.
-      numerator_ = 2;
-      denominator_ = 3;
-      break;
-    }
     case NetEqDecoder::kDecoderAVT:
     case NetEqDecoder::kDecoderCNGnb:
     case NetEqDecoder::kDecoderCNGwb:
-    case NetEqDecoder::kDecoderCNGswb32kHz: {
+    case NetEqDecoder::kDecoderCNGswb32kHz:
+    case NetEqDecoder::kDecoderCNGswb48kHz: {
       // Do not change the timestamp scaling settings for DTMF or CNG.
       break;
     }
@@ -87,8 +79,6 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
     assert(denominator_ > 0);  // Should not be possible.
     external_ref_ = external_timestamp;
     internal_ref_ += (external_diff * numerator_) / denominator_;
-    LOG(LS_VERBOSE) << "Converting timestamp: " << external_timestamp <<
-        " -> " << internal_ref_;
     return internal_ref_;
   } else {
     // No scaling.
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc b/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
index 2d2a7e3dd4..eed95753f0 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -43,13 +43,14 @@ bool AudioLoop::Init(const std::string file_name,
   return true;
 }
 
-const int16_t* AudioLoop::GetNextBlock() {
+rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
   // Check that the AudioLoop is initialized.
-  if (block_length_samples_ == 0) return NULL;
+  if (block_length_samples_ == 0)
+    return rtc::ArrayView<const int16_t>();
 
   const int16_t* output_ptr = &audio_array_[next_index_];
   next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
-  return output_ptr;
+  return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
 }
 
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
index a897ee5aef..14e20f68ac 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
@@ -13,6 +13,7 @@
 
 #include <string>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
@@ -40,10 +41,9 @@ class AudioLoop {
   bool Init(const std::string file_name, size_t max_loop_length_samples,
             size_t block_length_samples);
 
-  // Returns a pointer to the next block of audio. The number given as
-  // |block_length_samples| to the Init() function determines how many samples
-  // that can be safely read from the pointer.
-  const int16_t* GetNextBlock();
+  // Returns a (pointer,size) pair for the next block of audio. The size is
+  // equal to the |block_length_samples| Init() argument.
+  rtc::ArrayView<const int16_t> GetNextBlock();
 
  private:
   size_t next_index_;
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
index 3bd2df5ca8..489a8b2ad8 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
index dc07030dd6..5a9f79f877 100644
--- a/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
@@ -13,7 +13,7 @@
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 49750c26c8..694b9ed153 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -12,6 +12,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 
 namespace webrtc {
 namespace test {
@@ -21,26 +22,25 @@ NetEqExternalDecoderTest::NetEqExternalDecoderTest(NetEqDecoder codec,
     : codec_(codec),
       decoder_(decoder),
       sample_rate_hz_(CodecSampleRateHz(codec_)),
-      channels_(static_cast<int>(decoder_->Channels())) {
+      channels_(decoder_->Channels()) {
   NetEq::Config config;
   config.sample_rate_hz = sample_rate_hz_;
   neteq_.reset(NetEq::Create(config));
-  printf("%d\n", channels_);
+  printf("%" PRIuS "\n", channels_);
 }
 
 void NetEqExternalDecoderTest::Init() {
-  ASSERT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
-                            decoder_, codec_, kPayloadType, sample_rate_hz_));
+  ASSERT_EQ(NetEq::kOK,
+            neteq_->RegisterExternalDecoder(decoder_, codec_, name_,
+                                            kPayloadType, sample_rate_hz_));
 }
 
-void NetEqExternalDecoderTest::InsertPacket(WebRtcRTPHeader rtp_header,
-                                            const uint8_t* payload,
-                                            size_t payload_size_bytes,
-                                            uint32_t receive_timestamp) {
-  ASSERT_EQ(
-      NetEq::kOK,
-      neteq_->InsertPacket(
-          rtp_header, payload, payload_size_bytes, receive_timestamp));
+void NetEqExternalDecoderTest::InsertPacket(
+    WebRtcRTPHeader rtp_header,
+    rtc::ArrayView<const uint8_t> payload,
+    uint32_t receive_timestamp) {
+  ASSERT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(rtp_header, payload, receive_timestamp));
 }
 
 size_t NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
@@ -48,7 +48,7 @@ size_t NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
                                                 NetEqOutputType* output_type) {
   // Get audio from regular instance.
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(max_length,
                              output,
@@ -58,6 +58,7 @@ size_t NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
   EXPECT_EQ(channels_, num_channels);
   EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
             samples_per_channel);
+  EXPECT_EQ(sample_rate_hz_, neteq_->last_output_sample_rate_hz());
   return samples_per_channel;
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index 0a41c6ec20..d7b01fe33a 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -11,10 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
 
+#include <string>
+
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace test {
@@ -36,8 +38,8 @@ class NetEqExternalDecoderTest {
   // |payload_size_bytes| bytes. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
-  virtual void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
-                            size_t payload_size_bytes,
+  virtual void InsertPacket(WebRtcRTPHeader rtp_header,
+                            rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp);
 
   // Get 10 ms of audio data. The data is written to |output|, which can hold
@@ -49,9 +51,10 @@ class NetEqExternalDecoderTest {
 
  private:
   NetEqDecoder codec_;
+  std::string name_ = "dummy name";
   AudioDecoder* decoder_;
   int sample_rate_hz_;
-  int channels_;
+  size_t channels_;
   rtc::scoped_ptr<NetEq> neteq_;
 };
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 9fe4dffa91..7d1f9f9798 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
 
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
@@ -34,6 +34,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
   const int kSampRateHz = 32000;
   const webrtc::NetEqDecoder kDecoderType =
       webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz;
+  const std::string kDecoderName = "pcm16-swb32";
   const int kPayloadType = 95;
 
   // Initialize NetEq instance.
@@ -41,7 +42,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
   config.sample_rate_hz = kSampRateHz;
   NetEq* neteq = NetEq::Create(config);
   // Register decoder in |neteq|.
-  if (neteq->RegisterPayloadType(kDecoderType, kPayloadType) != 0)
+  if (neteq->RegisterPayloadType(kDecoderType, kDecoderName, kPayloadType) != 0)
     return -1;
 
   // Set up AudioLoop object.
@@ -62,12 +63,13 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
   bool drift_flipped = false;
   int32_t packet_input_time_ms =
       rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
-  const int16_t* input_samples = audio_loop.GetNextBlock();
-  if (!input_samples) exit(1);
+  auto input_samples = audio_loop.GetNextBlock();
+  if (input_samples.empty())
+    exit(1);
   uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
-  size_t payload_len =
-      WebRtcPcm16b_Encode(input_samples, kInputBlockSizeSamples, input_payload);
-  assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+  size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                           input_samples.size(), input_payload);
+  RTC_CHECK_EQ(sizeof(input_payload), payload_len);
 
   // Main loop.
   webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
@@ -81,9 +83,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
       }
       if (!lost) {
         // Insert packet.
-        int error = neteq->InsertPacket(
-            rtp_header, input_payload, payload_len,
-            packet_input_time_ms * kSampRateHz / 1000);
+        int error =
+            neteq->InsertPacket(rtp_header, input_payload,
+                                packet_input_time_ms * kSampRateHz / 1000);
         if (error != NetEq::kOK)
           return -1;
       }
@@ -93,10 +95,10 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
                                                   kInputBlockSizeSamples,
                                                   &rtp_header);
       input_samples = audio_loop.GetNextBlock();
-      if (!input_samples) return -1;
-      payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
-                                        kInputBlockSizeSamples,
-                                        input_payload);
+      if (input_samples.empty())
+        return -1;
+      payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                        input_samples.size(), input_payload);
       assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
     }
 
@@ -107,7 +109,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
     static const size_t kOutDataLen =
         kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
     int16_t out_data[kOutDataLen];
-    int num_channels;
+    size_t num_channels;
     size_t samples_per_channel;
     int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                 &num_channels, NULL);
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 6826d1be74..9c64e0fb48 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -210,7 +210,7 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
                                    int out_sampling_khz,
                                    NetEqDecoder decoder_type)
     : decoder_type_(decoder_type),
-      channels_(FLAGS_channels),
+      channels_(static_cast<size_t>(FLAGS_channels)),
       decoded_time_ms_(0),
       decodable_time_ms_(0),
       drift_factor_(FLAGS_drift_factor),
@@ -292,7 +292,8 @@ bool GilbertElliotLoss::Lost() {
 }
 
 void NetEqQualityTest::SetUp() {
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType));
+  ASSERT_EQ(0,
+            neteq_->RegisterPayloadType(decoder_type_, "noname", kPayloadType));
   rtp_generator_->set_drift_factor(drift_factor_);
 
   int units = block_duration_ms_ / kPacketLossTimeUnitMs;
@@ -377,9 +378,10 @@ int NetEqQualityTest::Transmit() {
         << " ms ";
   if (payload_size_bytes_ > 0) {
     if (!PacketLost()) {
-      int ret = neteq_->InsertPacket(rtp_header_, &payload_[0],
-                                     payload_size_bytes_,
-                                     packet_input_time_ms * in_sampling_khz_);
+      int ret = neteq_->InsertPacket(
+          rtp_header_,
+          rtc::ArrayView<const uint8_t>(payload_.get(), payload_size_bytes_),
+          packet_input_time_ms * in_sampling_khz_);
       if (ret != NetEq::kOK)
         return -1;
       Log() << "was sent.";
@@ -392,7 +394,7 @@ int NetEqQualityTest::Transmit() {
 }
 
 int NetEqQualityTest::DecodeBlock() {
-  int channels;
+  size_t channels;
   size_t samples;
   int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
                              &samples, &channels, NULL);
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
index e20be5796b..c2b2effee2 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -99,7 +99,7 @@ class NetEqQualityTest : public ::testing::Test {
   std::ofstream& Log();
 
   NetEqDecoder decoder_type_;
-  const int channels_;
+  const size_t channels_;
 
  private:
   int decoded_time_ms_;
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 0aaf8c71fd..3d79e5b5a2 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -26,7 +26,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
@@ -34,7 +34,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -189,8 +189,9 @@ std::string CodecName(webrtc::NetEqDecoder codec) {
 
 void RegisterPayloadType(NetEq* neteq,
                          webrtc::NetEqDecoder codec,
+                         const std::string& name,
                          google::int32 flag) {
-  if (neteq->RegisterPayloadType(codec, static_cast<uint8_t>(flag))) {
+  if (neteq->RegisterPayloadType(codec, name, static_cast<uint8_t>(flag))) {
     std::cerr << "Cannot register payload type " << flag << " as "
               << CodecName(codec) << std::endl;
     exit(1);
@@ -200,30 +201,40 @@ void RegisterPayloadType(NetEq* neteq,
 // Registers all decoders in |neteq|.
 void RegisterPayloadTypes(NetEq* neteq) {
   assert(neteq);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCMu, FLAGS_pcmu);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCMa, FLAGS_pcma);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderILBC, FLAGS_ilbc);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderISAC, FLAGS_isac);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderISACswb,
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCMu, "pcmu",
+                      FLAGS_pcmu);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCMa, "pcma",
+                      FLAGS_pcma);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderILBC, "ilbc",
+                      FLAGS_ilbc);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderISAC, "isac",
+                      FLAGS_isac);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderISACswb, "isac-swb",
                       FLAGS_isac_swb);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderOpus, FLAGS_opus);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16B,
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderOpus, "opus",
+                      FLAGS_opus);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16B, "pcm16-nb",
                       FLAGS_pcm16b);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16Bwb,
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb",
                       FLAGS_pcm16b_wb);
   RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz,
-                      FLAGS_pcm16b_swb32);
+                      "pcm16-swb32", FLAGS_pcm16b_swb32);
   RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz,
-                      FLAGS_pcm16b_swb48);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderG722, FLAGS_g722);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderAVT, FLAGS_avt);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderRED, FLAGS_red);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGnb, FLAGS_cn_nb);
-  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGwb, FLAGS_cn_wb);
+                      "pcm16-swb48", FLAGS_pcm16b_swb48);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderG722, "g722",
+                      FLAGS_g722);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderAVT, "avt",
+                      FLAGS_avt);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderRED, "red",
+                      FLAGS_red);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGnb, "cng-nb",
+                      FLAGS_cn_nb);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGwb, "cng-wb",
+                      FLAGS_cn_wb);
   RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGswb32kHz,
-                      FLAGS_cn_swb32);
+                      "cng-swb32", FLAGS_cn_swb32);
   RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGswb48kHz,
-                      FLAGS_cn_swb48);
+                      "cng-swb48", FLAGS_cn_swb48);
 }
 
 void PrintCodecMappingEntry(webrtc::NetEqDecoder codec, google::int32 flag) {
@@ -399,23 +410,12 @@ int main(int argc, char* argv[]) {
 
   printf("Input file: %s\n", argv[1]);
 
-  // TODO(ivoc): Modify the RtpFileSource::Create and RtcEventLogSource::Create
-  //             functions to return a nullptr on failure instead of crashing
-  //             the program.
-
-  // This temporary solution uses a RtpFileReader directly to check if the file
-  // is a valid RtpDump file.
   bool is_rtp_dump = false;
-  {
-    rtc::scoped_ptr<webrtc::test::RtpFileReader> rtp_reader(
-        webrtc::test::RtpFileReader::Create(
-            webrtc::test::RtpFileReader::kRtpDump, argv[1]));
-    if (rtp_reader)
-      is_rtp_dump = true;
-  }
   rtc::scoped_ptr<webrtc::test::PacketSource> file_source;
   webrtc::test::RtcEventLogSource* event_log_source = nullptr;
-  if (is_rtp_dump) {
+  if (webrtc::test::RtpFileSource::ValidRtpDump(argv[1]) ||
+      webrtc::test::RtpFileSource::ValidPcap(argv[1])) {
+    is_rtp_dump = true;
     file_source.reset(webrtc::test::RtpFileSource::Create(argv[1]));
   } else {
     event_log_source = webrtc::test::RtcEventLogSource::Create(argv[1]);
@@ -558,7 +558,7 @@ int main(int argc, char* argv[]) {
         payload_ptr = payload.get();
       }
       int error = neteq->InsertPacket(
-          rtp_header, payload_ptr, payload_len,
+          rtp_header, rtc::ArrayView<const uint8_t>(payload_ptr, payload_len),
           static_cast<uint32_t>(packet->time_ms() * sample_rate_hz / 1000));
       if (error != NetEq::kOK) {
         if (neteq->LastError() == NetEq::kUnknownRtpPayloadType) {
@@ -609,7 +609,7 @@ int main(int argc, char* argv[]) {
       static const size_t kOutDataLen =
           kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
       int16_t out_data[kOutDataLen];
-      int num_channels;
+      size_t num_channels;
       size_t samples_per_channel;
       int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                    &num_channels, NULL);
diff --git a/webrtc/modules/audio_coding/neteq/tools/packet.cc b/webrtc/modules/audio_coding/neteq/tools/packet.cc
index b8b27afdec..2b2fcc286e 100644
--- a/webrtc/modules/audio_coding/neteq/tools/packet.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -12,8 +12,8 @@
 
 #include <string.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 namespace webrtc {
 namespace test {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index 9b17ba8f64..dad72eaecd 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -18,7 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index 7150bcfe89..90d5931224 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -16,7 +16,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
index 9681ad17ea..b7a3109c01 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -20,7 +20,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 namespace webrtc {
@@ -32,6 +32,18 @@ RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
   return source;
 }
 
+bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
+  rtc::scoped_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  return !!temp_file;
+}
+
+bool RtpFileSource::ValidPcap(const std::string& file_name) {
+  rtc::scoped_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  return !!temp_file;
+}
+
 RtpFileSource::~RtpFileSource() {
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
index d0856a819c..2febf68b91 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -18,7 +18,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
@@ -34,6 +34,10 @@ class RtpFileSource : public PacketSource {
   // opened, or has the wrong format, NULL will be returned.
   static RtpFileSource* Create(const std::string& file_name);
 
+  // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
+  static bool ValidRtpDump(const std::string& file_name);
+  static bool ValidPcap(const std::string& file_name);
+
   virtual ~RtpFileSource();
 
   // Registers an RTP header extension and binds it to |id|.
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
index 6c16192daa..53371be8f6 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/ACMTest.h b/webrtc/modules/audio_coding/test/ACMTest.h
index f73961f5e5..d7e87d34ba 100644
--- a/webrtc/modules/audio_coding/main/test/ACMTest.h
+++ b/webrtc/modules/audio_coding/test/ACMTest.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
 
 class ACMTest {
  public:
@@ -18,4 +18,4 @@ class ACMTest {
   virtual void Perform() = 0;
 };
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
diff --git a/webrtc/modules/audio_coding/main/test/APITest.cc b/webrtc/modules/audio_coding/test/APITest.cc
index 1313f35332..bf04d7c825 100644
--- a/webrtc/modules/audio_coding/main/test/APITest.cc
+++ b/webrtc/modules/audio_coding/test/APITest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
 
 #include <ctype.h>
 #include <stdio.h>
@@ -20,13 +20,13 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -36,12 +36,6 @@ namespace webrtc {
 #define TEST_DURATION_SEC 600
 #define NUMBER_OF_SENDER_TESTS 6
 #define MAX_FILE_NAME_LENGTH_BYTE 500
-#define CHECK_THREAD_NULLITY(myThread, S)                                      \
-  if(myThread != NULL) {                                                       \
-    (myThread)->Start();                                                       \
-  } else {                                                                     \
-    ADD_FAILURE() << S;                                                        \
-  }
 
 void APITest::Wait(uint32_t waitLengthMs) {
   if (_randomTest) {
@@ -522,38 +516,34 @@ void APITest::Perform() {
   //--- THREADS
   // A
   // PUSH
-  rtc::scoped_ptr<ThreadWrapper> myPushAudioThreadA =
-      ThreadWrapper::CreateThread(PushAudioThreadA, this, "PushAudioThreadA");
-  CHECK_THREAD_NULLITY(myPushAudioThreadA, "Unable to start A::PUSH thread");
+  rtc::PlatformThread myPushAudioThreadA(PushAudioThreadA, this,
+                                         "PushAudioThreadA");
+  myPushAudioThreadA.Start();
   // PULL
-  rtc::scoped_ptr<ThreadWrapper> myPullAudioThreadA =
-      ThreadWrapper::CreateThread(PullAudioThreadA, this, "PullAudioThreadA");
-  CHECK_THREAD_NULLITY(myPullAudioThreadA, "Unable to start A::PULL thread");
+  rtc::PlatformThread myPullAudioThreadA(PullAudioThreadA, this,
+                                         "PullAudioThreadA");
+  myPullAudioThreadA.Start();
   // Process
-  rtc::scoped_ptr<ThreadWrapper> myProcessThreadA = ThreadWrapper::CreateThread(
-      ProcessThreadA, this, "ProcessThreadA");
-  CHECK_THREAD_NULLITY(myProcessThreadA, "Unable to start A::Process thread");
+  rtc::PlatformThread myProcessThreadA(ProcessThreadA, this, "ProcessThreadA");
+  myProcessThreadA.Start();
   // API
-  rtc::scoped_ptr<ThreadWrapper> myAPIThreadA = ThreadWrapper::CreateThread(
-      APIThreadA, this, "APIThreadA");
-  CHECK_THREAD_NULLITY(myAPIThreadA, "Unable to start A::API thread");
+  rtc::PlatformThread myAPIThreadA(APIThreadA, this, "APIThreadA");
+  myAPIThreadA.Start();
   // B
   // PUSH
-  rtc::scoped_ptr<ThreadWrapper> myPushAudioThreadB =
-      ThreadWrapper::CreateThread(PushAudioThreadB, this, "PushAudioThreadB");
-  CHECK_THREAD_NULLITY(myPushAudioThreadB, "Unable to start B::PUSH thread");
+  rtc::PlatformThread myPushAudioThreadB(PushAudioThreadB, this,
+                                         "PushAudioThreadB");
+  myPushAudioThreadB.Start();
   // PULL
-  rtc::scoped_ptr<ThreadWrapper> myPullAudioThreadB =
-      ThreadWrapper::CreateThread(PullAudioThreadB, this, "PullAudioThreadB");
-  CHECK_THREAD_NULLITY(myPullAudioThreadB, "Unable to start B::PULL thread");
+  rtc::PlatformThread myPullAudioThreadB(PullAudioThreadB, this,
+                                         "PullAudioThreadB");
+  myPullAudioThreadB.Start();
   // Process
-  rtc::scoped_ptr<ThreadWrapper> myProcessThreadB = ThreadWrapper::CreateThread(
-      ProcessThreadB, this, "ProcessThreadB");
-  CHECK_THREAD_NULLITY(myProcessThreadB, "Unable to start B::Process thread");
+  rtc::PlatformThread myProcessThreadB(ProcessThreadB, this, "ProcessThreadB");
+  myProcessThreadB.Start();
   // API
-  rtc::scoped_ptr<ThreadWrapper> myAPIThreadB = ThreadWrapper::CreateThread(
-      APIThreadB, this, "APIThreadB");
-  CHECK_THREAD_NULLITY(myAPIThreadB, "Unable to start B::API thread");
+  rtc::PlatformThread myAPIThreadB(APIThreadB, this, "APIThreadB");
+  myAPIThreadB.Start();
 
   //_apiEventA->StartTimer(true, 5000);
   //_apiEventB->StartTimer(true, 5000);
@@ -587,15 +577,15 @@ void APITest::Perform() {
   //(unsigned long)((unsigned long)TEST_DURATION_SEC * (unsigned long)1000));
   delete completeEvent;
 
-  myPushAudioThreadA->Stop();
-  myPullAudioThreadA->Stop();
-  myProcessThreadA->Stop();
-  myAPIThreadA->Stop();
+  myPushAudioThreadA.Stop();
+  myPullAudioThreadA.Stop();
+  myProcessThreadA.Stop();
+  myAPIThreadA.Stop();
 
-  myPushAudioThreadB->Stop();
-  myPullAudioThreadB->Stop();
-  myProcessThreadB->Stop();
-  myAPIThreadB->Stop();
+  myPushAudioThreadB.Stop();
+  myPullAudioThreadB.Stop();
+  myProcessThreadB.Stop();
+  myAPIThreadB.Stop();
 }
 
 void APITest::CheckVADStatus(char side) {
@@ -823,9 +813,11 @@ void APITest::TestRegisteration(char sendSide) {
       exit(-1);
   }
 
-  CodecInst myCodec;
-  if (sendACM->SendCodec(&myCodec) < 0) {
-    AudioCodingModule::Codec(_codecCntrA, &myCodec);
+  auto myCodec = sendACM->SendCodec();
+  if (!myCodec) {
+    CodecInst ci;
+    AudioCodingModule::Codec(_codecCntrA, &ci);
+    myCodec = rtc::Optional<CodecInst>(ci);
   }
 
   if (!_randomTest) {
@@ -837,12 +829,12 @@ void APITest::TestRegisteration(char sendSide) {
     *thereIsDecoder = false;
   }
   //myEvent->Wait(20);
-  CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
+  CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
   Wait(1000);
 
-  int currentPayload = myCodec.pltype;
+  int currentPayload = myCodec->pltype;
 
-  if (!FixedPayloadTypeCodec(myCodec.plname)) {
+  if (!FixedPayloadTypeCodec(myCodec->plname)) {
     int32_t i;
     for (i = 0; i < 32; i++) {
       if (!_payloadUsed[i]) {
@@ -850,9 +842,9 @@ void APITest::TestRegisteration(char sendSide) {
           fprintf(stdout,
                   "Register receive codec with new Payload, AUDIO BACK.\n");
         }
-        //myCodec.pltype = i + 96;
-        //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
-        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
+        //myCodec->pltype = i + 96;
+        //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
         //myEvent->Wait(20);
         //{
         //    WriteLockScoped wl(_apiTestRWLock);
@@ -868,17 +860,17 @@ void APITest::TestRegisteration(char sendSide) {
         //    *thereIsDecoder = false;
         //}
         //myEvent->Wait(20);
-        //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
+        //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
         Wait(1000);
 
-        myCodec.pltype = currentPayload;
+        myCodec->pltype = currentPayload;
         if (!_randomTest) {
           fprintf(stdout,
                   "Register receive codec with default Payload, AUDIO BACK.\n");
           fflush (stdout);
         }
-        CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
-        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
+        CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
         myEvent->Wait(20);
         {
           WriteLockScoped wl(_apiTestRWLock);
@@ -890,7 +882,7 @@ void APITest::TestRegisteration(char sendSide) {
       }
     }
     if (i == 32) {
-      CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+      CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
       {
         WriteLockScoped wl(_apiTestRWLock);
         *thereIsDecoder = true;
@@ -902,9 +894,9 @@ void APITest::TestRegisteration(char sendSide) {
               "Register receive codec with fixed Payload, AUDIO BACK.\n");
       fflush (stdout);
     }
-    CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
-    //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
-    //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+    CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+    //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
+    //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
     myEvent->Wait(20);
     {
       WriteLockScoped wl(_apiTestRWLock);
@@ -1001,22 +993,17 @@ void APITest::TestSendVAD(char side) {
 }
 
 void APITest::CurrentCodec(char side) {
-  CodecInst myCodec;
-  if (side == 'A') {
-    _acmA->SendCodec(&myCodec);
-  } else {
-    _acmB->SendCodec(&myCodec);
-  }
+  auto myCodec = (side == 'A' ? _acmA : _acmB)->SendCodec();
 
   if (!_randomTest) {
     fprintf(stdout, "\n\n");
     fprintf(stdout, "Send codec in Side A\n");
     fprintf(stdout, "----------------------------\n");
-    fprintf(stdout, "Name................. %s\n", myCodec.plname);
-    fprintf(stdout, "Sampling Frequency... %d\n", myCodec.plfreq);
-    fprintf(stdout, "Rate................. %d\n", myCodec.rate);
-    fprintf(stdout, "Payload-type......... %d\n", myCodec.pltype);
-    fprintf(stdout, "Packet-size.......... %d\n", myCodec.pacsize);
+    fprintf(stdout, "Name................. %s\n", myCodec->plname);
+    fprintf(stdout, "Sampling Frequency... %d\n", myCodec->plfreq);
+    fprintf(stdout, "Rate................. %d\n", myCodec->rate);
+    fprintf(stdout, "Payload-type......... %d\n", myCodec->pltype);
+    fprintf(stdout, "Packet-size.......... %d\n", myCodec->pacsize);
   }
 
   Wait(100);
diff --git a/webrtc/modules/audio_coding/main/test/APITest.h b/webrtc/modules/audio_coding/test/APITest.h
index d4c5b1ecdd..a1937c2b00 100644
--- a/webrtc/modules/audio_coding/main/test/APITest.h
+++ b/webrtc/modules/audio_coding/test/APITest.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
@@ -160,4 +160,4 @@ class APITest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
diff --git a/webrtc/modules/audio_coding/main/test/Channel.cc b/webrtc/modules/audio_coding/test/Channel.cc
index 02bd783a38..31521fe1e3 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.cc
+++ b/webrtc/modules/audio_coding/test/Channel.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
 
 #include <assert.h>
 #include <iostream>
diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h
index 39d4dabd98..b047aa9909 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/webrtc/modules/audio_coding/test/Channel.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
 
 #include <stdio.h>
 
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -127,4 +127,4 @@ class Channel : public AudioPacketizationCallback {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index d062af0fb9..ba3c8d9ad2 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 #include <sstream>
 #include <stdio.h>
@@ -17,9 +17,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -52,7 +52,7 @@ Sender::Sender()
 }
 
 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                   std::string in_file_name, int sample_rate, int channels) {
+                   std::string in_file_name, int sample_rate, size_t channels) {
   struct CodecInst sendCodec;
   int noOfCodecs = acm->NumberOfCodecs();
   int codecNo;
@@ -63,6 +63,10 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
   if (channels == 2) {
     _pcmFile.ReadStereo(true);
   }
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _pcmFile.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  _pcmFile.FastForward(100);
 
   // Set the codec for the current test.
   if ((testMode == 0) || (testMode == 1)) {
@@ -119,7 +123,7 @@ Receiver::Receiver()
 }
 
 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                     std::string out_file_name, int channels) {
+                     std::string out_file_name, size_t channels) {
   struct CodecInst recvCodec = CodecInst();
   int noOfCodecs;
   EXPECT_EQ(0, acm->InitializeReceiver());
@@ -339,8 +343,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
   _sender.codeId = codeId;
 
   _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
-  struct CodecInst sendCodecInst;
-  if (acm->SendCodec(&sendCodecInst) >= 0) {
+  if (acm->SendCodec()) {
     _sender.Run();
   }
   _sender.Teardown();
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
index 4ad92cec15..f9a9a5bb52 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
 
 #include <stdio.h>
 #include <string.h>
 
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/RTPFile.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/RTPFile.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -48,7 +48,7 @@ class Sender {
  public:
   Sender();
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string in_file_name, int sample_rate, int channels);
+             std::string in_file_name, int sample_rate, size_t channels);
   void Teardown();
   void Run();
   bool Add10MsData();
@@ -71,7 +71,7 @@ class Receiver {
   Receiver();
   virtual ~Receiver() {};
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string out_file_name, int channels);
+             std::string out_file_name, size_t channels);
   void Teardown();
   void Run();
   virtual bool IncomingPacket();
@@ -120,4 +120,4 @@ class EncodeDecodeTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/test/PCMFile.cc
index d0ae7830de..9289d73baa 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/test/PCMFile.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "PCMFile.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 
 #include <ctype.h>
 #include <stdio.h>
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -137,6 +137,9 @@ int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) {
   audio_frame.num_channels_ = channels;
   audio_frame.timestamp_ = timestamp_;
   timestamp_ += samples_10ms_;
+  ++blocks_read_;
+  if (num_10ms_blocks_to_read_ && blocks_read_ >= *num_10ms_blocks_to_read_)
+    end_of_file_ = true;
   return samples_10ms_;
 }
 
@@ -182,11 +185,21 @@ void PCMFile::Write10MsData(int16_t* playout_buffer, size_t length_smpls) {
 void PCMFile::Close() {
   fclose(pcm_file_);
   pcm_file_ = NULL;
+  blocks_read_ = 0;
+}
+
+void PCMFile::FastForward(int num_10ms_blocks) {
+  const int channels = read_stereo_ ? 2 : 1;
+  long num_bytes_to_move =
+      num_10ms_blocks * sizeof(int16_t) * samples_10ms_ * channels;
+  int error = fseek(pcm_file_, num_bytes_to_move, SEEK_CUR);
+  RTC_DCHECK_EQ(error, 0);
 }
 
 void PCMFile::Rewind() {
   rewind(pcm_file_);
   end_of_file_ = false;
+  blocks_read_ = 0;
 }
 
 bool PCMFile::Rewinded() {
@@ -201,4 +214,8 @@ void PCMFile::ReadStereo(bool is_stereo) {
   read_stereo_ = is_stereo;
 }
 
+void PCMFile::SetNum10MsBlocksToRead(int value) {
+  num_10ms_blocks_to_read_ = rtc::Optional<int>(value);
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.h b/webrtc/modules/audio_coding/test/PCMFile.h
index 8353898f03..840933a1bd 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/test/PCMFile.h
@@ -8,15 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
 
 #include <stdio.h>
 #include <stdlib.h>
 
 #include <string>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -45,12 +46,21 @@ class PCMFile {
   bool EndOfFile() const {
     return end_of_file_;
   }
+  // Moves forward the specified number of 10 ms blocks. If a limit has been set
+  // with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
+  // limit.
+  void FastForward(int num_10ms_blocks);
   void Rewind();
   static int16_t ChooseFile(std::string* file_name, int16_t max_len,
                             uint16_t* frequency_hz);
   bool Rewinded();
   void SaveStereo(bool is_stereo = true);
   void ReadStereo(bool is_stereo = true);
+  // If set, the reading will stop after the specified number of blocks have
+  // been read. When that has happened, EndOfFile() will return true. Calling
+  // Rewind() will reset the counter and start over.
+  void SetNum10MsBlocksToRead(int value);
+
  private:
   FILE* pcm_file_;
   uint16_t samples_10ms_;
@@ -61,8 +71,10 @@ class PCMFile {
   uint32_t timestamp_;
   bool read_stereo_;
   bool save_stereo_;
+  rtc::Optional<int> num_10ms_blocks_to_read_;
+  int blocks_read_ = 0;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc b/webrtc/modules/audio_coding/test/PacketLossTest.cc
index f19d491d2d..ad3e83403e 100644
--- a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc
+++ b/webrtc/modules/audio_coding/test/PacketLossTest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common.h"
@@ -143,8 +143,7 @@ void PacketLossTest::Perform() {
 
   sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
                  expected_loss_rate_);
-  struct CodecInst sendCodecInst;
-  if (acm->SendCodec(&sendCodecInst) >= 0) {
+  if (acm->SendCodec()) {
     sender_->Run();
   }
   sender_->Teardown();
diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.h b/webrtc/modules/audio_coding/test/PacketLossTest.h
index d25dea264f..f3570ae1ca 100644
--- a/webrtc/modules/audio_coding/main/test/PacketLossTest.h
+++ b/webrtc/modules/audio_coding/test/PacketLossTest.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 
 #include <string>
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 namespace webrtc {
 
@@ -64,4 +64,4 @@ class PacketLossTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.cc b/webrtc/modules/audio_coding/test/RTPFile.cc
index 60777178c6..60777178c6 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.cc
+++ b/webrtc/modules/audio_coding/test/RTPFile.cc
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/test/RTPFile.h
index c79b63e164..696d41ebd2 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/webrtc/modules/audio_coding/test/RTPFile.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
 
 #include <stdio.h>
 #include <queue>
 
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
@@ -123,4 +123,4 @@ class RTPFile : public RTPStream {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
diff --git a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc b/webrtc/modules/audio_coding/test/SpatialAudio.cc
index 17d4fc88b2..c9f8080826 100644
--- a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
+++ b/webrtc/modules/audio_coding/test/SpatialAudio.cc
@@ -14,7 +14,7 @@
 #include <math.h>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/test/SpatialAudio.h"
+#include "webrtc/modules/audio_coding/test/SpatialAudio.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_coding/main/test/SpatialAudio.h b/webrtc/modules/audio_coding/test/SpatialAudio.h
index fc258977f3..3548cc98eb 100644
--- a/webrtc/modules/audio_coding/main/test/SpatialAudio.h
+++ b/webrtc/modules/audio_coding/test/SpatialAudio.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 
 #define MAX_FILE_NAME_LENGTH_BYTE 500
 
@@ -44,4 +44,4 @@ class SpatialAudio : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
index 19189b6b8f..bacfd37188 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
 
 #include <cstdio>
 #include <limits>
@@ -18,9 +18,9 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
@@ -422,8 +422,12 @@ void TestAllCodecs::Run(TestPack* channel) {
   uint32_t timestamp_diff;
   channel->reset_payload_size();
   int error_count = 0;
-
   int counter = 0;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  infile_a_.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  infile_a_.FastForward(100);
+
   while (!infile_a_.EndOfFile()) {
     // Add 10 msec to ACM.
     infile_a_.Read10MsData(audio_frame);
@@ -477,8 +481,7 @@ void TestAllCodecs::OpenOutFile(int test_number) {
 
 void TestAllCodecs::DisplaySendReceiveCodec() {
   CodecInst my_codec_param;
-  acm_a_->SendCodec(&my_codec_param);
-  printf("%s -> ", my_codec_param.plname);
+  printf("%s -> ", acm_a_->SendCodec()->plname);
   acm_b_->ReceiveCodec(&my_codec_param);
   printf("%s\n", my_codec_param.plname);
 }
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/test/TestAllCodecs.h
index 1cdc0cba98..e79bd69faa 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -81,4 +81,4 @@ class TestAllCodecs : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
diff --git a/webrtc/modules/audio_coding/main/test/TestRedFec.cc b/webrtc/modules/audio_coding/test/TestRedFec.cc
index 0627ae2d74..a1bdc04e53 100644
--- a/webrtc/modules/audio_coding/main/test/TestRedFec.cc
+++ b/webrtc/modules/audio_coding/test/TestRedFec.cc
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
 
 #include <assert.h>
 
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -453,6 +453,10 @@ int16_t TestRedFec::RegisterSendCodec(char side, const char* codecName,
 void TestRedFec::Run() {
   AudioFrame audioFrame;
   int32_t outFreqHzB = _outFileB.SamplingFrequency();
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _inFileA.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  _inFileA.FastForward(100);
 
   while (!_inFileA.EndOfFile()) {
     EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
diff --git a/webrtc/modules/audio_coding/main/test/TestRedFec.h b/webrtc/modules/audio_coding/test/TestRedFec.h
index ac0b6cdfc7..6343d8e374 100644
--- a/webrtc/modules/audio_coding/main/test/TestRedFec.h
+++ b/webrtc/modules/audio_coding/test/TestRedFec.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
 
 #include <string>
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 
 namespace webrtc {
 
@@ -48,4 +48,4 @@ class TestRedFec : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/test/TestStereo.cc
index 69cc3272bb..9bf560d323 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/test/TestStereo.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
 
 #include <assert.h>
 
@@ -17,8 +17,8 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -735,6 +735,12 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
   int error_count = 0;
   int variable_bytes = 0;
   int variable_packets = 0;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  in_file_mono_->SetNum10MsBlocksToRead(50);
+  in_file_stereo_->SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_->FastForward(100);
+  in_file_mono_->FastForward(100);
 
   while (1) {
     // Simulate packet loss by setting |packet_loss_| to "true" in
@@ -800,7 +806,7 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
   // such as Opus.
   if (variable_packets > 0) {
     variable_bytes /= variable_packets;
-    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 3);
+    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 18);
   }
 
   if (in_file_mono_->EndOfFile()) {
@@ -823,14 +829,15 @@ void TestStereo::OpenOutFile(int16_t test_number) {
 }
 
 void TestStereo::DisplaySendReceiveCodec() {
-  CodecInst my_codec_param;
-  acm_a_->SendCodec(&my_codec_param);
+  auto send_codec = acm_a_->SendCodec();
   if (test_mode_ != 0) {
-    printf("%s -> ", my_codec_param.plname);
+    ASSERT_TRUE(send_codec);
+    printf("%s -> ", send_codec->plname);
   }
-  acm_b_->ReceiveCodec(&my_codec_param);
+  CodecInst receive_codec;
+  acm_b_->ReceiveCodec(&receive_codec);
   if (test_mode_ != 0) {
-    printf("%s\n", my_codec_param.plname);
+    printf("%s\n", receive_codec.plname);
   }
 }
 
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/test/TestStereo.h
index b56e995272..4526be6960 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.h
+++ b/webrtc/modules/audio_coding/test/TestStereo.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
 
 #include <math.h>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 
 #define PCMA_AND_PCMU
 
@@ -114,4 +114,4 @@ class TestStereo : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/test/TestVADDTX.cc
index bd0335a5f3..229dc2d474 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/test/TestVADDTX.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
 
 #include <string>
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -87,6 +87,11 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
   PCMFile in_file;
   in_file.Open(in_filename, frequency, "rb");
   in_file.ReadStereo(channels > 1);
+  // Set test length to 1000 ms (100 blocks of 10 ms each).
+  in_file.SetNum10MsBlocksToRead(100);
+  // Fast-forward both files 500 ms (50 blocks). The first second of the file is
+  // silence, but we want to keep half of that to test silence periods.
+  in_file.FastForward(50);
 
   PCMFile out_file;
   if (append) {
@@ -209,9 +214,9 @@ void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
   EXPECT_EQ(0, acm_send_->SetVAD(enable_dtx, enable_vad, vad_mode));
   EXPECT_EQ(0, acm_send_->VAD(&dtx_enabled_, &vad_enabled_, &mode));
 
-  CodecInst codec_param;
-  acm_send_->SendCodec(&codec_param);
-  if (STR_CASE_CMP(codec_param.plname, "opus") == 0) {
+  auto codec_param = acm_send_->SendCodec();
+  ASSERT_TRUE(codec_param);
+  if (STR_CASE_CMP(codec_param->plname, "opus") == 0) {
     // If send codec is Opus, WebRTC VAD/DTX cannot be used.
     enable_dtx = enable_vad = false;
   }
@@ -229,10 +234,10 @@ void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
 // Following is the implementation of TestOpusDtx.
 void TestOpusDtx::Perform() {
 #ifdef WEBRTC_CODEC_ISAC
-  // If we set other codec than Opus, DTX cannot be toggled.
+  // If we set other codec than Opus, DTX cannot be switched on.
   RegisterCodec(kIsacWb);
   EXPECT_EQ(-1, acm_send_->EnableOpusDtx());
-  EXPECT_EQ(-1, acm_send_->DisableOpusDtx());
+  EXPECT_EQ(0, acm_send_->DisableOpusDtx());
 #endif
 
 #ifdef WEBRTC_CODEC_OPUS
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.h b/webrtc/modules/audio_coding/test/TestVADDTX.h
index 07596e2e86..1e7f0ef4d7 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.h
+++ b/webrtc/modules/audio_coding/test/TestVADDTX.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
 
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
 
 namespace webrtc {
 
@@ -99,4 +99,4 @@ class TestOpusDtx final : public TestVadDtx {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
diff --git a/webrtc/modules/audio_coding/main/test/Tester.cc b/webrtc/modules/audio_coding/test/Tester.cc
index 7302e5dcbe..a27f0bc58b 100644
--- a/webrtc/modules/audio_coding/main/test/Tester.cc
+++ b/webrtc/modules/audio_coding/test/Tester.cc
@@ -13,20 +13,19 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/APITest.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
-#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
-#include "webrtc/modules/audio_coding/main/test/opus_test.h"
-#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
-#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
-#include "webrtc/modules/audio_coding/main/test/TestRedFec.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
-#include "webrtc/modules/audio_coding/main/test/TwoWayCommunication.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using webrtc::Trace;
 
@@ -42,7 +41,11 @@ TEST(AudioCodingModuleTest, TestAllCodecs) {
   Trace::ReturnTrace();
 }
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
+#else
+TEST(AudioCodingModuleTest, TestEncodeDecode) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_encodedecode_trace.txt").c_str());
@@ -50,51 +53,54 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
   Trace::ReturnTrace();
 }
 
-#ifdef WEBRTC_CODEC_RED
-#define IF_RED(x) x
+#if defined(WEBRTC_CODEC_RED)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestRedFec) {
 #else
-#define IF_RED(x) DISABLED_##x
+TEST(AudioCodingModuleTest, TestRedFec) {
 #endif
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_RED(TestRedFec))) {
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_fec_trace.txt").c_str());
   webrtc::TestRedFec().Perform();
   Trace::ReturnTrace();
 }
+#endif
 
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
 #else
-#define IF_ISAC(x) DISABLED_##x
+TEST(AudioCodingModuleTest, TestIsac) {
 #endif
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_ISAC(TestIsac))) {
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_isac_trace.txt").c_str());
   webrtc::ISACTest(ACM_TEST_MODE).Perform();
   Trace::ReturnTrace();
 }
+#endif
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
-#define IF_ALL_CODECS(x) x
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
 #else
-#define IF_ALL_CODECS(x) DISABLED_##x
+TEST(AudioCodingModuleTest, TwoWayCommunication) {
 #endif
-
-TEST(AudioCodingModuleTest,
-     DISABLED_ON_ANDROID(IF_ALL_CODECS(TwoWayCommunication))) {
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_twowaycom_trace.txt").c_str());
   webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
   Trace::ReturnTrace();
 }
+#endif
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
+#else
+TEST(AudioCodingModuleTest, TestStereo) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_stereo_trace.txt").c_str());
@@ -102,7 +108,11 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
   Trace::ReturnTrace();
 }
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestWebRtcVadDtx)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestWebRtcVadDtx) {
+#else
+TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_vaddtx_trace.txt").c_str());
diff --git a/webrtc/modules/audio_coding/main/test/TimedTrace.cc b/webrtc/modules/audio_coding/test/TimedTrace.cc
index ff9b5eeb76..ff9b5eeb76 100644
--- a/webrtc/modules/audio_coding/main/test/TimedTrace.cc
+++ b/webrtc/modules/audio_coding/test/TimedTrace.cc
diff --git a/webrtc/modules/audio_coding/main/test/TimedTrace.h b/webrtc/modules/audio_coding/test/TimedTrace.h
index ef9609a267..0793eb0c0c 100644
--- a/webrtc/modules/audio_coding/main/test/TimedTrace.h
+++ b/webrtc/modules/audio_coding/test/TimedTrace.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TIMED_TRACE_H
-#define TIMED_TRACE_H
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -33,4 +33,4 @@ class TimedTrace {
 
 };
 
-#endif
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
index 2ff2a85afe..56e136bd34 100644
--- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
@@ -21,8 +21,8 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -250,10 +250,8 @@ void TwoWayCommunication::Perform() {
 
   AudioFrame audioFrame;
 
-  CodecInst codecInst_B;
-  CodecInst dummy;
-
-  EXPECT_EQ(0, _acmB->SendCodec(&codecInst_B));
+  auto codecInst_B = _acmB->SendCodec();
+  ASSERT_TRUE(codecInst_B);
 
   // In the following loop we tests that the code can handle misuse of the APIs.
   // In the middle of a session with data flowing between two sides, called A
@@ -285,15 +283,15 @@ void TwoWayCommunication::Perform() {
     }
     // Re-register send codec on side B.
     if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
-      EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
-      EXPECT_EQ(0, _acmB->SendCodec(&dummy));
+      EXPECT_EQ(0, _acmB->RegisterSendCodec(*codecInst_B));
+      EXPECT_TRUE(_acmB->SendCodec());
     }
     // Initialize receiver on side A.
     if (((secPassed % 7) == 6) && (msecPassed == 0))
       EXPECT_EQ(0, _acmA->InitializeReceiver());
     // Re-register codec on side A.
     if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
-      EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
+      EXPECT_EQ(0, _acmA->RegisterReceiveCodec(*codecInst_B));
     }
   }
 }
diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h b/webrtc/modules/audio_coding/test/TwoWayCommunication.h
index bf969fe683..77639935da 100644
--- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h
+++ b/webrtc/modules/audio_coding/test/TwoWayCommunication.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 
 namespace webrtc {
 
@@ -57,4 +57,4 @@ class TwoWayCommunication : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc
index 6186d67fc9..a8c137f501 100644
--- a/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/test/delay_test.cc
@@ -19,12 +19,12 @@
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -33,7 +33,6 @@ DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
 DEFINE_int32(num_channels, 1, "Number of Channels.");
 DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
 DEFINE_int32(delay, 0, "Delay in millisecond.");
-DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
@@ -89,10 +88,6 @@ class DelayTest {
         "Couldn't initialize receiver.\n";
     ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
         "Couldn't initialize receiver.\n";
-    if (FLAGS_init_delay > 0) {
-      ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
-          "Failed to set initial delay.\n";
-    }
 
     if (FLAGS_delay > 0) {
       ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
@@ -172,7 +167,7 @@ class DelayTest {
   void OpenOutFile(const char* output_id) {
     std::stringstream file_stream;
     file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
-        << "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
+        << "Hz" << "_" << FLAGS_delay << "ms.pcm";
     std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
     std::string file_name = webrtc::test::OutputPath() + file_stream.str();
     out_file_b_.Open(file_name.c_str(), 32000, "wb");
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/test/iSACTest.cc
index 35c34d5947..9f223fb81f 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/test/iSACTest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
 
 #include <ctype.h>
 #include <stdio.h>
@@ -23,8 +23,8 @@
 #include <time.h>
 #endif
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -47,21 +47,21 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
 
   if ((isacConfig.currentRateBitPerSec != 0)
       || (isacConfig.currentFrameSizeMsec != 0)) {
-    CodecInst sendCodec;
-    EXPECT_EQ(0, acm->SendCodec(&sendCodec));
+    auto sendCodec = acm->SendCodec();
+    EXPECT_TRUE(sendCodec);
     if (isacConfig.currentRateBitPerSec < 0) {
       // Register iSAC in adaptive (channel-dependent) mode.
-      sendCodec.rate = -1;
-      EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
+      sendCodec->rate = -1;
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
     } else {
       if (isacConfig.currentRateBitPerSec != 0) {
-        sendCodec.rate = isacConfig.currentRateBitPerSec;
+        sendCodec->rate = isacConfig.currentRateBitPerSec;
       }
       if (isacConfig.currentFrameSizeMsec != 0) {
-        sendCodec.pacsize = isacConfig.currentFrameSizeMsec
-            * (sendCodec.plfreq / 1000);
+        sendCodec->pacsize = isacConfig.currentFrameSizeMsec
+            * (sendCodec->plfreq / 1000);
       }
-      EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
     }
   }
 
@@ -117,6 +117,10 @@ void ISACTest::Setup() {
   EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
 
   _inFileA.Open(file_name_swb_, 32000, "rb");
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _inFileA.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  _inFileA.FastForward(100);
   std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm";
   std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm";
   _outFileA.Open(fileNameA, 32000, "wb");
@@ -238,7 +242,6 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
   _channel_B2A->ResetStats();
 
   char currentTime[500];
-  CodecInst sendCodec;
   EventTimerWrapper* myEvent = EventTimerWrapper::Create();
   EXPECT_TRUE(myEvent->StartTimer(true, 10));
   while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
@@ -248,8 +251,8 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
 
     if ((adaptiveMode) && (_testMode != 0)) {
       myEvent->Wait(5000);
-      EXPECT_EQ(0, _acmA->SendCodec(&sendCodec));
-      EXPECT_EQ(0, _acmB->SendCodec(&sendCodec));
+      EXPECT_TRUE(_acmA->SendCodec());
+      EXPECT_TRUE(_acmB->SendCodec());
     }
   }
 
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.h b/webrtc/modules/audio_coding/test/iSACTest.h
index 0693d935e1..c5bb515437 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.h
+++ b/webrtc/modules/audio_coding/test/iSACTest.h
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
 
 #include <string.h>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 
 #define MAX_FILE_NAME_LENGTH_BYTE 500
 #define NO_OF_CLIENTS             15
@@ -76,4 +76,4 @@ class ISACTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
index ea7266567e..481df55ffd 100644
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -14,10 +14,10 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -42,7 +42,6 @@ DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
 DEFINE_string(delay, "", "Log for delay.");
 
 // Other setups
-DEFINE_int32(init_delay, 0, "Initial delay.");
 DEFINE_bool(verbose, false, "Verbosity.");
 DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
 
@@ -122,9 +121,6 @@ class InsertPacketWithTiming {
         << " Hz." << std::endl;
 
     // Other setups
-    if (FLAGS_init_delay > 0)
-      EXPECT_EQ(0, receive_acm_->SetInitialPlayoutDelay(FLAGS_init_delay));
-
     if (FLAGS_loss_rate > 0)
       loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
     else
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index 00c66cb3aa..104b5e587b 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/test/opus_test.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
 
 #include <assert.h>
 
@@ -17,11 +17,10 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -63,7 +62,7 @@ void OpusTest::Perform() {
   return;
 #else
   uint16_t frequency_hz;
-  int audio_channels;
+  size_t audio_channels;
   int16_t test_cntr = 0;
 
   // Open both mono and stereo test files in 32 kHz.
@@ -206,17 +205,17 @@ void OpusTest::Perform() {
 #endif
 }
 
-void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
-                   int frame_length, int percent_loss) {
+void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
+                   size_t frame_length, int percent_loss) {
   AudioFrame audio_frame;
   int32_t out_freq_hz_b = out_file_.SamplingFrequency();
-  const int kBufferSizeSamples = 480 * 12 * 2;  // Can hold 120 ms stereo audio.
+  const size_t kBufferSizeSamples = 480 * 12 * 2;  // 120 ms stereo audio.
   int16_t audio[kBufferSizeSamples];
   int16_t out_audio[kBufferSizeSamples];
   int16_t audio_type;
-  int written_samples = 0;
-  int read_samples = 0;
-  int decoded_samples = 0;
+  size_t written_samples = 0;
+  size_t read_samples = 0;
+  size_t decoded_samples = 0;
   bool first_packet = true;
   uint32_t start_time_stamp = 0;
 
@@ -236,8 +235,12 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
                                         kOpusComplexity5));
 #endif
 
-  // Make sure the runtime is less than 60 seconds to pass Android test.
-  for (size_t audio_length = 0; audio_length < 10000; audio_length += 10) {
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_.FastForward(100);
+  in_file_mono_.FastForward(100);
+
+  // Limit the runtime to 1000 blocks of 10 ms each.
+  for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
     bool lost_packet = false;
 
     // Get 10 msec of audio.
@@ -265,14 +268,14 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
 
     // Sometimes we need to loop over the audio vector to produce the right
     // number of packets.
-    int loop_encode = (written_samples - read_samples) /
+    size_t loop_encode = (written_samples - read_samples) /
         (channels * frame_length);
 
     if (loop_encode > 0) {
-      const int kMaxBytes = 1000;  // Maximum number of bytes for one packet.
+      const size_t kMaxBytes = 1000;  // Maximum number of bytes for one packet.
       size_t bitstream_len_byte;
       uint8_t bitstream[kMaxBytes];
-      for (int i = 0; i < loop_encode; i++) {
+      for (size_t i = 0; i < loop_encode; i++) {
         int bitstream_len_byte_int = WebRtcOpus_Encode(
             (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
             &audio[read_samples], frame_length, kMaxBytes, bitstream);
@@ -323,7 +326,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
           first_packet = false;
           start_time_stamp = rtp_timestamp_;
         }
-        rtp_timestamp_ += frame_length;
+        rtp_timestamp_ += static_cast<uint32_t>(frame_length);
         read_samples += frame_length * channels;
       }
       if (read_samples == written_samples) {
@@ -341,8 +344,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
         audio_frame.samples_per_channel_ * audio_frame.num_channels_);
 
     // Write stand-alone speech to file.
-    out_file_standalone_.Write10MsData(
-        out_audio, static_cast<size_t>(decoded_samples) * channels);
+    out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
 
     if (audio_frame.timestamp_ > start_time_stamp) {
       // Number of channels should be the same for both stand-alone and
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/test/opus_test.h
index 379bb86d5d..93c9ffb263 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.h
+++ b/webrtc/modules/audio_coding/test/opus_test.h
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
 
 #include <math.h>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
 
 namespace webrtc {
 
@@ -31,7 +31,10 @@ class OpusTest : public ACMTest {
   void Perform();
 
  private:
-  void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
+  void Run(TestPackStereo* channel,
+           size_t channels,
+           int bitrate,
+           size_t frame_length,
            int percent_loss = 0);
 
   void OpenOutFile(int test_number);
@@ -44,7 +47,7 @@ class OpusTest : public ACMTest {
   PCMFile out_file_standalone_;
   int counter_;
   uint8_t payload_type_;
-  int rtp_timestamp_;
+  uint32_t rtp_timestamp_;
   acm2::ACMResampler resampler_;
   WebRtcOpusEncInst* opus_mono_encoder_;
   WebRtcOpusEncInst* opus_stereo_encoder_;
@@ -54,4 +57,4 @@ class OpusTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index 20b10a376e..195e9d8145 100644
--- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -11,13 +11,12 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -154,7 +153,7 @@ class TargetDelayTest : public ::testing::Test {
       ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame));
       // Had to use ASSERT_TRUE, ASSERT_EQ generated error.
       ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
-      ASSERT_EQ(1, frame.num_channels_);
+      ASSERT_EQ(1u, frame.num_channels_);
       ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_);
     }
   }
@@ -199,23 +198,50 @@ class TargetDelayTest : public ::testing::Test {
   uint8_t payload_[kPayloadLenBytes];
 };
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OutOfRangeInput DISABLED_OutOfRangeInput
+#else
+#define MAYBE_OutOfRangeInput OutOfRangeInput
+#endif
+TEST_F(TargetDelayTest, MAYBE_OutOfRangeInput) {
   OutOfRangeInput();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NoTargetDelayBufferSizeChanges \
+  DISABLED_NoTargetDelayBufferSizeChanges
+#else
+#define MAYBE_NoTargetDelayBufferSizeChanges NoTargetDelayBufferSizeChanges
+#endif
+TEST_F(TargetDelayTest, MAYBE_NoTargetDelayBufferSizeChanges) {
   NoTargetDelayBufferSizeChanges();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_WithTargetDelayBufferNotChanging \
+  DISABLED_WithTargetDelayBufferNotChanging
+#else
+#define MAYBE_WithTargetDelayBufferNotChanging WithTargetDelayBufferNotChanging
+#endif
+TEST_F(TargetDelayTest, MAYBE_WithTargetDelayBufferNotChanging) {
   WithTargetDelayBufferNotChanging();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_RequiredDelayAtCorrectRange DISABLED_RequiredDelayAtCorrectRange
+#else
+#define MAYBE_RequiredDelayAtCorrectRange RequiredDelayAtCorrectRange
+#endif
+TEST_F(TargetDelayTest, MAYBE_RequiredDelayAtCorrectRange) {
   RequiredDelayAtCorrectRange();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
+#else
+#define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
+#endif
+TEST_F(TargetDelayTest, MAYBE_TargetDelayBufferMinMax) {
   TargetDelayBufferMinMax();
 }
 
diff --git a/webrtc/modules/audio_coding/main/test/utility.cc b/webrtc/modules/audio_coding/test/utility.cc
index 34af5e703f..89368bce51 100644
--- a/webrtc/modules/audio_coding/main/test/utility.cc
+++ b/webrtc/modules/audio_coding/test/utility.cc
@@ -18,8 +18,8 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 
 #define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
 
diff --git a/webrtc/modules/audio_coding/main/test/utility.h b/webrtc/modules/audio_coding/test/utility.h
index e936ec1cdd..23869be7ed 100644
--- a/webrtc/modules/audio_coding/main/test/utility.h
+++ b/webrtc/modules/audio_coding/test/utility.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 
 namespace webrtc {
 
@@ -136,4 +136,4 @@ void UseNewAcm(webrtc::Config* config);
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn
index 3b9e2769ac..36391c7abc 100644
--- a/webrtc/modules/audio_conference_mixer/BUILD.gn
+++ b/webrtc/modules/audio_conference_mixer/BUILD.gn
@@ -9,15 +9,15 @@
 config("audio_conference_mixer_config") {
   visibility = [ ":*" ]  # Only targets in this file can depend on this.
   include_dirs = [
-    "interface",
-    "../interface",
+    "include",
+    "../include",
   ]
 }
 
 source_set("audio_conference_mixer") {
   sources = [
-    "interface/audio_conference_mixer.h",
-    "interface/audio_conference_mixer_defines.h",
+    "include/audio_conference_mixer.h",
+    "include/audio_conference_mixer_defines.h",
     "source/audio_conference_mixer_impl.cc",
     "source/audio_conference_mixer_impl.h",
     "source/audio_frame_manipulator.cc",
diff --git a/webrtc/modules/audio_conference_mixer/OWNERS b/webrtc/modules/audio_conference_mixer/OWNERS
index 34bc7389ba..ea2062a9a0 100644
--- a/webrtc/modules/audio_conference_mixer/OWNERS
+++ b/webrtc/modules/audio_conference_mixer/OWNERS
@@ -1,3 +1,8 @@
-andrew@webrtc.org
+minyue@webrtc.org
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
 
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
index 5aa3cc449b..9d7179504c 100644
--- a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
+++ b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
@@ -17,8 +17,8 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'interface/audio_conference_mixer.h',
-        'interface/audio_conference_mixer_defines.h',
+        'include/audio_conference_mixer.h',
+        'include/audio_conference_mixer_defines.h',
         'source/audio_frame_manipulator.cc',
         'source/audio_frame_manipulator.h',
         'source/memory_pool.h',
diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
index 7ff39579ee..7370442704 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
 
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioMixerOutputReceiver;
@@ -74,4 +74,4 @@ protected:
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
index d15b7fca02..5d58f42435 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -57,4 +57,4 @@ protected:
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index 2d2cf9dbb8..afb060f46d 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
@@ -50,8 +50,8 @@ void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
 }
 
 // Return the max number of channels from a |list| composed of AudioFrames.
-int MaxNumChannels(const AudioFrameList* list) {
-  int max_num_channels = 1;
+size_t MaxNumChannels(const AudioFrameList* list) {
+  size_t max_num_channels = 1;
   for (AudioFrameList::const_iterator iter = list->begin();
        iter != list->end();
        ++iter) {
@@ -278,7 +278,7 @@ int32_t AudioConferenceMixerImpl::Process() {
         //                with an API instead of dynamically.
 
         // Find the max channels over all mixing lists.
-        const int num_mixed_channels = std::max(MaxNumChannels(&mixList),
+        const size_t num_mixed_channels = std::max(MaxNumChannels(&mixList),
             std::max(MaxNumChannels(&additionalFramesList),
                      MaxNumChannels(&rampOutList)));
 
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
index bc9a27e9f0..2466112769 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
@@ -16,10 +16,10 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioProcessing;
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
index 636698e9c1..9c5d3b939d 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace {
diff --git a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
index d4fbd205f1..293bfa0db9 100644
--- a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
+++ b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
@@ -10,8 +10,8 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index 8875178c15..5897176845 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -10,7 +10,7 @@ import("../../build/webrtc.gni")
 
 config("audio_device_config") {
   include_dirs = [
-    "../interface",
+    "../include",
     "include",
     "dummy",  # Contains dummy audio device implementations.
   ]
@@ -50,54 +50,50 @@ source_set("audio_device") {
     include_dirs += [ "android" ]
   }
   defines = []
+  cflags = []
   if (rtc_include_internal_audio_device) {
     defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
     sources += [
-      "android/audio_device_template.h",
-      "android/audio_manager.cc",
-      "android/audio_manager.h",
-      "android/audio_record_jni.cc",
-      "android/audio_record_jni.h",
-      "android/audio_track_jni.cc",
-      "android/audio_track_jni.h",
-      "android/opensles_common.cc",
-      "android/opensles_common.h",
-      "android/opensles_player.cc",
-      "android/opensles_player.h",
       "audio_device_impl.cc",
       "audio_device_impl.h",
-      "ios/audio_device_ios.h",
-      "ios/audio_device_ios.mm",
-      "linux/alsasymboltable_linux.cc",
-      "linux/alsasymboltable_linux.h",
-      "linux/audio_device_alsa_linux.cc",
-      "linux/audio_device_alsa_linux.h",
-      "linux/audio_mixer_manager_alsa_linux.cc",
-      "linux/audio_mixer_manager_alsa_linux.h",
-      "linux/latebindingsymboltable_linux.cc",
-      "linux/latebindingsymboltable_linux.h",
-      "mac/audio_device_mac.cc",
-      "mac/audio_device_mac.h",
-      "mac/audio_mixer_manager_mac.cc",
-      "mac/audio_mixer_manager_mac.h",
-      "mac/portaudio/pa_memorybarrier.h",
-      "mac/portaudio/pa_ringbuffer.c",
-      "mac/portaudio/pa_ringbuffer.h",
-      "win/audio_device_core_win.cc",
-      "win/audio_device_core_win.h",
-      "win/audio_device_wave_win.cc",
-      "win/audio_device_wave_win.h",
-      "win/audio_mixer_manager_win.cc",
-      "win/audio_mixer_manager_win.h",
     ]
+    if (is_android) {
+      sources += [
+        "android/audio_device_template.h",
+        "android/audio_manager.cc",
+        "android/audio_manager.h",
+        "android/audio_record_jni.cc",
+        "android/audio_record_jni.h",
+        "android/audio_track_jni.cc",
+        "android/audio_track_jni.h",
+        "android/build_info.cc",
+        "android/build_info.h",
+        "android/opensles_common.cc",
+        "android/opensles_common.h",
+        "android/opensles_player.cc",
+        "android/opensles_player.h",
+      ]
+      libs = [
+        "log",
+        "OpenSLES",
+      ]
+    }
     if (is_linux) {
+      sources += [
+        "linux/alsasymboltable_linux.cc",
+        "linux/alsasymboltable_linux.h",
+        "linux/audio_device_alsa_linux.cc",
+        "linux/audio_device_alsa_linux.h",
+        "linux/audio_mixer_manager_alsa_linux.cc",
+        "linux/audio_mixer_manager_alsa_linux.h",
+        "linux/latebindingsymboltable_linux.cc",
+        "linux/latebindingsymboltable_linux.h",
+      ]
       defines += [ "LINUX_ALSA" ]
-
       libs = [
         "dl",
         "X11",
       ]
-
       if (rtc_include_pulse_audio) {
         sources += [
           "linux/audio_device_pulse_linux.cc",
@@ -107,26 +103,47 @@ source_set("audio_device") {
           "linux/pulseaudiosymboltable_linux.cc",
           "linux/pulseaudiosymboltable_linux.h",
         ]
-
         defines += [ "LINUX_PULSE" ]
       }
     }
     if (is_mac) {
+      sources += [
+        "mac/audio_device_mac.cc",
+        "mac/audio_device_mac.h",
+        "mac/audio_mixer_manager_mac.cc",
+        "mac/audio_mixer_manager_mac.h",
+        "mac/portaudio/pa_memorybarrier.h",
+        "mac/portaudio/pa_ringbuffer.c",
+        "mac/portaudio/pa_ringbuffer.h",
+      ]
       libs = [
         "AudioToolbox.framework",
         "CoreAudio.framework",
       ]
     }
     if (is_ios) {
+      sources += [
+        "ios/audio_device_ios.h",
+        "ios/audio_device_ios.mm",
+        "ios/audio_device_not_implemented_ios.mm",
+      ]
       cflags += [ "-fobjc-arc" ]  # CLANG_ENABLE_OBJC_ARC = YES.
-
       libs = [
         "AudioToolbox.framework",
         "AVFoundation.framework",
         "Foundation.framework",
+        "UIKit.framework",
       ]
     }
     if (is_win) {
+      sources += [
+        "win/audio_device_core_win.cc",
+        "win/audio_device_core_win.h",
+        "win/audio_device_wave_win.cc",
+        "win/audio_device_wave_win.h",
+        "win/audio_mixer_manager_win.cc",
+        "win/audio_mixer_manager_win.h",
+      ]
       libs = [
         # Required for the built-in WASAPI AEC.
         "dmoguids.lib",
diff --git a/webrtc/modules/audio_device/OWNERS b/webrtc/modules/audio_device/OWNERS
index bb11a4ec0e..12d67c035b 100644
--- a/webrtc/modules/audio_device/OWNERS
+++ b/webrtc/modules/audio_device/OWNERS
@@ -4,8 +4,6 @@ niklas.enbom@webrtc.org
 tkchin@webrtc.org
 xians@webrtc.org
 
-per-file *.isolate=kjellander@webrtc.org
-
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
 per-file *.gyp=*
diff --git a/webrtc/modules/audio_device/android/audio_device_unittest.cc b/webrtc/modules/audio_device/android/audio_device_unittest.cc
index 7b2d6354c4..768047df51 100644
--- a/webrtc/modules/audio_device/android/audio_device_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -383,7 +383,7 @@ class MockAudioTransport : public AudioTransport {
                 int32_t(const void* audioSamples,
                         const size_t nSamples,
                         const size_t nBytesPerSample,
-                        const uint8_t nChannels,
+                        const size_t nChannels,
                         const uint32_t samplesPerSec,
                         const uint32_t totalDelayMS,
                         const int32_t clockDrift,
@@ -393,7 +393,7 @@ class MockAudioTransport : public AudioTransport {
   MOCK_METHOD8(NeedMorePlayData,
                int32_t(const size_t nSamples,
                        const size_t nBytesPerSample,
-                       const uint8_t nChannels,
+                       const size_t nChannels,
                        const uint32_t samplesPerSec,
                        void* audioSamples,
                        size_t& nSamplesOut,
@@ -423,7 +423,7 @@ class MockAudioTransport : public AudioTransport {
   int32_t RealRecordedDataIsAvailable(const void* audioSamples,
                                       const size_t nSamples,
                                       const size_t nBytesPerSample,
-                                      const uint8_t nChannels,
+                                      const size_t nChannels,
                                       const uint32_t samplesPerSec,
                                       const uint32_t totalDelayMS,
                                       const int32_t clockDrift,
@@ -445,7 +445,7 @@ class MockAudioTransport : public AudioTransport {
 
   int32_t RealNeedMorePlayData(const size_t nSamples,
                                const size_t nBytesPerSample,
-                               const uint8_t nChannels,
+                               const size_t nChannels,
                                const uint32_t samplesPerSec,
                                void* audioSamples,
                                size_t& nSamplesOut,
@@ -521,10 +521,10 @@ class AudioDeviceTest : public ::testing::Test {
   int record_sample_rate() const {
     return record_parameters_.sample_rate();
   }
-  int playout_channels() const {
+  size_t playout_channels() const {
     return playout_parameters_.channels();
   }
-  int record_channels() const {
+  size_t record_channels() const {
     return record_parameters_.channels();
   }
   size_t playout_frames_per_10ms_buffer() const {
@@ -931,7 +931,7 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
 // not contain any explicit verification that the audio quality is perfect.
 TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
   // TODO(henrika): extend test when mono output is supported.
-  EXPECT_EQ(1, playout_channels());
+  EXPECT_EQ(1u, playout_channels());
   NiceMock<MockAudioTransport> mock(kPlayout);
   const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
   std::string file_name = GetFileName(playout_sample_rate());
diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc
index 169a1929ce..1d08a6adc0 100644
--- a/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/webrtc/modules/audio_device/android/audio_manager.cc
@@ -10,13 +10,15 @@
 
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 
+#include <utility>
+
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 #define TAG "AudioManager"
 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
@@ -29,15 +31,16 @@ namespace webrtc {
 
 // AudioManager::JavaAudioManager implementation
 AudioManager::JavaAudioManager::JavaAudioManager(
-    NativeRegistration* native_reg, rtc::scoped_ptr<GlobalRef> audio_manager)
-    : audio_manager_(audio_manager.Pass()),
+    NativeRegistration* native_reg,
+    rtc::scoped_ptr<GlobalRef> audio_manager)
+    : audio_manager_(std::move(audio_manager)),
       init_(native_reg->GetMethodId("init", "()Z")),
       dispose_(native_reg->GetMethodId("dispose", "()V")),
       is_communication_mode_enabled_(
           native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")),
       is_device_blacklisted_for_open_sles_usage_(
-          native_reg->GetMethodId(
-              "isDeviceBlacklistedForOpenSLESUsage", "()Z")) {
+          native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
+                                  "()Z")) {
   ALOGD("JavaAudioManager::ctor%s", GetThreadInfo().c_str());
 }
 
@@ -71,13 +74,12 @@ AudioManager::AudioManager()
       hardware_agc_(false),
       hardware_ns_(false),
       low_latency_playout_(false),
-      delay_estimate_in_milliseconds_(0),
-      output_stream_type_(0) {
+      delay_estimate_in_milliseconds_(0) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
   RTC_CHECK(j_environment_);
   JNINativeMethod native_methods[] = {
       {"nativeCacheAudioParameters",
-       "(IIZZZZIIIJ)V",
+       "(IIZZZZIIJ)V",
        reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
   j_native_registration_ = j_environment_->RegisterNatives(
       "org/webrtc/voiceengine/WebRtcAudioManager",
@@ -180,14 +182,12 @@ void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
                                                 jboolean low_latency_output,
                                                 jint output_buffer_size,
                                                 jint input_buffer_size,
-                                                jint output_stream_type,
                                                 jlong native_audio_manager) {
   webrtc::AudioManager* this_object =
       reinterpret_cast<webrtc::AudioManager*>(native_audio_manager);
   this_object->OnCacheAudioParameters(
       env, sample_rate, channels, hardware_aec, hardware_agc, hardware_ns,
-      low_latency_output, output_buffer_size, input_buffer_size,
-      output_stream_type);
+      low_latency_output, output_buffer_size, input_buffer_size);
 }
 
 void AudioManager::OnCacheAudioParameters(JNIEnv* env,
@@ -198,8 +198,7 @@ void AudioManager::OnCacheAudioParameters(JNIEnv* env,
                                           jboolean hardware_ns,
                                           jboolean low_latency_output,
                                           jint output_buffer_size,
-                                          jint input_buffer_size,
-                                          jint output_stream_type) {
+                                          jint input_buffer_size) {
   ALOGD("OnCacheAudioParameters%s", GetThreadInfo().c_str());
   ALOGD("hardware_aec: %d", hardware_aec);
   ALOGD("hardware_agc: %d", hardware_agc);
@@ -209,17 +208,15 @@ void AudioManager::OnCacheAudioParameters(JNIEnv* env,
   ALOGD("channels: %d", channels);
   ALOGD("output_buffer_size: %d", output_buffer_size);
   ALOGD("input_buffer_size: %d", input_buffer_size);
-  ALOGD("output_stream_type: %d", output_stream_type);
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   hardware_aec_ = hardware_aec;
   hardware_agc_ = hardware_agc;
   hardware_ns_ = hardware_ns;
   low_latency_playout_ = low_latency_output;
-  output_stream_type_ = output_stream_type;
   // TODO(henrika): add support for stereo output.
-  playout_parameters_.reset(sample_rate, channels,
+  playout_parameters_.reset(sample_rate, static_cast<size_t>(channels),
                             static_cast<size_t>(output_buffer_size));
-  record_parameters_.reset(sample_rate, channels,
+  record_parameters_.reset(sample_rate, static_cast<size_t>(channels),
                            static_cast<size_t>(input_buffer_size));
 }
 
diff --git a/webrtc/modules/audio_device/android/audio_manager.h b/webrtc/modules/audio_device/android/audio_manager.h
index 5f23147b8a..26caf61afe 100644
--- a/webrtc/modules/audio_device/android/audio_manager.h
+++ b/webrtc/modules/audio_device/android/audio_manager.h
@@ -19,8 +19,8 @@
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
@@ -93,8 +93,6 @@ class AudioManager {
   // webrtc::kHighLatencyModeDelayEstimateInMilliseconds.
   int GetDelayEstimateInMilliseconds() const;
 
-  int OutputStreamType() const { return output_stream_type_; }
-
  private:
   // Called from Java side so we can cache the native audio parameters.
   // This method will be called by the WebRtcAudioManager constructor, i.e.
@@ -109,7 +107,6 @@ class AudioManager {
                                            jboolean low_latency_output,
                                            jint output_buffer_size,
                                            jint input_buffer_size,
-                                           jint output_stream_type,
                                            jlong native_audio_manager);
   void OnCacheAudioParameters(JNIEnv* env,
                               jint sample_rate,
@@ -119,8 +116,7 @@ class AudioManager {
                               jboolean hardware_ns,
                               jboolean low_latency_output,
                               jint output_buffer_size,
-                              jint input_buffer_size,
-                              jint output_stream_type);
+                              jint input_buffer_size);
 
   // Stores thread ID in the constructor.
   // We can then use ThreadChecker::CalledOnValidThread() to ensure that
@@ -159,13 +155,6 @@ class AudioManager {
   // device supports low-latency output or not.
   int delay_estimate_in_milliseconds_;
 
-  // Contains the output stream type provided to this class at construction by
-  // the AudioManager in Java land. Possible values are:
-  //  - AudioManager.STREAM_VOICE_CALL = 0
-  //  - AudioManager.STREAM_RING = 2
-  //  - AudioManager.STREAM_MUSIC = 3
-  int output_stream_type_;
-
   // Contains native parameters (e.g. sample rate, channel configuration).
   // Set at construction in OnCacheAudioParameters() which is called from
   // Java on the same thread as this object is created on.
diff --git a/webrtc/modules/audio_device/android/audio_manager_unittest.cc b/webrtc/modules/audio_device/android/audio_manager_unittest.cc
index a5bc840dff..ddae73067a 100644
--- a/webrtc/modules/audio_device/android/audio_manager_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_manager_unittest.cc
@@ -82,14 +82,14 @@ TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
   PRINT("%saudio layer: %s\n", kTag,
         low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
   PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
-  PRINT("%schannels: %d\n", kTag, playout_parameters_.channels());
+  PRINT("%schannels: %" PRIuS "\n", kTag, playout_parameters_.channels());
   PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
         playout_parameters_.frames_per_buffer(),
         playout_parameters_.GetBufferSizeInMilliseconds());
   PRINT("RECORD: \n");
   PRINT("%saudio layer: %s\n", kTag, "Java/JNI based AudioRecord");
   PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
-  PRINT("%schannels: %d\n", kTag, record_parameters_.channels());
+  PRINT("%schannels: %" PRIuS "\n", kTag, record_parameters_.channels());
   PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
         record_parameters_.frames_per_buffer(),
         record_parameters_.GetBufferSizeInMilliseconds());
@@ -119,7 +119,7 @@ TEST_F(AudioManagerTest, AudioParametersWithDefaultConstruction) {
   AudioParameters params;
   EXPECT_FALSE(params.is_valid());
   EXPECT_EQ(0, params.sample_rate());
-  EXPECT_EQ(0, params.channels());
+  EXPECT_EQ(0U, params.channels());
   EXPECT_EQ(0U, params.frames_per_buffer());
   EXPECT_EQ(0U, params.frames_per_10ms_buffer());
   EXPECT_EQ(0U, params.GetBytesPerFrame());
@@ -131,7 +131,7 @@ TEST_F(AudioManagerTest, AudioParametersWithDefaultConstruction) {
 // Basic test of the AudioParameters class using non default construction.
 TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) {
   const int kSampleRate = 48000;
-  const int kChannels = 1;
+  const size_t kChannels = 1;
   const size_t kFramesPerBuffer = 480;
   const size_t kFramesPer10msBuffer = 480;
   const size_t kBytesPerFrame = 2;
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.cc b/webrtc/modules/audio_device/android/audio_record_jni.cc
index ba3212afc3..5dda7249ac 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_record_jni.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 
+#include <utility>
+
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
@@ -28,23 +30,20 @@ namespace webrtc {
 
 // AudioRecordJni::JavaAudioRecord implementation.
 AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
-    NativeRegistration* native_reg, rtc::scoped_ptr<GlobalRef> audio_record)
-    : audio_record_(audio_record.Pass()),
+    NativeRegistration* native_reg,
+    rtc::scoped_ptr<GlobalRef> audio_record)
+    : audio_record_(std::move(audio_record)),
       init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
       start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
       stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
-      enable_built_in_aec_(native_reg->GetMethodId(
-          "enableBuiltInAEC", "(Z)Z")),
-      enable_built_in_agc_(native_reg->GetMethodId(
-          "enableBuiltInAGC", "(Z)Z")),
-      enable_built_in_ns_(native_reg->GetMethodId(
-          "enableBuiltInNS", "(Z)Z")) {
-}
+      enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
+      enable_built_in_agc_(native_reg->GetMethodId("enableBuiltInAGC", "(Z)Z")),
+      enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
 
 AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
 
 int AudioRecordJni::JavaAudioRecord::InitRecording(
-    int sample_rate, int channels) {
+    int sample_rate, size_t channels) {
   return audio_record_->CallIntMethod(init_recording_,
                                       static_cast<jint>(sample_rate),
                                       static_cast<jint>(channels));
@@ -186,8 +185,8 @@ void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   const int sample_rate_hz = audio_parameters_.sample_rate();
   ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
-  const int channels = audio_parameters_.channels();
-  ALOGD("SetRecordingChannels(%d)", channels);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetRecordingChannels(%" PRIuS ")", channels);
   audio_device_buffer_->SetRecordingChannels(channels);
   total_delay_in_milliseconds_ =
       audio_manager_->GetDelayEstimateInMilliseconds();
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.h b/webrtc/modules/audio_device/android/audio_record_jni.h
index efd516425a..766316a83a 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.h
+++ b/webrtc/modules/audio_device/android/audio_record_jni.h
@@ -17,8 +17,8 @@
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
@@ -49,7 +49,7 @@ class AudioRecordJni {
                    rtc::scoped_ptr<GlobalRef> audio_track);
     ~JavaAudioRecord();
 
-    int InitRecording(int sample_rate, int channels);
+    int InitRecording(int sample_rate, size_t channels);
     bool StartRecording();
     bool StopRecording();
     bool EnableBuiltInAEC(bool enable);
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.cc b/webrtc/modules/audio_device/android/audio_track_jni.cc
index 29b21ae998..057e016405 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_track_jni.cc
@@ -11,6 +11,8 @@
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
 
+#include <utility>
+
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
@@ -28,16 +30,16 @@ namespace webrtc {
 
 // AudioTrackJni::JavaAudioTrack implementation.
 AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
-    NativeRegistration* native_reg, rtc::scoped_ptr<GlobalRef> audio_track)
-    : audio_track_(audio_track.Pass()),
+    NativeRegistration* native_reg,
+    rtc::scoped_ptr<GlobalRef> audio_track)
+    : audio_track_(std::move(audio_track)),
       init_playout_(native_reg->GetMethodId("initPlayout", "(II)V")),
       start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
       stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
       set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
-      get_stream_max_volume_(native_reg->GetMethodId(
-          "getStreamMaxVolume", "()I")),
-      get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {
-}
+      get_stream_max_volume_(
+          native_reg->GetMethodId("getStreamMaxVolume", "()I")),
+      get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {}
 
 AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
 
@@ -200,8 +202,8 @@ void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   const int sample_rate_hz = audio_parameters_.sample_rate();
   ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
-  const int channels = audio_parameters_.channels();
-  ALOGD("SetPlayoutChannels(%d)", channels);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
   audio_device_buffer_->SetPlayoutChannels(channels);
 }
 
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.h b/webrtc/modules/audio_device/android/audio_track_jni.h
index 43bfcad657..067dc6c651 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.h
+++ b/webrtc/modules/audio_device/android/audio_track_jni.h
@@ -18,8 +18,8 @@
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/build_info.cc b/webrtc/modules/audio_device/android/build_info.cc
index cb5dc293d7..6289697073 100644
--- a/webrtc/modules/audio_device/android/build_info.cc
+++ b/webrtc/modules/audio_device/android/build_info.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/audio_device/android/build_info.h"
 
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/build_info.h b/webrtc/modules/audio_device/android/build_info.h
index d9b2871841..1490fa0772 100644
--- a/webrtc/modules/audio_device/android/build_info.h
+++ b/webrtc/modules/audio_device/android/build_info.h
@@ -14,7 +14,7 @@
 #include <jni.h>
 #include <string>
 
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/android/ensure_initialized.cc b/webrtc/modules/audio_device/android/ensure_initialized.cc
index e8197b7ca0..b63aec1f27 100644
--- a/webrtc/modules/audio_device/android/ensure_initialized.cc
+++ b/webrtc/modules/audio_device/android/ensure_initialized.cc
@@ -14,11 +14,12 @@
 
 // Note: this dependency is dangerous since it reaches into Chromium's base.
 // There's a risk of e.g. macro clashes. This file may only be used in tests.
+#include "base/android/context_utils.h"
 #include "base/android/jni_android.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 namespace audiodevicemodule {
diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
index 9b90f4ab54..c3ab043868 100644
--- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
+++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
@@ -10,6 +10,7 @@
 
 package org.webrtc.voiceengine;
 
+import android.annotation.TargetApi;
 import android.media.audiofx.AcousticEchoCanceler;
 import android.media.audiofx.AudioEffect;
 import android.media.audiofx.AudioEffect.Descriptor;
@@ -119,6 +120,7 @@ class WebRtcAudioEffects {
 
   // Returns true if the platform AEC should be excluded based on its UUID.
   // AudioEffect.queryEffects() can throw IllegalStateException.
+  @TargetApi(18)
   private static boolean isAcousticEchoCancelerExcludedByUUID() {
     for (Descriptor d : AudioEffect.queryEffects()) {
       if (d.type.equals(AudioEffect.EFFECT_TYPE_AEC) &&
@@ -131,6 +133,7 @@ class WebRtcAudioEffects {
 
   // Returns true if the platform AGC should be excluded based on its UUID.
   // AudioEffect.queryEffects() can throw IllegalStateException.
+  @TargetApi(18)
   private static boolean isAutomaticGainControlExcludedByUUID() {
     for (Descriptor d : AudioEffect.queryEffects()) {
       if (d.type.equals(AudioEffect.EFFECT_TYPE_AGC) &&
@@ -143,6 +146,7 @@ class WebRtcAudioEffects {
 
   // Returns true if the platform NS should be excluded based on its UUID.
   // AudioEffect.queryEffects() can throw IllegalStateException.
+  @TargetApi(18)
   private static boolean isNoiseSuppressorExcludedByUUID() {
     for (Descriptor d : AudioEffect.queryEffects()) {
       if (d.type.equals(AudioEffect.EFFECT_TYPE_NS) &&
@@ -208,15 +212,6 @@ class WebRtcAudioEffects {
 
   private WebRtcAudioEffects() {
     Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
-    for (Descriptor d : AudioEffect.queryEffects()) {
-      if (effectTypeIsVoIP(d.type) || DEBUG) {
-        // Only log information for VoIP effects (AEC, AEC and NS).
-        Logging.d(TAG, "name: " + d.name + ", "
-            + "mode: " + d.connectMode + ", "
-            + "implementor: " + d.implementor + ", "
-            + "UUID: " + d.uuid);
-      }
-    }
   }
 
   // Call this method to enable or disable the platform AEC. It modifies
@@ -282,6 +277,17 @@ class WebRtcAudioEffects {
     assertTrue(agc == null);
     assertTrue(ns == null);
 
+    // Add logging of supported effects but filter out "VoIP effects", i.e.,
+    // AEC, AEC and NS.
+    for (Descriptor d : AudioEffect.queryEffects()) {
+      if (effectTypeIsVoIP(d.type) || DEBUG) {
+        Logging.d(TAG, "name: " + d.name + ", "
+            + "mode: " + d.connectMode + ", "
+            + "implementor: " + d.implementor + ", "
+            + "UUID: " + d.uuid);
+      }
+    }
+
     if (isAcousticEchoCancelerSupported()) {
       // Create an AcousticEchoCanceler and attach it to the AudioRecord on
       // the specified audio session.
@@ -366,7 +372,11 @@ class WebRtcAudioEffects {
   // AudioEffect.Descriptor array that are actually not available on the device.
   // As an example: Samsung Galaxy S6 includes an AGC in the descriptor but
   // AutomaticGainControl.isAvailable() returns false.
+  @TargetApi(18)
   private boolean effectTypeIsVoIP(UUID type) {
+    if (!WebRtcAudioUtils.runningOnJellyBeanMR2OrHigher())
+      return false;
+
     return (AudioEffect.EFFECT_TYPE_AEC.equals(type)
         && isAcousticEchoCancelerSupported())
         || (AudioEffect.EFFECT_TYPE_AGC.equals(type)
diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
index cf2f03a2f1..1213f333d9 100644
--- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
+++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
@@ -10,6 +10,7 @@
 
 package org.webrtc.voiceengine;
 
+import android.annotation.TargetApi;
 import android.content.Context;
 import android.content.pm.PackageManager;
 import android.media.AudioFormat;
@@ -33,11 +34,24 @@ import java.lang.Math;
 // recommended to always use AudioManager.MODE_IN_COMMUNICATION.
 // This class also adds support for output volume control of the
 // STREAM_VOICE_CALL-type stream.
-class WebRtcAudioManager {
+public class WebRtcAudioManager {
   private static final boolean DEBUG = false;
 
   private static final String TAG = "WebRtcAudioManager";
 
+  private static boolean blacklistDeviceForOpenSLESUsage = false;
+  private static boolean blacklistDeviceForOpenSLESUsageIsOverridden = false;
+
+  // Call this method to override the deault list of blacklisted devices
+  // specified in WebRtcAudioUtils.BLACKLISTED_OPEN_SL_ES_MODELS.
+  // Allows an app to take control over which devices to exlude from using
+  // the OpenSL ES audio output path
+  public static synchronized void setBlacklistDeviceForOpenSLESUsage(
+      boolean enable) {
+    blacklistDeviceForOpenSLESUsageIsOverridden = true;
+    blacklistDeviceForOpenSLESUsage = enable;
+  }
+
   // Default audio data format is PCM 16 bit per sample.
   // Guaranteed to be supported by all devices.
   private static final int BITS_PER_SAMPLE = 16;
@@ -71,7 +85,6 @@ class WebRtcAudioManager {
   private int channels;
   private int outputBufferSize;
   private int inputBufferSize;
-  private int outputStreamType;
 
   WebRtcAudioManager(Context context, long nativeAudioManager) {
     Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
@@ -85,7 +98,7 @@ class WebRtcAudioManager {
     storeAudioParameters();
     nativeCacheAudioParameters(
         sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS,
-        lowLatencyOutput, outputBufferSize, inputBufferSize, outputStreamType,
+        lowLatencyOutput, outputBufferSize, inputBufferSize,
         nativeAudioManager);
   }
 
@@ -110,8 +123,9 @@ class WebRtcAudioManager {
     return (audioManager.getMode() == AudioManager.MODE_IN_COMMUNICATION);
   }
 
-   private boolean isDeviceBlacklistedForOpenSLESUsage() {
-    boolean blacklisted =
+  private boolean isDeviceBlacklistedForOpenSLESUsage() {
+    boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden ?
+        blacklistDeviceForOpenSLESUsage :
         WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
     if (blacklisted) {
       Logging.e(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");
@@ -133,8 +147,6 @@ class WebRtcAudioManager {
         getMinOutputFrameSize(sampleRate, channels);
     // TODO(henrika): add support for low-latency input.
     inputBufferSize = getMinInputFrameSize(sampleRate, channels);
-    outputStreamType = WebRtcAudioUtils.getOutputStreamTypeFromAudioMode(
-        audioManager.getMode());
   }
 
   // Gets the current earpiece state.
@@ -178,20 +190,26 @@ class WebRtcAudioManager {
     // No overrides available. Deliver best possible estimate based on default
     // Android AudioManager APIs.
     final int sampleRateHz;
-    if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
-      sampleRateHz = WebRtcAudioUtils.getDefaultSampleRateHz();
+    if (WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
+      sampleRateHz = getSampleRateOnJellyBeanMR10OrHigher();
     } else {
-      String sampleRateString = audioManager.getProperty(
-          AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
-      sampleRateHz = (sampleRateString == null)
-          ? WebRtcAudioUtils.getDefaultSampleRateHz()
-          : Integer.parseInt(sampleRateString);
+      sampleRateHz = WebRtcAudioUtils.getDefaultSampleRateHz();
     }
     Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
     return sampleRateHz;
   }
 
+  @TargetApi(17)
+  private int getSampleRateOnJellyBeanMR10OrHigher() {
+    String sampleRateString = audioManager.getProperty(
+        AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
+    return (sampleRateString == null)
+        ? WebRtcAudioUtils.getDefaultSampleRateHz()
+        : Integer.parseInt(sampleRateString);
+  }
+
   // Returns the native output buffer size for low-latency output streams.
+  @TargetApi(17)
   private int getLowLatencyOutputFramesPerBuffer() {
     assertTrue(isLowLatencyOutputSupported());
     if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
@@ -270,5 +288,5 @@ class WebRtcAudioManager {
   private native void nativeCacheAudioParameters(
     int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC,
     boolean hardwareNS, boolean lowLatencyOutput, int outputBufferSize,
-    int inputBufferSize, int outputStreamType, long nativeAudioManager);
+    int inputBufferSize, long nativeAudioManager);
 }
diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
index 7b31e08eed..ff77635843 100644
--- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
+++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
@@ -192,10 +192,6 @@ class  WebRtcAudioRecord {
         Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
     Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
     try {
-      // TODO(henrika): the only supported audio source for input is currently
-      // AudioSource.VOICE_COMMUNICATION. Is there any reason why we should
-      // support other types, e.g. DEFAULT or MIC? Only reason I can think of
-      // is if the device does not support VOICE_COMMUNICATION.
       audioRecord = new AudioRecord(AudioSource.VOICE_COMMUNICATION,
                                     sampleRate,
                                     AudioFormat.CHANNEL_IN_MONO,
diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
index ec0e109169..11eb51383d 100644
--- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
+++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
@@ -13,6 +13,7 @@ package org.webrtc.voiceengine;
 import java.lang.Thread;
 import java.nio.ByteBuffer;
 
+import android.annotation.TargetApi;
 import android.content.Context;
 import android.media.AudioFormat;
 import android.media.AudioManager;
@@ -39,7 +40,6 @@ class WebRtcAudioTrack {
   private final Context context;
   private final long nativeAudioTrack;
   private final AudioManager audioManager;
-  private final int streamType;
 
   private ByteBuffer byteBuffer;
 
@@ -91,13 +91,9 @@ class WebRtcAudioTrack {
         assertTrue(sizeInBytes <= byteBuffer.remaining());
         int bytesWritten = 0;
         if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
-          bytesWritten = audioTrack.write(byteBuffer,
-                                          sizeInBytes,
-                                          AudioTrack.WRITE_BLOCKING);
+          bytesWritten = writeOnLollipop(audioTrack, byteBuffer, sizeInBytes);
         } else {
-          bytesWritten = audioTrack.write(byteBuffer.array(),
-                                          byteBuffer.arrayOffset(),
-                                          sizeInBytes);
+          bytesWritten = writePreLollipop(audioTrack, byteBuffer, sizeInBytes);
         }
         if (bytesWritten != sizeInBytes) {
           Logging.e(TAG, "AudioTrack.write failed: " + bytesWritten);
@@ -124,6 +120,15 @@ class WebRtcAudioTrack {
       audioTrack.flush();
     }
 
+    @TargetApi(21)
+    private int writeOnLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
+      return audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
+    }
+
+    private int writePreLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
+      return audioTrack.write(byteBuffer.array(), byteBuffer.arrayOffset(), sizeInBytes);
+    }
+
     public void joinThread() {
       keepAlive = false;
       while (isAlive()) {
@@ -142,9 +147,6 @@ class WebRtcAudioTrack {
     this.nativeAudioTrack = nativeAudioTrack;
     audioManager = (AudioManager) context.getSystemService(
         Context.AUDIO_SERVICE);
-    this.streamType =
-        WebRtcAudioUtils.getOutputStreamTypeFromAudioMode(
-            audioManager.getMode());
     if (DEBUG) {
       WebRtcAudioUtils.logDeviceInfo(TAG);
     }
@@ -181,7 +183,7 @@ class WebRtcAudioTrack {
       // Create an AudioTrack object and initialize its associated audio buffer.
       // The size of this buffer determines how long an AudioTrack can play
       // before running out of data.
-      audioTrack = new AudioTrack(streamType,
+      audioTrack = new AudioTrack(AudioManager.STREAM_VOICE_CALL,
                                   sampleRate,
                                   AudioFormat.CHANNEL_OUT_MONO,
                                   AudioFormat.ENCODING_PCM_16BIT,
@@ -193,7 +195,7 @@ class WebRtcAudioTrack {
     }
     assertTrue(audioTrack.getState() == AudioTrack.STATE_INITIALIZED);
     assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_STOPPED);
-    assertTrue(audioTrack.getStreamType() == streamType);
+    assertTrue(audioTrack.getStreamType() == AudioManager.STREAM_VOICE_CALL);
   }
 
   private boolean startPlayout() {
@@ -217,32 +219,37 @@ class WebRtcAudioTrack {
     return true;
   }
 
-  /** Get max possible volume index given type of audio stream. */
+  /** Get max possible volume index for a phone call audio stream. */
   private int getStreamMaxVolume() {
     Logging.d(TAG, "getStreamMaxVolume");
     assertTrue(audioManager != null);
-    return audioManager.getStreamMaxVolume(streamType);
+    return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
   }
 
-  /** Set current volume level given type of audio stream. */
+  /** Set current volume level for a phone call audio stream. */
   private boolean setStreamVolume(int volume) {
     Logging.d(TAG, "setStreamVolume(" + volume + ")");
     assertTrue(audioManager != null);
-    if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
-      if (audioManager.isVolumeFixed()) {
-        Logging.e(TAG, "The device implements a fixed volume policy.");
-        return false;
-      }
+    if (isVolumeFixed()) {
+      Logging.e(TAG, "The device implements a fixed volume policy.");
+      return false;
     }
-    audioManager.setStreamVolume(streamType, volume, 0);
+    audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
     return true;
   }
 
-  /** Get current volume level given type of audio stream. */
+  @TargetApi(21)
+  private boolean isVolumeFixed() {
+    if (!WebRtcAudioUtils.runningOnLollipopOrHigher())
+      return false;
+    return audioManager.isVolumeFixed();
+  }
+
+  /** Get current volume level for a phone call audio stream. */
   private int getStreamVolume() {
     Logging.d(TAG, "getStreamVolume");
     assertTrue(audioManager != null);
-    return audioManager.getStreamVolume(streamType);
+    return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
   }
 
   /** Helper method which throws an exception  when an assertion has failed. */
diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
index f08e11dad8..45f564a4dd 100644
--- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
+++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
@@ -144,6 +144,11 @@ public final class WebRtcAudioUtils {
     return Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN_MR1;
   }
 
+  public static boolean runningOnJellyBeanMR2OrHigher() {
+    // July 24, 2013: Android 4.3. API Level 18.
+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN_MR2;
+  }
+
   public static boolean runningOnLollipopOrHigher() {
     // API Level 21.
     return Build.VERSION.SDK_INT >= Build.VERSION_CODES.LOLLIPOP;
@@ -193,37 +198,5 @@ public final class WebRtcAudioUtils {
         permission,
         Process.myPid(),
         Process.myUid()) == PackageManager.PERMISSION_GRANTED;
-  }
-
-  // Convert the provided audio |mode| into most suitable audio output stream
-  // type. The stream type is used for creating audio streams and for volume
-  // changes. It is essential that the mode and type are in-line to ensure
-  // correct behavior. If for example a STREAM_MUSIC type of stream is created
-  // in a MODE_IN_COMMUNICATION mode, audio will be played out and the volume
-  // icon will look OK but the actual volume will not be changed when the user
-  // changes the volume slider.
-  // TODO(henrika): there is currently no mapping to STREAM_ALARM, STREAM_DTMF,
-  // or STREAM_NOTIFICATION types since I am unable to see a reason for using
-  // them. There are only four different modes.
-  public static int getOutputStreamTypeFromAudioMode(int mode) {
-    Logging.d(TAG, "getOutputStreamTypeFromAudioMode(mode=" + mode + ")");
-    switch (mode) {
-      case AudioManager.MODE_NORMAL:
-        // The audio stream for music playback.
-        Logging.d(TAG, "AudioManager.STREAM_MUSIC");
-        return AudioManager.STREAM_MUSIC;
-      case AudioManager.MODE_RINGTONE:
-        // Audio stream for the phone ring.
-        Logging.d(TAG, "AudioManager.STREAM_RING");
-        return AudioManager.STREAM_RING;
-      case AudioManager.MODE_IN_CALL:
-      case AudioManager.MODE_IN_COMMUNICATION:
-        // Audio stream for phone calls.
-        Logging.d(TAG, "AudioManager.STREAM_VOICE_CALL");
-        return AudioManager.STREAM_VOICE_CALL;
-      default:
-        Logging.d(TAG, "AudioManager.USE_DEFAULT_STREAM_TYPE");
-        return AudioManager.USE_DEFAULT_STREAM_TYPE;
     }
-  }
 }
diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc
index 40967c5fb9..d2bff4905e 100644
--- a/webrtc/modules/audio_device/android/opensles_player.cc
+++ b/webrtc/modules/audio_device/android/opensles_player.cc
@@ -15,6 +15,7 @@
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
 
@@ -38,7 +39,6 @@ namespace webrtc {
 
 OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
     : audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
-      stream_type_(audio_manager->OutputStreamType()),
       audio_device_buffer_(NULL),
       initialized_(false),
       playing_(false),
@@ -47,11 +47,9 @@ OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
       engine_(nullptr),
       player_(nullptr),
       simple_buffer_queue_(nullptr),
-      volume_(nullptr) {
+      volume_(nullptr),
+      last_play_time_(0) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
-  RTC_DCHECK(stream_type_ == SL_ANDROID_STREAM_VOICE ||
-             stream_type_ == SL_ANDROID_STREAM_RING ||
-             stream_type_ == SL_ANDROID_STREAM_MEDIA) << stream_type_;
   // Use native audio output parameters provided by the audio manager and
   // define the PCM format structure.
   pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
@@ -99,6 +97,7 @@ int OpenSLESPlayer::InitPlayout() {
   CreateMix();
   initialized_ = true;
   buffer_index_ = 0;
+  last_play_time_ = rtc::Time();
   return 0;
 }
 
@@ -180,15 +179,15 @@ void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   const int sample_rate_hz = audio_parameters_.sample_rate();
   ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
-  const int channels = audio_parameters_.channels();
-  ALOGD("SetPlayoutChannels(%d)", channels);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
   audio_device_buffer_->SetPlayoutChannels(channels);
   RTC_CHECK(audio_device_buffer_);
   AllocateDataBuffers();
 }
 
 SLDataFormat_PCM OpenSLESPlayer::CreatePCMConfiguration(
-    int channels,
+    size_t channels,
     int sample_rate,
     size_t bits_per_sample) {
   ALOGD("CreatePCMConfiguration");
@@ -237,7 +236,16 @@ void OpenSLESPlayer::AllocateDataBuffers() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(!simple_buffer_queue_);
   RTC_CHECK(audio_device_buffer_);
-  bytes_per_buffer_ = audio_parameters_.GetBytesPerBuffer();
+  // Don't use the lowest possible size as native buffer size. Instead,
+  // use 10ms to better match the frame size that WebRTC uses. It will result
+  // in a reduced risk for audio glitches and also in a more "clean" sequence
+  // of callbacks from the OpenSL ES thread in to WebRTC when asking for audio
+  // to render.
+  ALOGD("lowest possible buffer size: %" PRIuS,
+      audio_parameters_.GetBytesPerBuffer());
+  bytes_per_buffer_ = audio_parameters_.GetBytesPerFrame() *
+      audio_parameters_.frames_per_10ms_buffer();
+  RTC_DCHECK_GE(bytes_per_buffer_, audio_parameters_.GetBytesPerBuffer());
   ALOGD("native buffer size: %" PRIuS, bytes_per_buffer_);
   // Create a modified audio buffer class which allows us to ask for any number
   // of samples (and not only multiple of 10ms) to match the native OpenSL ES
@@ -351,7 +359,7 @@ bool OpenSLESPlayer::CreateAudioPlayer() {
       false);
   // Set audio player configuration to SL_ANDROID_STREAM_VOICE which
   // corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
-  SLint32 stream_type = stream_type_;
+  SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
   RETURN_ON_ERROR(
       (*player_config)
           ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
@@ -422,6 +430,15 @@ void OpenSLESPlayer::FillBufferQueue() {
 }
 
 void OpenSLESPlayer::EnqueuePlayoutData() {
+  // Check delta time between two successive callbacks and provide a warning
+  // if it becomes very large.
+  // TODO(henrika): using 100ms as upper limit but this value is rather random.
+  const uint32_t current_time = rtc::Time();
+  const uint32_t diff = current_time - last_play_time_;
+  if (diff > 100) {
+    ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
+  }
+  last_play_time_ = current_time;
   // Read audio data from the WebRTC source using the FineAudioBuffer object
   // to adjust for differences in buffer size between WebRTC (10ms) and native
   // OpenSL ES.
diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h
index 96b1d49ac5..fa9e931218 100644
--- a/webrtc/modules/audio_device/android/opensles_player.h
+++ b/webrtc/modules/audio_device/android/opensles_player.h
@@ -22,7 +22,7 @@
 #include "webrtc/modules/audio_device/android/opensles_common.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 namespace webrtc {
 
@@ -52,7 +52,7 @@ class OpenSLESPlayer {
   // buffer count of 2 or more, and a buffer size and sample rate that are
   // compatible with the device's native output configuration provided via the
   // audio manager at construction.
-  static const int kNumOfOpenSLESBuffers = 2;
+  static const int kNumOfOpenSLESBuffers = 4;
 
   // There is no need for this class to use JNI.
   static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* context) {
@@ -94,7 +94,7 @@ class OpenSLESPlayer {
   void EnqueuePlayoutData();
 
   // Configures the SL_DATAFORMAT_PCM structure.
-  SLDataFormat_PCM CreatePCMConfiguration(int channels,
+  SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
                                           int sample_rate,
                                           size_t bits_per_sample);
 
@@ -130,20 +130,6 @@ class OpenSLESPlayer {
   // AudioManager.
   const AudioParameters audio_parameters_;
 
-  // Contains the stream type provided to this class at construction by the
-  // AudioManager. Possible input values are:
-  //  - AudioManager.STREAM_VOICE_CALL = 0
-  //  - AudioManager.STREAM_RING = 2
-  //  - AudioManager.STREAM_MUSIC = 3
-  // These value are mapped to the corresponding audio playback stream type
-  // values in the "OpenSL ES domain":
-  // - SL_ANDROID_STREAM_VOICE <=> STREAM_VOICE_CALL (0)
-  // - SL_ANDROID_STREAM_RING <=> STREAM_RING (2)
-  // - SL_ANDROID_STREAM_MEDIA <=> STREAM_MUSIC (3)
-  // when creating the audio player. See SLES/OpenSLES_AndroidConfiguration.h
-  // for details.
-  const int stream_type_;
-
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
   // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
   AudioDeviceBuffer* audio_device_buffer_;
@@ -209,6 +195,9 @@ class OpenSLESPlayer {
   // This interface exposes controls for manipulating the object’s audio volume
   // properties. This interface is supported on the Audio Player object.
   SLVolumeItf volume_;
+
+  // Last time the OpenSL ES layer asked for audio data to play out.
+  uint32_t last_play_time_;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_device/audio_device.gypi b/webrtc/modules/audio_device/audio_device.gypi
index 0678d33802..2db9f1e17b 100644
--- a/webrtc/modules/audio_device/audio_device.gypi
+++ b/webrtc/modules/audio_device/audio_device.gypi
@@ -20,13 +20,13 @@
       ],
       'include_dirs': [
         '.',
-        '../interface',
+        '../include',
         'include',
         'dummy',  # Contains dummy audio device implementations.
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-          '../interface',
+          '../include',
           'include',
         ],
       },
@@ -86,48 +86,26 @@
         }],
         ['include_internal_audio_device==1', {
           'sources': [
-            'android/audio_device_template.h',
-            'android/audio_manager.cc',
-            'android/audio_manager.h',
-            'android/audio_record_jni.cc',
-            'android/audio_record_jni.h',
-            'android/audio_track_jni.cc',
-            'android/audio_track_jni.h',
-            'android/build_info.cc',
-            'android/build_info.h',
-            'android/opensles_common.cc',
-            'android/opensles_common.h',
-            'android/opensles_player.cc',
-            'android/opensles_player.h',
             'audio_device_impl.cc',
             'audio_device_impl.h',
-            'ios/audio_device_ios.h',
-            'ios/audio_device_ios.mm',
-            'ios/audio_device_not_implemented_ios.mm',
-            'linux/alsasymboltable_linux.cc',
-            'linux/alsasymboltable_linux.h',
-            'linux/audio_device_alsa_linux.cc',
-            'linux/audio_device_alsa_linux.h',
-            'linux/audio_mixer_manager_alsa_linux.cc',
-            'linux/audio_mixer_manager_alsa_linux.h',
-            'linux/latebindingsymboltable_linux.cc',
-            'linux/latebindingsymboltable_linux.h',
-            'mac/audio_device_mac.cc',
-            'mac/audio_device_mac.h',
-            'mac/audio_mixer_manager_mac.cc',
-            'mac/audio_mixer_manager_mac.h',
-            'mac/portaudio/pa_memorybarrier.h',
-            'mac/portaudio/pa_ringbuffer.c',
-            'mac/portaudio/pa_ringbuffer.h',
-            'win/audio_device_core_win.cc',
-            'win/audio_device_core_win.h',
-            'win/audio_device_wave_win.cc',
-            'win/audio_device_wave_win.h',
-            'win/audio_mixer_manager_win.cc',
-            'win/audio_mixer_manager_win.h',
           ],
           'conditions': [
             ['OS=="android"', {
+              'sources': [
+                'android/audio_device_template.h',
+                'android/audio_manager.cc',
+                'android/audio_manager.h',
+                'android/audio_record_jni.cc',
+                'android/audio_record_jni.h',
+                'android/audio_track_jni.cc',
+                'android/audio_track_jni.h',
+                'android/build_info.cc',
+                'android/build_info.h',
+                'android/opensles_common.cc',
+                'android/opensles_common.h',
+                'android/opensles_player.cc',
+                'android/opensles_player.h',
+              ],
               'link_settings': {
                 'libraries': [
                   '-llog',
@@ -136,6 +114,16 @@
               },
             }],
             ['OS=="linux"', {
+              'sources': [
+                'linux/alsasymboltable_linux.cc',
+                'linux/alsasymboltable_linux.h',
+                'linux/audio_device_alsa_linux.cc',
+                'linux/audio_device_alsa_linux.h',
+                'linux/audio_mixer_manager_alsa_linux.cc',
+                'linux/audio_mixer_manager_alsa_linux.h',
+                'linux/latebindingsymboltable_linux.cc',
+                'linux/latebindingsymboltable_linux.h',
+              ],
               'defines': [
                 'LINUX_ALSA',
               ],
@@ -161,6 +149,15 @@
               ],
             }],
             ['OS=="mac"', {
+              'sources': [
+                'mac/audio_device_mac.cc',
+                'mac/audio_device_mac.h',
+                'mac/audio_mixer_manager_mac.cc',
+                'mac/audio_mixer_manager_mac.h',
+                'mac/portaudio/pa_memorybarrier.h',
+                'mac/portaudio/pa_ringbuffer.c',
+                'mac/portaudio/pa_ringbuffer.h',
+              ],
               'link_settings': {
                 'libraries': [
                   '$(SDKROOT)/System/Library/Frameworks/AudioToolbox.framework',
@@ -169,6 +166,11 @@
               },
             }],
             ['OS=="ios"', {
+              'sources': [
+                'ios/audio_device_ios.h',
+                'ios/audio_device_ios.mm',
+                'ios/audio_device_not_implemented_ios.mm',
+              ],
               'xcode_settings': {
                 'CLANG_ENABLE_OBJC_ARC': 'YES',
               },
@@ -184,6 +186,14 @@
               },
             }],
             ['OS=="win"', {
+              'sources': [
+                'win/audio_device_core_win.cc',
+                'win/audio_device_core_win.h',
+                'win/audio_device_wave_win.cc',
+                'win/audio_device_wave_win.h',
+                'win/audio_mixer_manager_win.cc',
+                'win/audio_mixer_manager_win.h',
+              ],
               'link_settings': {
                 'libraries': [
                   # Required for the built-in WASAPI AEC.
@@ -194,18 +204,40 @@
                 ],
               },
             }],
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-bool-conversion',
+                    '-Wno-delete-non-virtual-dtor',
+                    '-Wno-logical-op-parentheses',
+                    '-Wno-microsoft-extra-qualification',
+                    '-Wno-microsoft-goto',
+                    '-Wno-missing-braces',
+                    '-Wno-parentheses-equality',
+                    '-Wno-reorder',
+                    '-Wno-shift-overflow',
+                    '-Wno-tautological-compare',
+                    '-Wno-unused-private-field',
+                  ],
+                },
+              },
+            }],
           ], # conditions
         }], # include_internal_audio_device==1
       ], # conditions
     },
   ],
   'conditions': [
-    ['include_tests==1', {
+    # Does not compile on iOS: webrtc:4755.
+    ['include_tests==1 and OS!="ios"', {
       'targets': [
         {
           'target_name': 'audio_device_tests',
-         'type': 'executable',
-         'dependencies': [
+          'type': 'executable',
+          'dependencies': [
             'audio_device',
             'webrtc_utility',
             '<(webrtc_root)/test/test.gyp:test_support_main',
@@ -236,7 +268,7 @@
           ],
         },
       ], # targets
-    }], # include_tests
+    }], # include_tests==1 and OS!=ios
   ],
 }
 
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index e7b487d687..48ae88ee90 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -169,7 +169,7 @@ int32_t AudioDeviceBuffer::PlayoutSampleRate() const
 //  SetRecordingChannels
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
+int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels)
 {
     CriticalSectionScoped lock(&_critSect);
     _recChannels = channels;
@@ -181,7 +181,7 @@ int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
 //  SetPlayoutChannels
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceBuffer::SetPlayoutChannels(uint8_t channels)
+int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels)
 {
     CriticalSectionScoped lock(&_critSect);
     _playChannels = channels;
@@ -239,7 +239,7 @@ int32_t AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& chan
 //  RecordingChannels
 // ----------------------------------------------------------------------------
 
-uint8_t AudioDeviceBuffer::RecordingChannels() const
+size_t AudioDeviceBuffer::RecordingChannels() const
 {
     return _recChannels;
 }
@@ -248,7 +248,7 @@ uint8_t AudioDeviceBuffer::RecordingChannels() const
 //  PlayoutChannels
 // ----------------------------------------------------------------------------
 
-uint8_t AudioDeviceBuffer::PlayoutChannels() const
+size_t AudioDeviceBuffer::PlayoutChannels() const
 {
     return _playChannels;
 }
@@ -487,7 +487,7 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples)
 {
     uint32_t playSampleRate = 0;
     size_t playBytesPerSample = 0;
-    uint8_t playChannels = 0;
+    size_t playChannels = 0;
     {
         CriticalSectionScoped lock(&_critSect);
 
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
index 2ab7ff5547..1095971040 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -40,10 +40,10 @@ public:
     int32_t RecordingSampleRate() const;
     int32_t PlayoutSampleRate() const;
 
-    virtual int32_t SetRecordingChannels(uint8_t channels);
-    virtual int32_t SetPlayoutChannels(uint8_t channels);
-    uint8_t RecordingChannels() const;
-    uint8_t PlayoutChannels() const;
+    virtual int32_t SetRecordingChannels(size_t channels);
+    virtual int32_t SetPlayoutChannels(size_t channels);
+    size_t RecordingChannels() const;
+    size_t PlayoutChannels() const;
     int32_t SetRecordingChannel(
         const AudioDeviceModule::ChannelType channel);
     int32_t RecordingChannel(
@@ -80,8 +80,8 @@ private:
     uint32_t                  _recSampleRate;
     uint32_t                  _playSampleRate;
 
-    uint8_t                   _recChannels;
-    uint8_t                   _playChannels;
+    size_t                   _recChannels;
+    size_t                   _playChannels;
 
     // selected recording channel (left/right/both)
     AudioDeviceModule::ChannelType _recChannel;
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc
index 9c7bf069d8..aac0962a50 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -7,21 +7,20 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include <iostream>
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
 #include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 namespace webrtc {
 
-int kRecordingFixedSampleRate = 48000;
-int kRecordingNumChannels = 2;
-int kPlayoutFixedSampleRate = 48000;
-int kPlayoutNumChannels = 2;
-int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100
-                         * kPlayoutNumChannels * 2;
-int kRecordingBufferSize = kRecordingFixedSampleRate / 100
-                           * kRecordingNumChannels * 2;
+const int kRecordingFixedSampleRate = 48000;
+const size_t kRecordingNumChannels = 2;
+const int kPlayoutFixedSampleRate = 48000;
+const size_t kPlayoutNumChannels = 2;
+const size_t kPlayoutBufferSize =
+    kPlayoutFixedSampleRate / 100 * kPlayoutNumChannels * 2;
+const size_t kRecordingBufferSize =
+    kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2;
 
 FileAudioDevice::FileAudioDevice(const int32_t id,
                                  const char* inputFilename,
@@ -195,9 +194,7 @@ int32_t FileAudioDevice::StartPlayout() {
   _playoutFramesLeft = 0;
 
   if (!_playoutBuffer) {
-      _playoutBuffer = new int8_t[2 *
-                                  kPlayoutNumChannels *
-                                  kPlayoutFixedSampleRate/100];
+      _playoutBuffer = new int8_t[kPlayoutBufferSize];
   }
   if (!_playoutBuffer) {
     _playing = false;
@@ -214,17 +211,10 @@ int32_t FileAudioDevice::StartPlayout() {
     return -1;
   }
 
-  const char* threadName = "webrtc_audio_module_play_thread";
-  _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
-                                               threadName);
-  if (!_ptrThreadPlay->Start()) {
-      _ptrThreadPlay.reset();
-      _playing = false;
-      delete [] _playoutBuffer;
-      _playoutBuffer = NULL;
-      return -1;
-  }
-  _ptrThreadPlay->SetPriority(kRealtimePriority);
+  _ptrThreadPlay.reset(new rtc::PlatformThread(
+      PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+  _ptrThreadPlay->Start();
+  _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
   return 0;
 }
 
@@ -277,17 +267,11 @@ int32_t FileAudioDevice::StartRecording() {
     return -1;
   }
 
-  const char* threadName = "webrtc_audio_module_capture_thread";
-  _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this, threadName);
+  _ptrThreadRec.reset(new rtc::PlatformThread(
+      RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
 
-  if (!_ptrThreadRec->Start()) {
-      _ptrThreadRec.reset();
-      _recording = false;
-      delete [] _recordingBuffer;
-      _recordingBuffer = NULL;
-      return -1;
-  }
-  _ptrThreadRec->SetPriority(kRealtimePriority);
+  _ptrThreadRec->Start();
+  _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
   return 0;
 }
@@ -514,7 +498,12 @@ bool FileAudioDevice::PlayThreadProcess()
     }
     _playoutFramesLeft = 0;
     _critSect.Leave();
-    SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
+
+    uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
+    if(deltaTimeMillis < 10) {
+      SleepMs(10 - deltaTimeMillis);
+    }
+
     return true;
 }
 
@@ -544,7 +533,12 @@ bool FileAudioDevice::RecThreadProcess()
     }
 
     _critSect.Leave();
-    SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
+
+    uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
+    if(deltaTimeMillis < 10) {
+      SleepMs(10 - deltaTimeMillis);
+    }
+
     return true;
 }
 
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.h b/webrtc/modules/audio_device/dummy/file_audio_device.h
index 0e1665ea72..77179409ea 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.h
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.h
@@ -20,9 +20,12 @@
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
+namespace rtc {
+class PlatformThread;
+}  // namespace rtc
+
 namespace webrtc {
 class EventWrapper;
-class ThreadWrapper;
 
 // This is a fake audio device which plays audio from a file as its microphone
 // and plays out into a file.
@@ -178,8 +181,9 @@ class FileAudioDevice : public AudioDeviceGeneric {
   size_t _recordingFramesIn10MS;
   size_t _playoutFramesIn10MS;
 
-  rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
-  rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
+  // TODO(pbos): Make plain members instead of pointers and stop resetting them.
+  rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
+  rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
   bool _playing;
   bool _recording;
diff --git a/webrtc/modules/audio_device/include/audio_device.h b/webrtc/modules/audio_device/include/audio_device.h
index c2c2b88103..15e08730c7 100644
--- a/webrtc/modules/audio_device/include/audio_device.h
+++ b/webrtc/modules/audio_device/include/audio_device.h
@@ -12,7 +12,7 @@
 #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
 
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_device/include/audio_device_defines.h b/webrtc/modules/audio_device/include/audio_device_defines.h
index 3ebbd23cc5..b847729f05 100644
--- a/webrtc/modules/audio_device/include/audio_device_defines.h
+++ b/webrtc/modules/audio_device/include/audio_device_defines.h
@@ -49,7 +49,7 @@ class AudioTransport {
   virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
                                           const size_t nSamples,
                                           const size_t nBytesPerSample,
-                                          const uint8_t nChannels,
+                                          const size_t nChannels,
                                           const uint32_t samplesPerSec,
                                           const uint32_t totalDelayMS,
                                           const int32_t clockDrift,
@@ -59,7 +59,7 @@ class AudioTransport {
 
   virtual int32_t NeedMorePlayData(const size_t nSamples,
                                    const size_t nBytesPerSample,
-                                   const uint8_t nChannels,
+                                   const size_t nChannels,
                                    const uint32_t samplesPerSec,
                                    void* audioSamples,
                                    size_t& nSamplesOut,
@@ -82,10 +82,10 @@ class AudioTransport {
   // TODO(xians): Remove this interface after Chrome and Libjingle switches
   // to OnData().
   virtual int OnDataAvailable(const int voe_channels[],
-                              int number_of_voe_channels,
+                              size_t number_of_voe_channels,
                               const int16_t* audio_data,
                               int sample_rate,
-                              int number_of_channels,
+                              size_t number_of_channels,
                               size_t number_of_frames,
                               int audio_delay_milliseconds,
                               int current_volume,
@@ -103,7 +103,7 @@ class AudioTransport {
                       const void* audio_data,
                       int bits_per_sample,
                       int sample_rate,
-                      int number_of_channels,
+                      size_t number_of_channels,
                       size_t number_of_frames) {}
 
   // Method to push the captured audio data to the specific VoE channel.
@@ -116,7 +116,7 @@ class AudioTransport {
                                const void* audio_data,
                                int bits_per_sample,
                                int sample_rate,
-                               int number_of_channels,
+                               size_t number_of_channels,
                                size_t number_of_frames) {}
 
   // Method to pull mixed render audio data from all active VoE channels.
@@ -125,7 +125,7 @@ class AudioTransport {
   // channel.
   virtual void PullRenderData(int bits_per_sample,
                               int sample_rate,
-                              int number_of_channels,
+                              size_t number_of_channels,
                               size_t number_of_frames,
                               void* audio_data,
                               int64_t* elapsed_time_ms,
@@ -149,27 +149,27 @@ class AudioParameters {
         channels_(0),
         frames_per_buffer_(0),
         frames_per_10ms_buffer_(0) {}
-  AudioParameters(int sample_rate, int channels, size_t frames_per_buffer)
+  AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
       : sample_rate_(sample_rate),
         channels_(channels),
         frames_per_buffer_(frames_per_buffer),
         frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
-  void reset(int sample_rate, int channels, size_t frames_per_buffer) {
+  void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
     sample_rate_ = sample_rate;
     channels_ = channels;
     frames_per_buffer_ = frames_per_buffer;
     frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
   }
   size_t bits_per_sample() const { return kBitsPerSample; }
-  void reset(int sample_rate, int channels, double ms_per_buffer) {
+  void reset(int sample_rate, size_t channels, double ms_per_buffer) {
     reset(sample_rate, channels,
           static_cast<size_t>(sample_rate * ms_per_buffer + 0.5));
   }
-  void reset(int sample_rate, int channels) {
+  void reset(int sample_rate, size_t channels) {
     reset(sample_rate, channels, static_cast<size_t>(0));
   }
   int sample_rate() const { return sample_rate_; }
-  int channels() const { return channels_; }
+  size_t channels() const { return channels_; }
   size_t frames_per_buffer() const { return frames_per_buffer_; }
   size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
   size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
@@ -200,7 +200,7 @@ class AudioParameters {
 
  private:
   int sample_rate_;
-  int channels_;
+  size_t channels_;
   size_t frames_per_buffer_;
   size_t frames_per_10ms_buffer_;
 };
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.h b/webrtc/modules/audio_device/ios/audio_device_ios.h
index 8f8ba0a9c5..c4eb0d6f64 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.h
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.h
@@ -182,7 +182,10 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
   bool InitPlayOrRecord();
 
   // Closes and deletes the voice-processing I/O unit.
-  bool ShutdownPlayOrRecord();
+  void ShutdownPlayOrRecord();
+
+  // Helper method for destroying the existing audio unit.
+  void DisposeAudioUnit();
 
   // Callback function called on a real-time priority I/O thread from the audio
   // unit. This method is used to signal that recorded audio is available.
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index f26e9f1cc7..f6dee5b3cf 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -19,12 +19,24 @@
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
-#include "webrtc/modules/utility/interface/helpers_ios.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
 
 namespace webrtc {
 
+// Protects |g_audio_session_users|.
+static rtc::GlobalLockPod g_lock;
+
+// Counts number of users (=instances of this object) who needs an active
+// audio session. This variable is used to ensure that we only activate an audio
+// session for the first user and deactivate it for the last.
+// Member is static to ensure that the value is counted for all instances
+// and not per instance.
+static int g_audio_session_users GUARDED_BY(g_lock) = 0;
+
 #define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
 
 #define LOG_AND_RETURN_IF_ERROR(error, message) \
@@ -74,25 +86,62 @@ const UInt32 kBytesPerSample = 2;
 // Can most likely be removed.
 const UInt16 kFixedPlayoutDelayEstimate = 30;
 const UInt16 kFixedRecordDelayEstimate = 30;
+// Calls to AudioUnitInitialize() can fail if called back-to-back on different
+// ADM instances. A fall-back solution is to allow multiple sequential calls
+// with as small delay between each. This factor sets the max number of allowed
+// initialization attempts.
+const int kMaxNumberOfAudioUnitInitializeAttempts = 5;
+
 
 using ios::CheckAndLogError;
 
+// Verifies that the current audio session supports input audio and that the
+// required category and mode are enabled.
+static bool VerifyAudioSession(AVAudioSession* session) {
+  LOG(LS_INFO) << "VerifyAudioSession";
+  // Ensure that the device currently supports audio input.
+  if (!session.isInputAvailable) {
+    LOG(LS_ERROR) << "No audio input path is available!";
+    return false;
+  }
+
+  // Ensure that the required category and mode are actually activated.
+  if (![session.category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) {
+    LOG(LS_ERROR)
+        << "Failed to set category to AVAudioSessionCategoryPlayAndRecord";
+    return false;
+  }
+  if (![session.mode isEqualToString:AVAudioSessionModeVoiceChat]) {
+    LOG(LS_ERROR) << "Failed to set mode to AVAudioSessionModeVoiceChat";
+    return false;
+  }
+  return true;
+}
+
 // Activates an audio session suitable for full duplex VoIP sessions when
 // |activate| is true. Also sets the preferred sample rate and IO buffer
 // duration. Deactivates an active audio session if |activate| is set to false.
-static void ActivateAudioSession(AVAudioSession* session, bool activate) {
+static bool ActivateAudioSession(AVAudioSession* session, bool activate)
+    EXCLUSIVE_LOCKS_REQUIRED(g_lock) {
   LOG(LS_INFO) << "ActivateAudioSession(" << activate << ")";
   @autoreleasepool {
     NSError* error = nil;
     BOOL success = NO;
 
-    // Deactivate the audio session and return if |activate| is false.
     if (!activate) {
-      success = [session setActive:NO error:&error];
-      RTC_DCHECK(CheckAndLogError(success, error));
-      return;
+      // Deactivate the audio session using an extra option and then return.
+      // AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation is used to
+      // ensure that other audio sessions that were interrupted by our session
+      // can return to their active state. It is recommended for VoIP apps to
+      // use this option.
+      success = [session
+            setActive:NO
+          withOptions:AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation
+                error:&error];
+      return CheckAndLogError(success, error);
     }
 
+    // Go ahead and active our own audio session since |activate| is true.
     // Use a category which supports simultaneous recording and playback.
     // By default, using this category implies that our app’s audio is
     // nonmixable, hence activating the session will interrupt any other
@@ -121,7 +170,6 @@ static void ActivateAudioSession(AVAudioSession* session, bool activate) {
     RTC_DCHECK(CheckAndLogError(success, error));
 
     // Set the preferred audio I/O buffer duration, in seconds.
-    // TODO(henrika): add more comments here.
     error = nil;
     success = [session setPreferredIOBufferDuration:kPreferredIOBufferDuration
                                               error:&error];
@@ -131,13 +179,15 @@ static void ActivateAudioSession(AVAudioSession* session, bool activate) {
     // session (e.g. phone call) has higher priority than ours.
     error = nil;
     success = [session setActive:YES error:&error];
-    RTC_DCHECK(CheckAndLogError(success, error));
-    RTC_CHECK(session.isInputAvailable) << "No input path is available!";
+    if (!CheckAndLogError(success, error)) {
+      return false;
+    }
 
-    // Ensure that category and mode are actually activated.
-    RTC_DCHECK(
-        [session.category isEqualToString:AVAudioSessionCategoryPlayAndRecord]);
-    RTC_DCHECK([session.mode isEqualToString:AVAudioSessionModeVoiceChat]);
+    // Ensure that the active audio session has the correct category and mode.
+    if (!VerifyAudioSession(session)) {
+      LOG(LS_ERROR) << "Failed to verify audio session category and mode";
+      return false;
+    }
 
     // Try to set the preferred number of hardware audio channels. These calls
     // must be done after setting the audio session’s category and mode and
@@ -156,7 +206,52 @@ static void ActivateAudioSession(AVAudioSession* session, bool activate) {
         [session setPreferredOutputNumberOfChannels:kPreferredNumberOfChannels
                                               error:&error];
     RTC_DCHECK(CheckAndLogError(success, error));
+    return true;
+  }
+}
+
+// An application can create more than one ADM and start audio streaming
+// for all of them. It is essential that we only activate the app's audio
+// session once (for the first one) and deactivate it once (for the last).
+static bool ActivateAudioSession() {
+  LOGI() << "ActivateAudioSession";
+  rtc::GlobalLockScope ls(&g_lock);
+  if (g_audio_session_users == 0) {
+    // The system provides an audio session object upon launch of an
+    // application. However, we must initialize the session in order to
+    // handle interruptions. Implicit initialization occurs when obtaining
+    // a reference to the AVAudioSession object.
+    AVAudioSession* session = [AVAudioSession sharedInstance];
+    // Try to activate the audio session and ask for a set of preferred audio
+    // parameters.
+    if (!ActivateAudioSession(session, true)) {
+      LOG(LS_ERROR) << "Failed to activate the audio session";
+      return false;
+    }
+    LOG(LS_INFO) << "The audio session is now activated";
+  }
+  ++g_audio_session_users;
+  LOG(LS_INFO) << "Number of audio session users: " << g_audio_session_users;
+  return true;
+}
+
+// If more than one object is using the audio session, ensure that only the
+// last object deactivates. Apple recommends: "activate your audio session
+// only as needed and deactivate it when you are not using audio".
+static bool DeactivateAudioSession() {
+  LOGI() << "DeactivateAudioSession";
+  rtc::GlobalLockScope ls(&g_lock);
+  if (g_audio_session_users == 1) {
+    AVAudioSession* session = [AVAudioSession sharedInstance];
+    if (!ActivateAudioSession(session, false)) {
+      LOG(LS_ERROR) << "Failed to deactivate the audio session";
+      return false;
+    }
+    LOG(LS_INFO) << "Our audio session is now deactivated";
   }
+  --g_audio_session_users;
+  LOG(LS_INFO) << "Number of audio session users: " << g_audio_session_users;
+  return true;
 }
 
 #if !defined(NDEBUG)
@@ -198,12 +293,13 @@ AudioDeviceIOS::AudioDeviceIOS()
       initialized_(false),
       rec_is_initialized_(false),
       play_is_initialized_(false),
-      audio_interruption_observer_(nullptr) {
+      audio_interruption_observer_(nullptr),
+      route_change_observer_(nullptr) {
   LOGI() << "ctor" << ios::GetCurrentThreadDescription();
 }
 
 AudioDeviceIOS::~AudioDeviceIOS() {
-  LOGI() << "~dtor";
+  LOGI() << "~dtor" << ios::GetCurrentThreadDescription();
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Terminate();
 }
@@ -245,8 +341,16 @@ int32_t AudioDeviceIOS::Terminate() {
   if (!initialized_) {
     return 0;
   }
-  ShutdownPlayOrRecord();
+  StopPlayout();
+  StopRecording();
   initialized_ = false;
+  {
+    rtc::GlobalLockScope ls(&g_lock);
+    if (g_audio_session_users != 0) {
+      LOG(LS_WARNING) << "Object is destructed with an active audio session";
+    }
+    RTC_DCHECK_GE(g_audio_session_users, 0);
+  }
   return 0;
 }
 
@@ -258,7 +362,7 @@ int32_t AudioDeviceIOS::InitPlayout() {
   RTC_DCHECK(!playing_);
   if (!rec_is_initialized_) {
     if (!InitPlayOrRecord()) {
-      LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
+      LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitPlayout!";
       return -1;
     }
   }
@@ -274,7 +378,7 @@ int32_t AudioDeviceIOS::InitRecording() {
   RTC_DCHECK(!recording_);
   if (!play_is_initialized_) {
     if (!InitPlayOrRecord()) {
-      LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
+      LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitRecording!";
       return -1;
     }
   }
@@ -291,9 +395,11 @@ int32_t AudioDeviceIOS::StartPlayout() {
   if (!recording_) {
     OSStatus result = AudioOutputUnitStart(vpio_unit_);
     if (result != noErr) {
-      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
+      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed for StartPlayout: "
+                      << result;
       return -1;
     }
+    LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
   }
   rtc::AtomicOps::ReleaseStore(&playing_, 1);
   return 0;
@@ -322,9 +428,11 @@ int32_t AudioDeviceIOS::StartRecording() {
   if (!playing_) {
     OSStatus result = AudioOutputUnitStart(vpio_unit_);
     if (result != noErr) {
-      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
+      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed for StartRecording: "
+                      << result;
       return -1;
     }
+    LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
   }
   rtc::AtomicOps::ReleaseStore(&recording_, 1);
   return 0;
@@ -474,11 +582,12 @@ void AudioDeviceIOS::RegisterNotificationObservers() {
             LOG(LS_INFO) << " OldDeviceUnavailable";
             break;
           case AVAudioSessionRouteChangeReasonCategoryChange:
+            // It turns out that we see this notification (at least in iOS 9.2)
+            // when making a switch from a BT device to e.g. Speaker using the
+            // iOS Control Center and that we therefore must check if the sample
+            // rate has changed. And if so is the case, restart the audio unit.
             LOG(LS_INFO) << " CategoryChange";
             LOG(LS_INFO) << " New category: " << ios::GetAudioSessionCategory();
-            // Don't see this as route change since it can be triggered in
-            // combination with session interruptions as well.
-            valid_route_change = false;
             break;
           case AVAudioSessionRouteChangeReasonOverride:
             LOG(LS_INFO) << " Override";
@@ -490,9 +599,11 @@ void AudioDeviceIOS::RegisterNotificationObservers() {
             LOG(LS_INFO) << " NoSuitableRouteForCategory";
             break;
           case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
-            // Ignore this type of route change since we are focusing
+            // The set of input and output ports has not changed, but their
+            // configuration has, e.g., a port’s selected data source has
+            // changed. Ignore this type of route change since we are focusing
             // on detecting headset changes.
-            LOG(LS_INFO) << " RouteConfigurationChange";
+            LOG(LS_INFO) << " RouteConfigurationChange (ignored)";
             valid_route_change = false;
             break;
         }
@@ -630,7 +741,7 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
 
 bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
   LOGI() << "SetupAndInitializeVoiceProcessingAudioUnit";
-  RTC_DCHECK(!vpio_unit_);
+  RTC_DCHECK(!vpio_unit_) << "VoiceProcessingIO audio unit already exists";
   // Create an audio component description to identify the Voice-Processing
   // I/O audio unit.
   AudioComponentDescription vpio_unit_description;
@@ -639,34 +750,48 @@ bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
   vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
   vpio_unit_description.componentFlags = 0;
   vpio_unit_description.componentFlagsMask = 0;
+
   // Obtain an audio unit instance given the description.
   AudioComponent found_vpio_unit_ref =
       AudioComponentFindNext(nullptr, &vpio_unit_description);
 
   // Create a Voice-Processing IO audio unit.
-  LOG_AND_RETURN_IF_ERROR(
-      AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_),
-      "Failed to create a VoiceProcessingIO audio unit");
+  OSStatus result = noErr;
+  result = AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_);
+  if (result != noErr) {
+    vpio_unit_ = nullptr;
+    LOG(LS_ERROR) << "AudioComponentInstanceNew failed: " << result;
+    return false;
+  }
 
   // A VP I/O unit's bus 1 connects to input hardware (microphone). Enable
   // input on the input scope of the input element.
   AudioUnitElement input_bus = 1;
   UInt32 enable_input = 1;
-  LOG_AND_RETURN_IF_ERROR(
-      AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
-                           kAudioUnitScope_Input, input_bus, &enable_input,
-                           sizeof(enable_input)),
-      "Failed to enable input on input scope of input element");
+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Input, input_bus, &enable_input,
+                                sizeof(enable_input));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to enable input on input scope of input element: "
+                  << result;
+    return false;
+  }
 
   // A VP I/O unit's bus 0 connects to output hardware (speaker). Enable
   // output on the output scope of the output element.
   AudioUnitElement output_bus = 0;
   UInt32 enable_output = 1;
-  LOG_AND_RETURN_IF_ERROR(
-      AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
-                           kAudioUnitScope_Output, output_bus, &enable_output,
-                           sizeof(enable_output)),
-      "Failed to enable output on output scope of output element");
+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Output, output_bus,
+                                &enable_output, sizeof(enable_output));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR)
+        << "Failed to enable output on output scope of output element: "
+        << result;
+    return false;
+  }
 
   // Set the application formats for input and output:
   // - use same format in both directions
@@ -694,38 +819,55 @@ bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
 #endif
 
   // Set the application format on the output scope of the input element/bus.
-  LOG_AND_RETURN_IF_ERROR(
-      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
-                           kAudioUnitScope_Output, input_bus,
-                           &application_format, size),
-      "Failed to set application format on output scope of input element");
+  result = AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                                kAudioUnitScope_Output, input_bus,
+                                &application_format, size);
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR)
+        << "Failed to set application format on output scope of input bus: "
+        << result;
+    return false;
+  }
 
   // Set the application format on the input scope of the output element/bus.
-  LOG_AND_RETURN_IF_ERROR(
-      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
-                           kAudioUnitScope_Input, output_bus,
-                           &application_format, size),
-      "Failed to set application format on input scope of output element");
+  result = AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                                kAudioUnitScope_Input, output_bus,
+                                &application_format, size);
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR)
+        << "Failed to set application format on input scope of output bus: "
+        << result;
+    return false;
+  }
 
   // Specify the callback function that provides audio samples to the audio
   // unit.
   AURenderCallbackStruct render_callback;
   render_callback.inputProc = GetPlayoutData;
   render_callback.inputProcRefCon = this;
-  LOG_AND_RETURN_IF_ERROR(
-      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_SetRenderCallback,
-                           kAudioUnitScope_Input, output_bus, &render_callback,
-                           sizeof(render_callback)),
-      "Failed to specify the render callback on the output element");
+  result = AudioUnitSetProperty(
+      vpio_unit_, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
+      output_bus, &render_callback, sizeof(render_callback));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to specify the render callback on the output bus: "
+                  << result;
+    return false;
+  }
 
   // Disable AU buffer allocation for the recorder, we allocate our own.
   // TODO(henrika): not sure that it actually saves resource to make this call.
   UInt32 flag = 0;
-  LOG_AND_RETURN_IF_ERROR(
-      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
-                           kAudioUnitScope_Output, input_bus, &flag,
-                           sizeof(flag)),
-      "Failed to disable buffer allocation on the input element");
+  result = AudioUnitSetProperty(
+      vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
+      kAudioUnitScope_Output, input_bus, &flag, sizeof(flag));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to disable buffer allocation on the input bus: "
+                  << result;
+  }
 
   // Specify the callback to be called by the I/O thread to us when input audio
   // is available. The recorded samples can then be obtained by calling the
@@ -733,16 +875,39 @@ bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
   AURenderCallbackStruct input_callback;
   input_callback.inputProc = RecordedDataIsAvailable;
   input_callback.inputProcRefCon = this;
-  LOG_AND_RETURN_IF_ERROR(
-      AudioUnitSetProperty(vpio_unit_,
-                           kAudioOutputUnitProperty_SetInputCallback,
-                           kAudioUnitScope_Global, input_bus, &input_callback,
-                           sizeof(input_callback)),
-      "Failed to specify the input callback on the input element");
+  result = AudioUnitSetProperty(vpio_unit_,
+                                kAudioOutputUnitProperty_SetInputCallback,
+                                kAudioUnitScope_Global, input_bus,
+                                &input_callback, sizeof(input_callback));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to specify the input callback on the input bus: "
+                  << result;
+  }
 
   // Initialize the Voice-Processing I/O unit instance.
-  LOG_AND_RETURN_IF_ERROR(AudioUnitInitialize(vpio_unit_),
-                          "Failed to initialize the Voice-Processing I/O unit");
+  // Calls to AudioUnitInitialize() can fail if called back-to-back on
+  // different ADM instances. The error message in this case is -66635 which is
+  // undocumented. Tests have shown that calling AudioUnitInitialize a second
+  // time, after a short sleep, avoids this issue.
+  // See webrtc:5166 for details.
+  int failed_initalize_attempts = 0;
+  result = AudioUnitInitialize(vpio_unit_);
+  while (result != noErr) {
+    LOG(LS_ERROR) << "Failed to initialize the Voice-Processing I/O unit: "
+                  << result;
+    ++failed_initalize_attempts;
+    if (failed_initalize_attempts == kMaxNumberOfAudioUnitInitializeAttempts) {
+      // Max number of initialization attempts exceeded, hence abort.
+      LOG(LS_WARNING) << "Too many initialization attempts";
+      DisposeAudioUnit();
+      return false;
+    }
+    LOG(LS_INFO) << "pause 100ms and try audio unit initialization again...";
+    [NSThread sleepForTimeInterval:0.1f];
+    result = AudioUnitInitialize(vpio_unit_);
+  }
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now initialized";
   return true;
 }
 
@@ -772,18 +937,29 @@ bool AudioDeviceIOS::RestartAudioUnitWithNewFormat(float sample_rate) {
   // Prepare the audio unit to render audio again.
   LOG_AND_RETURN_IF_ERROR(AudioUnitInitialize(vpio_unit_),
                           "Failed to initialize the Voice-Processing I/O unit");
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now reinitialized";
 
   // Start rendering audio using the new format.
   LOG_AND_RETURN_IF_ERROR(AudioOutputUnitStart(vpio_unit_),
                           "Failed to start the Voice-Processing I/O unit");
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now restarted";
   return true;
 }
 
 bool AudioDeviceIOS::InitPlayOrRecord() {
   LOGI() << "InitPlayOrRecord";
+  // Activate the audio session if not already activated.
+  if (!ActivateAudioSession()) {
+    return false;
+  }
+
+  // Ensure that the active audio session has the correct category and mode.
   AVAudioSession* session = [AVAudioSession sharedInstance];
-  // Activate the audio session and ask for a set of preferred audio parameters.
-  ActivateAudioSession(session, true);
+  if (!VerifyAudioSession(session)) {
+    DeactivateAudioSession();
+    LOG(LS_ERROR) << "Failed to verify audio session category and mode";
+    return false;
+  }
 
   // Start observing audio session interruptions and route changes.
   RegisterNotificationObservers();
@@ -793,16 +969,16 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
 
   // Create, setup and initialize a new Voice-Processing I/O unit.
   if (!SetupAndInitializeVoiceProcessingAudioUnit()) {
+    // Reduce usage count for the audio session and possibly deactivate it if
+    // this object is the only user.
+    DeactivateAudioSession();
     return false;
   }
   return true;
 }
 
-bool AudioDeviceIOS::ShutdownPlayOrRecord() {
+void AudioDeviceIOS::ShutdownPlayOrRecord() {
   LOGI() << "ShutdownPlayOrRecord";
-  // Remove audio session notification observers.
-  UnregisterNotificationObservers();
-
   // Close and delete the voice-processing I/O unit.
   OSStatus result = -1;
   if (nullptr != vpio_unit_) {
@@ -814,18 +990,25 @@ bool AudioDeviceIOS::ShutdownPlayOrRecord() {
     if (result != noErr) {
       LOG_F(LS_ERROR) << "AudioUnitUninitialize failed: " << result;
     }
-    result = AudioComponentInstanceDispose(vpio_unit_);
-    if (result != noErr) {
-      LOG_F(LS_ERROR) << "AudioComponentInstanceDispose failed: " << result;
-    }
-    vpio_unit_ = nullptr;
+    DisposeAudioUnit();
   }
 
+  // Remove audio session notification observers.
+  UnregisterNotificationObservers();
+
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
-  AVAudioSession* session = [AVAudioSession sharedInstance];
-  ActivateAudioSession(session, false);
-  return true;
+  DeactivateAudioSession();
+}
+
+void AudioDeviceIOS::DisposeAudioUnit() {
+  if (nullptr == vpio_unit_)
+    return;
+  OSStatus result = AudioComponentInstanceDispose(vpio_unit_);
+  if (result != noErr) {
+    LOG(LS_ERROR) << "AudioComponentInstanceDispose failed:" << result;
+  }
+  vpio_unit_ = nullptr;
 }
 
 OSStatus AudioDeviceIOS::RecordedDataIsAvailable(
@@ -855,8 +1038,11 @@ OSStatus AudioDeviceIOS::OnRecordedDataIsAvailable(
   if (in_number_frames != record_parameters_.frames_per_buffer()) {
     // We have seen short bursts (1-2 frames) where |in_number_frames| changes.
     // Add a log to keep track of longer sequences if that should ever happen.
+    // Also return since calling AudioUnitRender in this state will only result
+    // in kAudio_ParamError (-50) anyhow.
     LOG(LS_WARNING) << "in_number_frames (" << in_number_frames
                     << ") != " << record_parameters_.frames_per_buffer();
+    return noErr;
   }
   // Obtain the recorded audio samples by initiating a rendering cycle.
   // Since it happens on the input bus, the |io_data| parameter is a reference
@@ -866,7 +1052,7 @@ OSStatus AudioDeviceIOS::OnRecordedDataIsAvailable(
   result = AudioUnitRender(vpio_unit_, io_action_flags, in_time_stamp,
                            in_bus_number, in_number_frames, io_data);
   if (result != noErr) {
-    LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
+    LOG_F(LS_ERROR) << "AudioUnitRender failed: " << result;
     return result;
   }
   // Get a pointer to the recorded audio and send it to the WebRTC ADB.
diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
index b892f28b1d..076a67430d 100644
--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
@@ -373,7 +373,7 @@ class MockAudioTransport : public AudioTransport {
                 int32_t(const void* audioSamples,
                         const size_t nSamples,
                         const size_t nBytesPerSample,
-                        const uint8_t nChannels,
+                        const size_t nChannels,
                         const uint32_t samplesPerSec,
                         const uint32_t totalDelayMS,
                         const int32_t clockDrift,
@@ -383,7 +383,7 @@ class MockAudioTransport : public AudioTransport {
   MOCK_METHOD8(NeedMorePlayData,
                int32_t(const size_t nSamples,
                        const size_t nBytesPerSample,
-                       const uint8_t nChannels,
+                       const size_t nChannels,
                        const uint32_t samplesPerSec,
                        void* audioSamples,
                        size_t& nSamplesOut,
@@ -413,7 +413,7 @@ class MockAudioTransport : public AudioTransport {
   int32_t RealRecordedDataIsAvailable(const void* audioSamples,
                                       const size_t nSamples,
                                       const size_t nBytesPerSample,
-                                      const uint8_t nChannels,
+                                      const size_t nChannels,
                                       const uint32_t samplesPerSec,
                                       const uint32_t totalDelayMS,
                                       const int32_t clockDrift,
@@ -428,14 +428,16 @@ class MockAudioTransport : public AudioTransport {
       audio_stream_->Write(audioSamples, nSamples);
     }
     if (ReceivedEnoughCallbacks()) {
-      test_is_done_->Set();
+      if (test_is_done_) {
+        test_is_done_->Set();
+      }
     }
     return 0;
   }
 
   int32_t RealNeedMorePlayData(const size_t nSamples,
                                const size_t nBytesPerSample,
-                               const uint8_t nChannels,
+                               const size_t nChannels,
                                const uint32_t samplesPerSec,
                                void* audioSamples,
                                size_t& nSamplesOut,
@@ -450,7 +452,9 @@ class MockAudioTransport : public AudioTransport {
       audio_stream_->Read(audioSamples, nSamples);
     }
     if (ReceivedEnoughCallbacks()) {
-      test_is_done_->Set();
+      if (test_is_done_) {
+        test_is_done_->Set();
+      }
     }
     return 0;
   }
@@ -636,6 +640,62 @@ TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
   EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
 }
 
+// Verify that we can create two ADMs and start playing on the second ADM.
+// Only the first active instance shall activate an audio session and the
+// last active instance shall deactivate the audio session. The test does not
+// explicitly verify correct audio session calls but instead focuses on
+// ensuring that audio starts for both ADMs.
+TEST_F(AudioDeviceTest, StartPlayoutOnTwoInstances) {
+  // Create and initialize a second/extra ADM instance. The default ADM is
+  // created by the test harness.
+  rtc::scoped_refptr<AudioDeviceModule> second_audio_device =
+      CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
+  EXPECT_NE(second_audio_device.get(), nullptr);
+  EXPECT_EQ(0, second_audio_device->Init());
+
+  // Start playout for the default ADM but don't wait here. Instead use the
+  // upcoming second stream for that. We set the same expectation on number
+  // of callbacks as for the second stream.
+  NiceMock<MockAudioTransport> mock(kPlayout);
+  mock.HandleCallbacks(nullptr, nullptr, 0);
+  EXPECT_CALL(
+      mock, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
+                             playout_channels(), playout_sample_rate(),
+                             NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+
+  // Initialize playout for the second ADM. If all is OK, the second ADM shall
+  // reuse the audio session activated when the first ADM started playing.
+  // This call will also ensure that we avoid a problem related to initializing
+  // two different audio unit instances back to back (see webrtc:5166 for
+  // details).
+  EXPECT_EQ(0, second_audio_device->InitPlayout());
+  EXPECT_TRUE(second_audio_device->PlayoutIsInitialized());
+
+  // Start playout for the second ADM and verify that it starts as intended.
+  // Passing this test ensures that initialization of the second audio unit
+  // has been done successfully and that there is no conflict with the already
+  // playing first ADM.
+  MockAudioTransport mock2(kPlayout);
+  mock2.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
+  EXPECT_CALL(
+      mock2, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
+                              playout_channels(), playout_sample_rate(),
+                              NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, second_audio_device->RegisterAudioCallback(&mock2));
+  EXPECT_EQ(0, second_audio_device->StartPlayout());
+  EXPECT_TRUE(second_audio_device->Playing());
+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
+  EXPECT_EQ(0, second_audio_device->StopPlayout());
+  EXPECT_FALSE(second_audio_device->Playing());
+  EXPECT_FALSE(second_audio_device->PlayoutIsInitialized());
+
+  EXPECT_EQ(0, second_audio_device->Terminate());
+}
+
 // Start playout and verify that the native audio layer starts asking for real
 // audio samples to play out using the NeedMorePlayData callback.
 TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
index 8fa4fdf6f0..bdbccde050 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -207,7 +207,7 @@ int32_t AudioDeviceLinuxALSA::Terminate()
     // RECORDING
     if (_ptrThreadRec)
     {
-        ThreadWrapper* tmpThread = _ptrThreadRec.release();
+        rtc::PlatformThread* tmpThread = _ptrThreadRec.release();
         _critSect.Leave();
 
         tmpThread->Stop();
@@ -219,7 +219,7 @@ int32_t AudioDeviceLinuxALSA::Terminate()
     // PLAYOUT
     if (_ptrThreadPlay)
     {
-        ThreadWrapper* tmpThread = _ptrThreadPlay.release();
+        rtc::PlatformThread* tmpThread = _ptrThreadPlay.release();
         _critSect.Leave();
 
         tmpThread->Stop();
@@ -1363,21 +1363,11 @@ int32_t AudioDeviceLinuxALSA::StartRecording()
         return -1;
     }
     // RECORDING
-    const char* threadName = "webrtc_audio_module_capture_thread";
-    _ptrThreadRec = ThreadWrapper::CreateThread(
-        RecThreadFunc, this, threadName);
+    _ptrThreadRec.reset(new rtc::PlatformThread(
+        RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
 
-    if (!_ptrThreadRec->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the rec audio thread");
-        _recording = false;
-        _ptrThreadRec.reset();
-        delete [] _recordingBuffer;
-        _recordingBuffer = NULL;
-        return -1;
-    }
-    _ptrThreadRec->SetPriority(kRealtimePriority);
+    _ptrThreadRec->Start();
+    _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
     errVal = LATE(snd_pcm_prepare)(_handleRecord);
     if (errVal < 0)
@@ -1517,20 +1507,10 @@ int32_t AudioDeviceLinuxALSA::StartPlayout()
     }
 
     // PLAYOUT
-    const char* threadName = "webrtc_audio_module_play_thread";
-    _ptrThreadPlay =  ThreadWrapper::CreateThread(PlayThreadFunc, this,
-                                                  threadName);
-    if (!_ptrThreadPlay->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the play audio thread");
-        _playing = false;
-        _ptrThreadPlay.reset();
-        delete [] _playoutBuffer;
-        _playoutBuffer = NULL;
-        return -1;
-    }
-    _ptrThreadPlay->SetPriority(kRealtimePriority);
+    _ptrThreadPlay.reset(new rtc::PlatformThread(
+        PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+    _ptrThreadPlay->Start();
+    _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
 
     int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
     if (errVal < 0)
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
index e2391a0456..4a1a5191be 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 #if defined(USE_X11)
 #include <X11/Xlib.h>
@@ -185,8 +185,10 @@ private:
 
     CriticalSectionWrapper& _critSect;
 
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
+    // TODO(pbos): Make plain members and start/stop instead of resetting these
+    // pointers. A thread can be reused.
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
     int32_t _id;
 
diff --git a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
index 929a758e40..42c3ea8295 100644
--- a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -200,33 +200,17 @@ int32_t AudioDeviceLinuxPulse::Init()
     }
 
     // RECORDING
-    const char* threadName = "webrtc_audio_module_rec_thread";
-    _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
-                                                threadName);
-    if (!_ptrThreadRec->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the rec audio thread");
-
-        _ptrThreadRec.reset();
-        return -1;
-    }
+    _ptrThreadRec.reset(new rtc::PlatformThread(
+        RecThreadFunc, this, "webrtc_audio_module_rec_thread"));
 
-    _ptrThreadRec->SetPriority(kRealtimePriority);
+    _ptrThreadRec->Start();
+    _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
     // PLAYOUT
-    threadName = "webrtc_audio_module_play_thread";
-    _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
-                                                 threadName);
-    if (!_ptrThreadPlay->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the play audio thread");
-
-        _ptrThreadPlay.reset();
-        return -1;
-    }
-    _ptrThreadPlay->SetPriority(kRealtimePriority);
+    _ptrThreadPlay.reset(new rtc::PlatformThread(
+        PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+    _ptrThreadPlay->Start();
+    _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
 
     _initialized = true;
 
@@ -246,7 +230,7 @@ int32_t AudioDeviceLinuxPulse::Terminate()
     // RECORDING
     if (_ptrThreadRec)
     {
-        ThreadWrapper* tmpThread = _ptrThreadRec.release();
+        rtc::PlatformThread* tmpThread = _ptrThreadRec.release();
 
         _timeEventRec.Set();
         tmpThread->Stop();
@@ -256,7 +240,7 @@ int32_t AudioDeviceLinuxPulse::Terminate()
     // PLAYOUT
     if (_ptrThreadPlay)
     {
-        ThreadWrapper* tmpThread = _ptrThreadPlay.release();
+        rtc::PlatformThread* tmpThread = _ptrThreadPlay.release();
 
         _timeEventPlay.Set();
         tmpThread->Stop();
diff --git a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
index 718331188d..de8df0be1b 100644
--- a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -11,11 +11,11 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
 
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
-#include "webrtc/base/thread_checker.h"
 
 #include <X11/Xlib.h>
 #include <pulse/pulseaudio.h>
@@ -284,8 +284,9 @@ private:
     EventWrapper& _recStartEvent;
     EventWrapper& _playStartEvent;
 
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
+    // TODO(pbos): Remove scoped_ptr and use directly without resetting.
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
     int32_t _id;
 
     AudioMixerManagerLinuxPulse _mixerManager;
diff --git a/webrtc/modules/audio_device/mac/audio_device_mac.cc b/webrtc/modules/audio_device/mac/audio_device_mac.cc
index db98675bf6..0f33d1124d 100644
--- a/webrtc/modules/audio_device/mac/audio_device_mac.cc
+++ b/webrtc/modules/audio_device/mac/audio_device_mac.cc
@@ -10,3227 +10,2758 @@
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/mac/audio_device_mac.h"
 #include "webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 #include <ApplicationServices/ApplicationServices.h>
-#include <libkern/OSAtomic.h>   // OSAtomicCompareAndSwap()
-#include <mach/mach.h>          // mach_task_self()
-#include <sys/sysctl.h>         // sysctlbyname()
-
-
-
-namespace webrtc
-{
-
-#define WEBRTC_CA_RETURN_ON_ERR(expr)                                   \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,               \
-                "Error in " #expr, (const char *)&err);                 \
-            return -1;                                                  \
-        }                                                               \
-    } while(0)
-
-#define WEBRTC_CA_LOG_ERR(expr)                                         \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,               \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
-
-#define WEBRTC_CA_LOG_WARN(expr)                                        \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceWarning, kTraceAudioDevice, _id,             \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
-
-enum
-{
-    MaxNumberDevices = 64
-};
-
-void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue)
-{
-    while (1)
-    {
-        int32_t oldValue = *theValue;
-        if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue)
-            == true)
-        {
-            return;
-        }
+#include <libkern/OSAtomic.h>  // OSAtomicCompareAndSwap()
+#include <mach/mach.h>         // mach_task_self()
+#include <sys/sysctl.h>        // sysctlbyname()
+
+namespace webrtc {
+
+#define WEBRTC_CA_RETURN_ON_ERR(expr)                                  \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+      return -1;                                                       \
+    }                                                                  \
+  } while (0)
+
+#define WEBRTC_CA_LOG_ERR(expr)                                        \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+    }                                                                  \
+  } while (0)
+
+#define WEBRTC_CA_LOG_WARN(expr)                                         \
+  do {                                                                   \
+    err = expr;                                                          \
+    if (err != noErr) {                                                  \
+      logCAMsg(kTraceWarning, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                     \
+    }                                                                    \
+  } while (0)
+
+enum { MaxNumberDevices = 64 };
+
+void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue) {
+  while (1) {
+    int32_t oldValue = *theValue;
+    if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue) == true) {
+      return;
     }
+  }
 }
 
-int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue)
-{
-    while (1)
-    {
-        int32_t value = *theValue;
-        if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true)
-        {
-            return value;
-        }
+int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue) {
+  while (1) {
+    int32_t value = *theValue;
+    if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true) {
+      return value;
     }
+  }
 }
 
 // CoreAudio errors are best interpreted as four character strings.
 void AudioDeviceMac::logCAMsg(const TraceLevel level,
                               const TraceModule module,
-                              const int32_t id, const char *msg,
-                              const char *err)
-{
+                              const int32_t id,
+                              const char* msg,
+                              const char* err) {
   RTC_DCHECK(msg != NULL);
   RTC_DCHECK(err != NULL);
 
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
+  WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
 #else
-    // We need to flip the characters in this case.
-    WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
-        + 2, err + 1, err);
+  // We need to flip the characters in this case.
+  WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err + 2,
+               err + 1, err);
 #endif
 }
 
-AudioDeviceMac::AudioDeviceMac(const int32_t id) :
-    _ptrAudioBuffer(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _stopEventRec(*EventWrapper::Create()),
-    _stopEvent(*EventWrapper::Create()),
-    _id(id),
-    _mixerManager(id),
-    _inputDeviceIndex(0),
-    _outputDeviceIndex(0),
-    _inputDeviceID(kAudioObjectUnknown),
-    _outputDeviceID(kAudioObjectUnknown),
-    _inputDeviceIsSpecified(false),
-    _outputDeviceIsSpecified(false),
-    _recChannels(N_REC_CHANNELS),
-    _playChannels(N_PLAY_CHANNELS),
-    _captureBufData(NULL),
-    _renderBufData(NULL),
-    _playBufType(AudioDeviceModule::kFixedBufferSize),
-    _initialized(false),
-    _isShutDown(false),
-    _recording(false),
-    _playing(false),
-    _recIsInitialized(false),
-    _playIsInitialized(false),
-    _AGC(false),
-    _renderDeviceIsAlive(1),
-    _captureDeviceIsAlive(1),
-    _twoDevices(true),
-    _doStop(false),
-    _doStopRec(false),
-    _macBookPro(false),
-    _macBookProPanRight(false),
-    _captureLatencyUs(0),
-    _renderLatencyUs(0),
-    _captureDelayUs(0),
-    _renderDelayUs(0),
-    _renderDelayOffsetSamples(0),
-    _playBufDelayFixed(20),
-    _playWarning(0),
-    _playError(0),
-    _recWarning(0),
-    _recError(0),
-    _paCaptureBuffer(NULL),
-    _paRenderBuffer(NULL),
-    _captureBufSizeSamples(0),
-    _renderBufSizeSamples(0),
-    prev_key_state_()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
-
-    RTC_DCHECK(&_stopEvent != NULL);
-    RTC_DCHECK(&_stopEventRec != NULL);
-
-    memset(_renderConvertData, 0, sizeof(_renderConvertData));
-    memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
-}
-
-
-AudioDeviceMac::~AudioDeviceMac()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
-
-    if (!_isShutDown)
-    {
-        Terminate();
-    }
+AudioDeviceMac::AudioDeviceMac(const int32_t id)
+    : _ptrAudioBuffer(NULL),
+      _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      _stopEventRec(*EventWrapper::Create()),
+      _stopEvent(*EventWrapper::Create()),
+      _id(id),
+      _mixerManager(id),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceID(kAudioObjectUnknown),
+      _outputDeviceID(kAudioObjectUnknown),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      _recChannels(N_REC_CHANNELS),
+      _playChannels(N_PLAY_CHANNELS),
+      _captureBufData(NULL),
+      _renderBufData(NULL),
+      _playBufType(AudioDeviceModule::kFixedBufferSize),
+      _initialized(false),
+      _isShutDown(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _AGC(false),
+      _renderDeviceIsAlive(1),
+      _captureDeviceIsAlive(1),
+      _twoDevices(true),
+      _doStop(false),
+      _doStopRec(false),
+      _macBookPro(false),
+      _macBookProPanRight(false),
+      _captureLatencyUs(0),
+      _renderLatencyUs(0),
+      _captureDelayUs(0),
+      _renderDelayUs(0),
+      _renderDelayOffsetSamples(0),
+      _playBufDelayFixed(20),
+      _playWarning(0),
+      _playError(0),
+      _recWarning(0),
+      _recError(0),
+      _paCaptureBuffer(NULL),
+      _paRenderBuffer(NULL),
+      _captureBufSizeSamples(0),
+      _renderBufSizeSamples(0),
+      prev_key_state_(),
+      get_mic_volume_counter_ms_(0) {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+
+  RTC_DCHECK(&_stopEvent != NULL);
+  RTC_DCHECK(&_stopEventRec != NULL);
+
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
+  memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+}
+
+AudioDeviceMac::~AudioDeviceMac() {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed",
+               __FUNCTION__);
+
+  if (!_isShutDown) {
+    Terminate();
+  }
 
-    RTC_DCHECK(!capture_worker_thread_.get());
-    RTC_DCHECK(!render_worker_thread_.get());
+  RTC_DCHECK(!capture_worker_thread_.get());
+  RTC_DCHECK(!render_worker_thread_.get());
 
-    if (_paRenderBuffer)
-    {
-        delete _paRenderBuffer;
-        _paRenderBuffer = NULL;
-    }
+  if (_paRenderBuffer) {
+    delete _paRenderBuffer;
+    _paRenderBuffer = NULL;
+  }
 
-    if (_paCaptureBuffer)
-    {
-        delete _paCaptureBuffer;
-        _paCaptureBuffer = NULL;
-    }
+  if (_paCaptureBuffer) {
+    delete _paCaptureBuffer;
+    _paCaptureBuffer = NULL;
+  }
 
-    if (_renderBufData)
-    {
-        delete[] _renderBufData;
-        _renderBufData = NULL;
-    }
+  if (_renderBufData) {
+    delete[] _renderBufData;
+    _renderBufData = NULL;
+  }
 
-    if (_captureBufData)
-    {
-        delete[] _captureBufData;
-        _captureBufData = NULL;
-    }
+  if (_captureBufData) {
+    delete[] _captureBufData;
+    _captureBufData = NULL;
+  }
 
-    kern_return_t kernErr = KERN_SUCCESS;
-    kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_destroy() error: %d", kernErr);
-    }
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_destroy() error: %d", kernErr);
+  }
 
-    kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_destroy() error: %d", kernErr);
-    }
+  kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_destroy() error: %d", kernErr);
+  }
 
-    delete &_stopEvent;
-    delete &_stopEventRec;
-    delete &_critSect;
+  delete &_stopEvent;
+  delete &_stopEventRec;
+  delete &_critSect;
 }
 
 // ============================================================================
 //                                     API
 // ============================================================================
 
-void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
+void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  _ptrAudioBuffer = audioBuffer;
 
-    _ptrAudioBuffer = audioBuffer;
-
-    // inform the AudioBuffer about default settings for this implementation
-    _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-    _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
-    _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
-    _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+  // inform the AudioBuffer about default settings for this implementation
+  _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+  _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
 }
 
 int32_t AudioDeviceMac::ActiveAudioLayer(
-    AudioDeviceModule::AudioLayer& audioLayer) const
-{
-    audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
-    return 0;
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+  return 0;
 }
 
-int32_t AudioDeviceMac::Init()
-{
+int32_t AudioDeviceMac::Init() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  if (_initialized) {
+    return 0;
+  }
 
-    if (_initialized)
-    {
-        return 0;
-    }
+  OSStatus err = noErr;
 
-    OSStatus err = noErr;
+  _isShutDown = false;
 
-    _isShutDown = false;
-
-    // PortAudio ring buffers require an elementCount which is a power of two.
-    if (_renderBufData == NULL)
-    {
-        UInt32 powerOfTwo = 1;
-        while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES)
-        {
-            powerOfTwo <<= 1;
-        }
-        _renderBufSizeSamples = powerOfTwo;
-        _renderBufData = new SInt16[_renderBufSizeSamples];
+  // PortAudio ring buffers require an elementCount which is a power of two.
+  if (_renderBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
     }
+    _renderBufSizeSamples = powerOfTwo;
+    _renderBufData = new SInt16[_renderBufSizeSamples];
+  }
 
-    if (_paRenderBuffer == NULL)
-    {
-        _paRenderBuffer = new PaUtilRingBuffer;
-        PaRingBufferSize bufSize = -1;
-        bufSize = PaUtil_InitializeRingBuffer(_paRenderBuffer, sizeof(SInt16),
-                                              _renderBufSizeSamples,
-                                              _renderBufData);
-        if (bufSize == -1)
-        {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
-                         _id, " PaUtil_InitializeRingBuffer() error");
-            return -1;
-        }
+  if (_paRenderBuffer == NULL) {
+    _paRenderBuffer = new PaUtilRingBuffer;
+    PaRingBufferSize bufSize = -1;
+    bufSize = PaUtil_InitializeRingBuffer(
+        _paRenderBuffer, sizeof(SInt16), _renderBufSizeSamples, _renderBufData);
+    if (bufSize == -1) {
+      WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                   " PaUtil_InitializeRingBuffer() error");
+      return -1;
     }
+  }
 
-    if (_captureBufData == NULL)
-    {
-        UInt32 powerOfTwo = 1;
-        while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES)
-        {
-            powerOfTwo <<= 1;
-        }
-        _captureBufSizeSamples = powerOfTwo;
-        _captureBufData = new Float32[_captureBufSizeSamples];
+  if (_captureBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
     }
+    _captureBufSizeSamples = powerOfTwo;
+    _captureBufData = new Float32[_captureBufSizeSamples];
+  }
 
-    if (_paCaptureBuffer == NULL)
-    {
-        _paCaptureBuffer = new PaUtilRingBuffer;
-        PaRingBufferSize bufSize = -1;
-        bufSize = PaUtil_InitializeRingBuffer(_paCaptureBuffer,
-                                              sizeof(Float32),
-                                              _captureBufSizeSamples,
-                                              _captureBufData);
-        if (bufSize == -1)
-        {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
-                         _id, " PaUtil_InitializeRingBuffer() error");
-            return -1;
-        }
+  if (_paCaptureBuffer == NULL) {
+    _paCaptureBuffer = new PaUtilRingBuffer;
+    PaRingBufferSize bufSize = -1;
+    bufSize =
+        PaUtil_InitializeRingBuffer(_paCaptureBuffer, sizeof(Float32),
+                                    _captureBufSizeSamples, _captureBufData);
+    if (bufSize == -1) {
+      WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                   " PaUtil_InitializeRingBuffer() error");
+      return -1;
     }
+  }
 
-    kern_return_t kernErr = KERN_SUCCESS;
-    kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
-                               SYNC_POLICY_FIFO, 0);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     " semaphore_create() error: %d", kernErr);
-        return -1;
-    }
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                 " semaphore_create() error: %d", kernErr);
+    return -1;
+  }
 
-    kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
-                               SYNC_POLICY_FIFO, 0);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     " semaphore_create() error: %d", kernErr);
-        return -1;
-    }
+  kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                 " semaphore_create() error: %d", kernErr);
+    return -1;
+  }
 
-    // Setting RunLoop to NULL here instructs HAL to manage its own thread for
-    // notifications. This was the default behaviour on OS X 10.5 and earlier,
-    // but now must be explicitly specified. HAL would otherwise try to use the
-    // main thread to issue notifications.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyRunLoop,
-            kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    CFRunLoopRef runLoop = NULL;
-    UInt32 size = sizeof(CFRunLoopRef);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, size, &runLoop));
-
-    // Listen for any device changes.
-    propertyAddress.mSelector = kAudioHardwarePropertyDevices;
-    WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(kAudioObjectSystemObject,
-            &propertyAddress, &objectListenerProc, this));
-
-    // Determine if this is a MacBook Pro
-    _macBookPro = false;
-    _macBookProPanRight = false;
-    char buf[128];
-    size_t length = sizeof(buf);
-    memset(buf, 0, length);
-
-    int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
-    if (intErr != 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Error in sysctlbyname(): %d", err);
-    } else
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Hardware model: %s", buf);
-        if (strncmp(buf, "MacBookPro", 10) == 0)
-        {
-            _macBookPro = true;
-        }
+  // Setting RunLoop to NULL here instructs HAL to manage its own thread for
+  // notifications. This was the default behaviour on OS X 10.5 and earlier,
+  // but now must be explicitly specified. HAL would otherwise try to use the
+  // main thread to issue notifications.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyRunLoop, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  CFRunLoopRef runLoop = NULL;
+  UInt32 size = sizeof(CFRunLoopRef);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, size, &runLoop));
+
+  // Listen for any device changes.
+  propertyAddress.mSelector = kAudioHardwarePropertyDevices;
+  WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
+
+  // Determine if this is a MacBook Pro
+  _macBookPro = false;
+  _macBookProPanRight = false;
+  char buf[128];
+  size_t length = sizeof(buf);
+  memset(buf, 0, length);
+
+  int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
+  if (intErr != 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Error in sysctlbyname(): %d", err);
+  } else {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Hardware model: %s",
+                 buf);
+    if (strncmp(buf, "MacBookPro", 10) == 0) {
+      _macBookPro = true;
     }
+  }
 
-    _playWarning = 0;
-    _playError = 0;
-    _recWarning = 0;
-    _recError = 0;
+  _playWarning = 0;
+  _playError = 0;
+  _recWarning = 0;
+  _recError = 0;
 
-    _initialized = true;
+  get_mic_volume_counter_ms_ = 0;
 
-    return 0;
+  _initialized = true;
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::Terminate()
-{
+int32_t AudioDeviceMac::Terminate() {
+  if (!_initialized) {
+    return 0;
+  }
 
-    if (!_initialized)
-    {
-        return 0;
-    }
+  if (_recording) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Recording must be stopped");
+    return -1;
+  }
 
-    if (_recording)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Recording must be stopped");
-        return -1;
-    }
+  if (_playing) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Playback must be stopped");
+    return -1;
+  }
 
-    if (_playing)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Playback must be stopped");
-        return -1;
-    }
+  _critSect.Enter();
 
-    _critSect.Enter();
+  _mixerManager.Close();
 
-    _mixerManager.Close();
+  OSStatus err = noErr;
+  int retVal = 0;
 
-    OSStatus err = noErr;
-    int retVal = 0;
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
 
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
-            &propertyAddress, &objectListenerProc, this));
-
-    err = AudioHardwareUnload();
-    if (err != noErr)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Error in AudioHardwareUnload()", (const char*) &err);
-        retVal = -1;
-    }
+  err = AudioHardwareUnload();
+  if (err != noErr) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Error in AudioHardwareUnload()", (const char*)&err);
+    retVal = -1;
+  }
 
-    _isShutDown = true;
-    _initialized = false;
-    _outputDeviceIsSpecified = false;
-    _inputDeviceIsSpecified = false;
+  _isShutDown = true;
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
 
-    _critSect.Leave();
+  _critSect.Leave();
 
-    return retVal;
+  return retVal;
 }
 
-bool AudioDeviceMac::Initialized() const
-{
-    return (_initialized);
+bool AudioDeviceMac::Initialized() const {
+  return (_initialized);
 }
 
-int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        available = false;
-        return 0;
-    }
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    available = false;
+    return 0;
+  }
 
-    // Given that InitSpeaker was successful, we know that a valid speaker
-    // exists.
-    available = true;
+  // Given that InitSpeaker was successful, we know that a valid speaker
+  // exists.
+  available = true;
 
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::InitSpeaker()
-{
+int32_t AudioDeviceMac::InitSpeaker() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_playing)
-    {
-        return -1;
-    }
+  if (_playing) {
+    return -1;
+  }
 
-    if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1)
-    {
-        return -1;
-    }
+  if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1) {
+    return -1;
+  }
 
-    if (_inputDeviceID == _outputDeviceID)
-    {
-        _twoDevices = false;
-    } else
-    {
-        _twoDevices = true;
-    }
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
 
-    if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1) {
+    return -1;
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available)
-{
+int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        available = false;
-        return 0;
-    }
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    available = false;
+    return 0;
+  }
 
-    // Given that InitMicrophone was successful, we know that a valid microphone
-    // exists.
-    available = true;
+  // Given that InitMicrophone was successful, we know that a valid microphone
+  // exists.
+  available = true;
 
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::InitMicrophone()
-{
-
-    CriticalSectionScoped lock(&_critSect);
+int32_t AudioDeviceMac::InitMicrophone() {
+  CriticalSectionScoped lock(&_critSect);
 
-    if (_recording)
-    {
-        return -1;
-    }
+  if (_recording) {
+    return -1;
+  }
 
-    if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1)
-    {
-        return -1;
-    }
+  if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1) {
+    return -1;
+  }
 
-    if (_inputDeviceID == _outputDeviceID)
-    {
-        _twoDevices = false;
-    } else
-    {
-        _twoDevices = true;
-    }
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
 
-    if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1) {
+    return -1;
+  }
 
-    return 0;
+  return 0;
 }
 
-bool AudioDeviceMac::SpeakerIsInitialized() const
-{
-    return (_mixerManager.SpeakerIsInitialized());
+bool AudioDeviceMac::SpeakerIsInitialized() const {
+  return (_mixerManager.SpeakerIsInitialized());
 }
 
-bool AudioDeviceMac::MicrophoneIsInitialized() const
-{
-    return (_mixerManager.MicrophoneIsInitialized());
+bool AudioDeviceMac::MicrophoneIsInitialized() const {
+  return (_mixerManager.MicrophoneIsInitialized());
 }
 
-int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available)
-{
+int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // If we end up here it means that the selected speaker has no volume
-        // control.
-        available = false;
-        return 0;
-    }
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
+    return 0;
+  }
 
-    // Given that InitSpeaker was successful, we know that a volume control exists
-    //
-    available = true;
+  // Given that InitSpeaker was successful, we know that a volume control exists
+  //
+  available = true;
 
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume)
-{
-
-    return (_mixerManager.SetSpeakerVolume(volume));
+int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume) {
+  return (_mixerManager.SetSpeakerVolume(volume));
 }
 
-int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const
-{
+int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const {
+  uint32_t level(0);
 
-    uint32_t level(0);
-
-    if (_mixerManager.SpeakerVolume(level) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
 
-    volume = level;
-    return 0;
+  volume = level;
+  return 0;
 }
 
 int32_t AudioDeviceMac::SetWaveOutVolume(uint16_t volumeLeft,
-                                         uint16_t volumeRight)
-{
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
+                                         uint16_t volumeRight) {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
+  return -1;
 }
 
-int32_t
-AudioDeviceMac::WaveOutVolume(uint16_t& /*volumeLeft*/,
-                              uint16_t& /*volumeRight*/) const
-{
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
+int32_t AudioDeviceMac::WaveOutVolume(uint16_t& /*volumeLeft*/,
+                                      uint16_t& /*volumeRight*/) const {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
+  return -1;
 }
 
-int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
+int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
 
-    uint32_t maxVol(0);
-
-    if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
 
-    maxVolume = maxVol;
-    return 0;
+  maxVolume = maxVol;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const
-{
-
-    uint32_t minVol(0);
+int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
 
-    if (_mixerManager.MinSpeakerVolume(minVol) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
 
-    minVolume = minVol;
-    return 0;
+  minVolume = minVol;
+  return 0;
 }
 
-int32_t
-AudioDeviceMac::SpeakerVolumeStepSize(uint16_t& stepSize) const
-{
-
-    uint16_t delta(0);
+int32_t AudioDeviceMac::SpeakerVolumeStepSize(uint16_t& stepSize) const {
+  uint16_t delta(0);
 
-    if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.SpeakerVolumeStepSize(delta) == -1) {
+    return -1;
+  }
 
-    stepSize = delta;
-    return 0;
+  stepSize = delta;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // If we end up here it means that the selected speaker has no volume
-        // control, hence it is safe to state that there is no mute control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
 
-    // Check if the selected speaker has a mute control
-    //
-    _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+  // Check if the selected speaker has a mute control
+  //
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
 
-    available = isAvailable;
+  available = isAvailable;
 
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetSpeakerMute(bool enable)
-{
-    return (_mixerManager.SetSpeakerMute(enable));
+int32_t AudioDeviceMac::SetSpeakerMute(bool enable) {
+  return (_mixerManager.SetSpeakerMute(enable));
 }
 
-int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const
-{
+int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const {
+  bool muted(0);
 
-    bool muted(0);
-
-    if (_mixerManager.SpeakerMute(muted) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
 
-    enabled = muted;
-    return 0;
+  enabled = muted;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available)
-{
+int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected input device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no boost control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
 
-    // Check if the selected microphone has a mute control
-    //
-    _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
-    available = isAvailable;
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
 
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetMicrophoneMute(bool enable)
-{
-    return (_mixerManager.SetMicrophoneMute(enable));
+int32_t AudioDeviceMac::SetMicrophoneMute(bool enable) {
+  return (_mixerManager.SetMicrophoneMute(enable));
 }
 
-int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const
-{
+int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const {
+  bool muted(0);
 
-    bool muted(0);
-
-    if (_mixerManager.MicrophoneMute(muted) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
 
-    enabled = muted;
-    return 0;
+  enabled = muted;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+int32_t AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
-    // Enumerate all avaliable microphone and make an attempt to open up the
-    // input mixer corresponding to the currently selected input device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
+  // Enumerate all avaliable microphone and make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no boost control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
 
-    // Check if the selected microphone has a boost control
-    //
-    _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
-    available = isAvailable;
+  // Check if the selected microphone has a boost control
+  //
+  _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
+  available = isAvailable;
 
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetMicrophoneBoost(bool enable)
-{
-
-    return (_mixerManager.SetMicrophoneBoost(enable));
+int32_t AudioDeviceMac::SetMicrophoneBoost(bool enable) {
+  return (_mixerManager.SetMicrophoneBoost(enable));
 }
 
-int32_t AudioDeviceMac::MicrophoneBoost(bool& enabled) const
-{
-
-    bool onOff(0);
+int32_t AudioDeviceMac::MicrophoneBoost(bool& enabled) const {
+  bool onOff(0);
 
-    if (_mixerManager.MicrophoneBoost(onOff) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MicrophoneBoost(onOff) == -1) {
+    return -1;
+  }
 
-    enabled = onOff;
-    return 0;
+  enabled = onOff;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // Cannot open the specified device
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone can record stereo
-    //
-    _mixerManager.StereoRecordingIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
+int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // Cannot open the specified device
+    available = false;
     return 0;
-}
+  }
 
-int32_t AudioDeviceMac::SetStereoRecording(bool enable)
-{
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoRecordingIsAvailable(isAvailable);
+  available = isAvailable;
 
-    if (enable)
-        _recChannels = 2;
-    else
-        _recChannels = 1;
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StereoRecording(bool& enabled) const
-{
+int32_t AudioDeviceMac::SetStereoRecording(bool enable) {
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
 
-    if (_recChannels == 2)
-        enabled = true;
-    else
-        enabled = false;
-
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+int32_t AudioDeviceMac::StereoRecording(bool& enabled) const {
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
 
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // Cannot open the specified device
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone can record stereo
-    //
-    _mixerManager.StereoPlayoutIsAvailable(isAvailable);
-    available = isAvailable;
+  return 0;
+}
 
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
+int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // Cannot open the specified device
+    available = false;
     return 0;
-}
+  }
 
-int32_t AudioDeviceMac::SetStereoPlayout(bool enable)
-{
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+  available = isAvailable;
 
-    if (enable)
-        _playChannels = 2;
-    else
-        _playChannels = 1;
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const
-{
-
-    if (_playChannels == 2)
-        enabled = true;
-    else
-        enabled = false;
+int32_t AudioDeviceMac::SetStereoPlayout(bool enable) {
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetAGC(bool enable)
-{
-
-    _AGC = enable;
+int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const {
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
 
-    return 0;
+  return 0;
 }
 
-bool AudioDeviceMac::AGC() const
-{
+int32_t AudioDeviceMac::SetAGC(bool enable) {
+  _AGC = enable;
 
-    return _AGC;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available)
-{
+bool AudioDeviceMac::AGC() const {
+  return _AGC;
+}
 
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control.
-        available = false;
-        return 0;
-    }
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control.
+    available = false;
+    return 0;
+  }
 
-    // Given that InitMicrophone was successful, we know that a volume control
-    // exists
-    //
-    available = true;
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists
+  //
+  available = true;
 
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume)
-{
-
-    return (_mixerManager.SetMicrophoneVolume(volume));
+int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
 }
 
-int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const
-{
+int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
 
-    uint32_t level(0);
-
-    if (_mixerManager.MicrophoneVolume(level) == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  failed to retrive current microphone level");
-        return -1;
-    }
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  failed to retrive current microphone level");
+    return -1;
+  }
 
-    volume = level;
-    return 0;
+  volume = level;
+  return 0;
 }
 
-int32_t
-AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
-
-    uint32_t maxVol(0);
+int32_t AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
 
-    if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
 
-    maxVolume = maxVol;
-    return 0;
+  maxVolume = maxVol;
+  return 0;
 }
 
-int32_t
-AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const
-{
-
-    uint32_t minVol(0);
+int32_t AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
 
-    if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
 
-    minVolume = minVol;
-    return 0;
+  minVolume = minVol;
+  return 0;
 }
 
-int32_t
-AudioDeviceMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
-{
-
-    uint16_t delta(0);
+int32_t AudioDeviceMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const {
+  uint16_t delta(0);
 
-    if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1) {
+    return -1;
+  }
 
-    stepSize = delta;
-    return 0;
+  stepSize = delta;
+  return 0;
 }
 
-int16_t AudioDeviceMac::PlayoutDevices()
-{
-
-    AudioDeviceID playDevices[MaxNumberDevices];
-    return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
-                            MaxNumberDevices);
+int16_t AudioDeviceMac::PlayoutDevices() {
+  AudioDeviceID playDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
+                          MaxNumberDevices);
 }
 
-int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index)
-{
-    CriticalSectionScoped lock(&_critSect);
+int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index) {
+  CriticalSectionScoped lock(&_critSect);
 
-    if (_playIsInitialized)
-    {
-        return -1;
-    }
+  if (_playIsInitialized) {
+    return -1;
+  }
 
-    AudioDeviceID playDevices[MaxNumberDevices];
-    uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
-                                         playDevices, MaxNumberDevices);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  number of availiable waveform-audio output devices is %u",
-                 nDevices);
-
-    if (index > (nDevices - 1))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  device index is out of range [0,%u]", (nDevices - 1));
-        return -1;
-    }
+  AudioDeviceID playDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
+                                       playDevices, MaxNumberDevices);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  number of availiable waveform-audio output devices is %u",
+               nDevices);
 
-    _outputDeviceIndex = index;
-    _outputDeviceIsSpecified = true;
+  if (index > (nDevices - 1)) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  device index is out of range [0,%u]", (nDevices - 1));
+    return -1;
+  }
 
-    return 0;
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
+
+  return 0;
 }
 
 int32_t AudioDeviceMac::SetPlayoutDevice(
-    AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+               "WindowsDeviceType not supported");
+  return -1;
 }
 
-int32_t AudioDeviceMac::PlayoutDeviceName(
-    uint16_t index,
-    char name[kAdmMaxDeviceNameSize],
-    char guid[kAdmMaxGuidSize])
-{
+int32_t AudioDeviceMac::PlayoutDeviceName(uint16_t index,
+                                          char name[kAdmMaxDeviceNameSize],
+                                          char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(PlayoutDevices());
 
-    const uint16_t nDevices(PlayoutDevices());
-
-    if ((index > (nDevices - 1)) || (name == NULL))
-    {
-        return -1;
-    }
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
 
-    memset(name, 0, kAdmMaxDeviceNameSize);
+  memset(name, 0, kAdmMaxDeviceNameSize);
 
-    if (guid != NULL)
-    {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
 
-    return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
+  return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
 }
 
-int32_t AudioDeviceMac::RecordingDeviceName(
-    uint16_t index,
-    char name[kAdmMaxDeviceNameSize],
-    char guid[kAdmMaxGuidSize])
-{
-
-    const uint16_t nDevices(RecordingDevices());
+int32_t AudioDeviceMac::RecordingDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(RecordingDevices());
 
-    if ((index > (nDevices - 1)) || (name == NULL))
-    {
-        return -1;
-    }
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
 
-    memset(name, 0, kAdmMaxDeviceNameSize);
+  memset(name, 0, kAdmMaxDeviceNameSize);
 
-    if (guid != NULL)
-    {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
 
-    return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
+  return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
 }
 
-int16_t AudioDeviceMac::RecordingDevices()
-{
-
-    AudioDeviceID recDevices[MaxNumberDevices];
-    return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
-                            MaxNumberDevices);
+int16_t AudioDeviceMac::RecordingDevices() {
+  AudioDeviceID recDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
+                          MaxNumberDevices);
 }
 
-int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index)
-{
+int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index) {
+  if (_recIsInitialized) {
+    return -1;
+  }
 
-    if (_recIsInitialized)
-    {
-        return -1;
-    }
+  AudioDeviceID recDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
+                                       recDevices, MaxNumberDevices);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  number of availiable waveform-audio input devices is %u",
+               nDevices);
 
-    AudioDeviceID recDevices[MaxNumberDevices];
-    uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
-                                         recDevices, MaxNumberDevices);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  number of availiable waveform-audio input devices is %u",
-                 nDevices);
-
-    if (index > (nDevices - 1))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  device index is out of range [0,%u]", (nDevices - 1));
-        return -1;
-    }
+  if (index > (nDevices - 1)) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  device index is out of range [0,%u]", (nDevices - 1));
+    return -1;
+  }
 
-    _inputDeviceIndex = index;
-    _inputDeviceIsSpecified = true;
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
 
-    return 0;
+  return 0;
 }
 
-
-int32_t
-AudioDeviceMac::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
+int32_t AudioDeviceMac::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+               "WindowsDeviceType not supported");
+  return -1;
 }
 
-int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available)
-{
-
-    available = true;
+int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available) {
+  available = true;
 
-    // Try to initialize the playout side
-    if (InitPlayout() == -1)
-    {
-        available = false;
-    }
+  // Try to initialize the playout side
+  if (InitPlayout() == -1) {
+    available = false;
+  }
 
-    // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
-    // We must actually start playout here in order to have the IOProc
-    // deleted by calling StopPlayout().
-    if (StartPlayout() == -1)
-    {
-        available = false;
-    }
+  // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
+  // We must actually start playout here in order to have the IOProc
+  // deleted by calling StopPlayout().
+  if (StartPlayout() == -1) {
+    available = false;
+  }
 
-    // Cancel effect of initialization
-    if (StopPlayout() == -1)
-    {
-        available = false;
-    }
+  // Cancel effect of initialization
+  if (StopPlayout() == -1) {
+    available = false;
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::RecordingIsAvailable(bool& available)
-{
-
-    available = true;
+int32_t AudioDeviceMac::RecordingIsAvailable(bool& available) {
+  available = true;
 
-    // Try to initialize the recording side
-    if (InitRecording() == -1)
-    {
-        available = false;
-    }
+  // Try to initialize the recording side
+  if (InitRecording() == -1) {
+    available = false;
+  }
 
-    // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
-    // We must actually start recording here in order to have the IOProc
-    // deleted by calling StopRecording().
-    if (StartRecording() == -1)
-    {
-        available = false;
-    }
+  // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
+  // We must actually start recording here in order to have the IOProc
+  // deleted by calling StopRecording().
+  if (StartRecording() == -1) {
+    available = false;
+  }
 
-    // Cancel effect of initialization
-    if (StopRecording() == -1)
-    {
-        available = false;
-    }
+  // Cancel effect of initialization
+  if (StopRecording() == -1) {
+    available = false;
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::InitPlayout()
-{
-    CriticalSectionScoped lock(&_critSect);
+int32_t AudioDeviceMac::InitPlayout() {
+  CriticalSectionScoped lock(&_critSect);
 
-    if (_playing)
-    {
-        return -1;
-    }
+  if (_playing) {
+    return -1;
+  }
 
-    if (!_outputDeviceIsSpecified)
-    {
-        return -1;
-    }
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
 
-    if (_playIsInitialized)
-    {
-        return 0;
-    }
+  if (_playIsInitialized) {
+    return 0;
+  }
 
-    // Initialize the speaker (devices might have been added or removed)
-    if (InitSpeaker() == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitSpeaker() failed");
-    }
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeaker() == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  InitSpeaker() failed");
+  }
 
-    if (!MicrophoneIsInitialized())
-    {
-        // Make this call to check if we are using
-        // one or two devices (_twoDevices)
-        bool available = false;
-        if (MicrophoneIsAvailable(available) == -1)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  MicrophoneIsAvailable() failed");
-        }
+  if (!MicrophoneIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (MicrophoneIsAvailable(available) == -1) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "  MicrophoneIsAvailable() failed");
     }
+  }
 
-    PaUtil_FlushRingBuffer(_paRenderBuffer);
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    _renderDelayOffsetSamples = 0;
-    _renderDelayUs = 0;
-    _renderLatencyUs = 0;
-    _renderDeviceIsAlive = 1;
-    _doStop = false;
+  PaUtil_FlushRingBuffer(_paRenderBuffer);
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  _renderDelayOffsetSamples = 0;
+  _renderDelayUs = 0;
+  _renderLatencyUs = 0;
+  _renderDeviceIsAlive = 1;
+  _doStop = false;
+
+  // The internal microphone of a MacBook Pro is located under the left speaker
+  // grille. When the internal speakers are in use, we want to fully stereo
+  // pan to the right.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  if (_macBookPro) {
+    _macBookProPanRight = false;
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      UInt32 dataSource = 0;
+      size = sizeof(dataSource);
+      WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, &size, &dataSource));
+
+      if (dataSource == 'ispk') {
+        _macBookProPanRight = true;
+        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                     "MacBook Pro using internal speakers; stereo"
+                     " panning right");
+      } else {
+        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                     "MacBook Pro not using internal speakers");
+      }
 
-    // The internal microphone of a MacBook Pro is located under the left speaker
-    // grille. When the internal speakers are in use, we want to fully stereo
-    // pan to the right.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyDataSource,
-                kAudioDevicePropertyScopeOutput, 0 };
-    if (_macBookPro)
-    {
-        _macBookProPanRight = false;
-        Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                     &propertyAddress);
-        if (hasProperty)
-        {
-            UInt32 dataSource = 0;
-            size = sizeof(dataSource);
-            WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(_outputDeviceID,
-                    &propertyAddress, 0, NULL, &size, &dataSource));
-
-            if (dataSource == 'ispk')
-            {
-                _macBookProPanRight = true;
-                WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
-                             _id,
-                             "MacBook Pro using internal speakers; stereo"
-                             " panning right");
-            } else
-            {
-                WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
-                             _id, "MacBook Pro not using internal speakers");
-            }
-
-            // Add a listener to determine if the status changes.
-            WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
-                    &propertyAddress, &objectListenerProc, this));
-        }
+      // Add a listener to determine if the status changes.
+      WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
     }
+  }
 
-    // Get current stream description
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
-    size = sizeof(_outStreamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-            &propertyAddress, 0, NULL, &size, &_outStreamFormat));
-
-    if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable output stream format -> mFormatID",
-                 (const char *) &_outStreamFormat.mFormatID);
-        return -1;
-    }
+  // Get current stream description
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
+  size = sizeof(_outStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &_outStreamFormat));
+
+  if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable output stream format -> mFormatID",
+             (const char*)&_outStreamFormat.mFormatID);
+    return -1;
+  }
 
-    if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Too many channels on output device (mChannelsPerFrame = %d)",
-            _outStreamFormat.mChannelsPerFrame);
-        return -1;
-    }
+  if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on output device (mChannelsPerFrame = %d)",
+                 _outStreamFormat.mChannelsPerFrame);
+    return -1;
+  }
 
-    if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Non-interleaved audio data is not supported.",
-                     "AudioHardware streams should not have this format.");
-        return -1;
-    }
+  if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Non-interleaved audio data is not supported.",
+                 "AudioHardware streams should not have this format.");
+    return -1;
+  }
 
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Ouput stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mSampleRate = %f, mChannelsPerFrame = %u",
+               _outStreamFormat.mSampleRate,
+               _outStreamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerPacket = %u, mFramesPerPacket = %u",
+               _outStreamFormat.mBytesPerPacket,
+               _outStreamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerFrame = %u, mBitsPerChannel = %u",
+               _outStreamFormat.mBytesPerFrame,
+               _outStreamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "mFormatFlags = %u",
+               _outStreamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&_outStreamFormat.mFormatID);
+
+  // Our preferred format to work with.
+  if (_outStreamFormat.mChannelsPerFrame < 2) {
+    // Disable stereo playout when we only have one channel on the device.
+    _playChannels = 1;
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "Ouput stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mSampleRate = %f, mChannelsPerFrame = %u",
-                 _outStreamFormat.mSampleRate,
-                 _outStreamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 _outStreamFormat.mBytesPerPacket,
-                 _outStreamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 _outStreamFormat.mBytesPerFrame,
-                 _outStreamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mFormatFlags = %u",
-                 _outStreamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &_outStreamFormat.mFormatID);
-
-    // Our preferred format to work with.
-    if (_outStreamFormat.mChannelsPerFrame < 2)
-    {
-        // Disable stereo playout when we only have one channel on the device.
-        _playChannels = 1;
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Stereo playout unavailable on this device");
-    }
-    WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+                 "Stereo playout unavailable on this device");
+  }
+  WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
 
-    // Listen for format changes.
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_outputDeviceID,
-                                                           &propertyAddress,
-                                                           &objectListenerProc,
-                                                           this));
-
-    // Listen for processor overloads.
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
-                                                      &propertyAddress,
-                                                      &objectListenerProc,
-                                                      this));
-
-    if (_twoDevices || !_recIsInitialized)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_outputDeviceID,
-                                                          deviceIOProc,
-                                                          this,
-                                                          &_deviceIOProcID));
-    }
+  // Listen for format changes.
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    _playIsInitialized = true;
+  // Listen for processor overloads.
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    return 0;
+  if (_twoDevices || !_recIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _outputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
+
+  _playIsInitialized = true;
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::InitRecording()
-{
+int32_t AudioDeviceMac::InitRecording() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  if (_recording) {
+    return -1;
+  }
 
-    if (_recording)
-    {
-        return -1;
-    }
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
 
-    if (!_inputDeviceIsSpecified)
-    {
-        return -1;
-    }
+  if (_recIsInitialized) {
+    return 0;
+  }
 
-    if (_recIsInitialized)
-    {
-        return 0;
-    }
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophone() == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  InitMicrophone() failed");
+  }
 
-    // Initialize the microphone (devices might have been added or removed)
-    if (InitMicrophone() == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitMicrophone() failed");
+  if (!SpeakerIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (SpeakerIsAvailable(available) == -1) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "  SpeakerIsAvailable() failed");
     }
+  }
 
-    if (!SpeakerIsInitialized())
-    {
-        // Make this call to check if we are using
-        // one or two devices (_twoDevices)
-        bool available = false;
-        if (SpeakerIsAvailable(available) == -1)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  SpeakerIsAvailable() failed");
-        }
-    }
+  OSStatus err = noErr;
+  UInt32 size = 0;
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-
-    PaUtil_FlushRingBuffer(_paCaptureBuffer);
-
-    _captureDelayUs = 0;
-    _captureLatencyUs = 0;
-    _captureDeviceIsAlive = 1;
-    _doStopRec = false;
-
-    // Get current stream description
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyStreamFormat,
-                kAudioDevicePropertyScopeInput, 0 };
-    memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
-    size = sizeof(_inStreamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &_inStreamFormat));
-
-    if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable input stream format -> mFormatID",
-                 (const char *) &_inStreamFormat.mFormatID);
-        return -1;
-    }
+  PaUtil_FlushRingBuffer(_paCaptureBuffer);
 
-    if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Too many channels on input device (mChannelsPerFrame = %d)",
-            _inStreamFormat.mChannelsPerFrame);
-        return -1;
-    }
+  _captureDelayUs = 0;
+  _captureLatencyUs = 0;
+  _captureDeviceIsAlive = 1;
+  _doStopRec = false;
 
-    const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
-        _inStreamFormat.mSampleRate / 100 * N_BLOCKS_IO;
-    if (io_block_size_samples > _captureBufSizeSamples)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Input IO block size (%d) is larger than ring buffer (%u)",
-            io_block_size_samples, _captureBufSizeSamples);
-        return -1;
-    }
+  // Get current stream description
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
+  size = sizeof(_inStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &_inStreamFormat));
 
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " Input stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mSampleRate = %f, mChannelsPerFrame = %u",
-                 _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 _inStreamFormat.mBytesPerPacket,
-                 _inStreamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 _inStreamFormat.mBytesPerFrame,
-                 _inStreamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mFormatFlags = %u",
-                 _inStreamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &_inStreamFormat.mFormatID);
+  if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable input stream format -> mFormatID",
+             (const char*)&_inStreamFormat.mFormatID);
+    return -1;
+  }
 
-    // Our preferred format to work with
-    if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
-    {
-        _inDesiredFormat.mChannelsPerFrame = 2;
-    } else
-    {
-        // Disable stereo recording when we only have one channel on the device.
-        _inDesiredFormat.mChannelsPerFrame = 1;
-        _recChannels = 1;
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Stereo recording unavailable on this device");
-    }
+  if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on input device (mChannelsPerFrame = %d)",
+                 _inStreamFormat.mChannelsPerFrame);
+    return -1;
+  }
 
-    if (_ptrAudioBuffer)
-    {
-        // Update audio buffer with the selected parameters
-        _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-        _ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
-    }
+  const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
+                                    _inStreamFormat.mSampleRate / 100 *
+                                    N_BLOCKS_IO;
+  if (io_block_size_samples > _captureBufSizeSamples) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Input IO block size (%d) is larger than ring buffer (%u)",
+                 io_block_size_samples, _captureBufSizeSamples);
+    return -1;
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Input stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mSampleRate = %f, mChannelsPerFrame = %u",
+               _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mBytesPerPacket = %u, mFramesPerPacket = %u",
+               _inStreamFormat.mBytesPerPacket,
+               _inStreamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mBytesPerFrame = %u, mBitsPerChannel = %u",
+               _inStreamFormat.mBytesPerFrame, _inStreamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " mFormatFlags = %u",
+               _inStreamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&_inStreamFormat.mFormatID);
+
+  // Our preferred format to work with
+  if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+    _inDesiredFormat.mChannelsPerFrame = 2;
+  } else {
+    // Disable stereo recording when we only have one channel on the device.
+    _inDesiredFormat.mChannelsPerFrame = 1;
+    _recChannels = 1;
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 "Stereo recording unavailable on this device");
+  }
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+    _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+  }
 
-    _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
-    _inDesiredFormat.mBytesPerPacket = _inDesiredFormat.mChannelsPerFrame
-        * sizeof(SInt16);
-    _inDesiredFormat.mFramesPerPacket = 1;
-    _inDesiredFormat.mBytesPerFrame = _inDesiredFormat.mChannelsPerFrame
-        * sizeof(SInt16);
-    _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+  _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
+  _inDesiredFormat.mBytesPerPacket =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mFramesPerPacket = 1;
+  _inDesiredFormat.mBytesPerFrame =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
 
-    _inDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
-        | kLinearPCMFormatFlagIsPacked;
+  _inDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+  _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
-    _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
-
-    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
-            &_captureConverter));
-
-    // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
-    // TODO(xians): investigate this block.
-    UInt32 bufByteCount = (UInt32)((_inStreamFormat.mSampleRate / 1000.0)
-        * 10.0 * N_BLOCKS_IO * _inStreamFormat.mChannelsPerFrame
-        * sizeof(Float32));
-    if (_inStreamFormat.mFramesPerPacket != 0)
-    {
-        if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0)
-        {
-            bufByteCount = ((UInt32)(bufByteCount
-                / _inStreamFormat.mFramesPerPacket) + 1)
-                * _inStreamFormat.mFramesPerPacket;
-        }
+  _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
+                                            &_captureConverter));
+
+  // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
+  // TODO(xians): investigate this block.
+  UInt32 bufByteCount =
+      (UInt32)((_inStreamFormat.mSampleRate / 1000.0) * 10.0 * N_BLOCKS_IO *
+               _inStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_inStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount =
+          ((UInt32)(bufByteCount / _inStreamFormat.mFramesPerPacket) + 1) *
+          _inStreamFormat.mFramesPerPacket;
     }
+  }
 
-    // Ensure the buffer size is within the acceptable range provided by the device.
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
-    AudioValueRange range;
-    size = sizeof(range);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &range));
-    if (range.mMinimum > bufByteCount)
-    {
-        bufByteCount = range.mMinimum;
-    } else if (range.mMaximum < bufByteCount)
-    {
-        bufByteCount = range.mMaximum;
-    }
+  // Ensure the buffer size is within the acceptable range provided by the
+  // device.
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
 
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
-    size = sizeof(bufByteCount);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, size, &bufByteCount));
-
-    // Get capture device latency
-    propertyAddress.mSelector = kAudioDevicePropertyLatency;
-    UInt32 latency = 0;
-    size = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &latency));
-    _captureLatencyUs = (UInt32)((1.0e6 * latency)
-        / _inStreamFormat.mSampleRate);
-
-    // Get capture stream latency
-    propertyAddress.mSelector = kAudioDevicePropertyStreams;
-    AudioStreamID stream = 0;
-    size = sizeof(AudioStreamID);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &stream));
-    propertyAddress.mSelector = kAudioStreamPropertyLatency;
-    size = sizeof(UInt32);
-    latency = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &latency));
-    _captureLatencyUs += (UInt32)((1.0e6 * latency)
-        / _inStreamFormat.mSampleRate);
-
-    // Listen for format changes
-    // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    // Listen for processor overloads
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    if (_twoDevices)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
-                inDeviceIOProc, this, &_inDeviceIOProcID));
-    } else if (!_playIsInitialized)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
-                deviceIOProc, this, &_deviceIOProcID));
-    }
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
+
+  // Get capture device latency
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs = (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
+
+  // Get capture stream latency
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs +=
+      (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
+
+  // Listen for format changes
+  // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  // Listen for processor overloads
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, inDeviceIOProc, this, &_inDeviceIOProcID));
+  } else if (!_playIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
 
-    // Mark recording side as initialized
-    _recIsInitialized = true;
+  // Mark recording side as initialized
+  _recIsInitialized = true;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StartRecording()
-{
-
-    CriticalSectionScoped lock(&_critSect);
+int32_t AudioDeviceMac::StartRecording() {
+  CriticalSectionScoped lock(&_critSect);
 
-    if (!_recIsInitialized)
-    {
-        return -1;
-    }
-
-    if (_recording)
-    {
-        return 0;
-    }
+  if (!_recIsInitialized) {
+    return -1;
+  }
 
-    if (!_initialized)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Recording worker thread has not been started");
-        return -1;
-    }
+  if (_recording) {
+    return 0;
+  }
 
-    RTC_DCHECK(!capture_worker_thread_.get());
-    capture_worker_thread_ =
-        ThreadWrapper::CreateThread(RunCapture, this, "CaptureWorkerThread");
-    RTC_DCHECK(capture_worker_thread_.get());
-    capture_worker_thread_->Start();
-    capture_worker_thread_->SetPriority(kRealtimePriority);
+  if (!_initialized) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Recording worker thread has not been started");
+    return -1;
+  }
 
-    OSStatus err = noErr;
-    if (_twoDevices)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
-    } else if (!_playing)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
-    }
+  RTC_DCHECK(!capture_worker_thread_.get());
+  capture_worker_thread_.reset(
+      new rtc::PlatformThread(RunCapture, this, "CaptureWorkerThread"));
+  RTC_DCHECK(capture_worker_thread_.get());
+  capture_worker_thread_->Start();
+  capture_worker_thread_->SetPriority(rtc::kRealtimePriority);
+
+  OSStatus err = noErr;
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(
+        AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
+  } else if (!_playing) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
+  }
 
-    _recording = true;
+  _recording = true;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StopRecording()
-{
-
-    CriticalSectionScoped lock(&_critSect);
+int32_t AudioDeviceMac::StopRecording() {
+  CriticalSectionScoped lock(&_critSect);
 
-    if (!_recIsInitialized)
-    {
-        return 0;
-    }
-
-    OSStatus err = noErr;
+  if (!_recIsInitialized) {
+    return 0;
+  }
 
-    // Stop device
-    int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
-    if (_twoDevices)
-    {
-        if (_recording && captureDeviceIsAlive == 1)
-        {
-            _recording = false;
-            _doStopRec = true; // Signal to io proc to stop audio device
-            _critSect.Leave(); // Cannot be under lock, risk of deadlock
-            if (kEventTimeout == _stopEventRec.Wait(2000))
-            {
-                CriticalSectionScoped critScoped(&_critSect);
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             " Timed out stopping the capture IOProc. "
-                             "We may have failed to detect a device removal.");
-
-                WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID,
-                                                   _inDeviceIOProcID));
-                WEBRTC_CA_LOG_WARN(
-                    AudioDeviceDestroyIOProcID(_inputDeviceID,
-                                               _inDeviceIOProcID));
-            }
-            _critSect.Enter();
-            _doStopRec = false;
-            WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                         " Recording stopped");
-        }
-    }
-    else
-    {
-        // We signal a stop for a shared device even when rendering has
-        // not yet ended. This is to ensure the IOProc will return early as
-        // intended (by checking |_recording|) before accessing
-        // resources we free below (e.g. the capture converter).
-        //
-        // In the case of a shared devcie, the IOProc will verify
-        // rendering has ended before stopping itself.
-        if (_recording && captureDeviceIsAlive == 1)
-        {
-            _recording = false;
-            _doStop = true; // Signal to io proc to stop audio device
-            _critSect.Leave(); // Cannot be under lock, risk of deadlock
-            if (kEventTimeout == _stopEvent.Wait(2000))
-            {
-                CriticalSectionScoped critScoped(&_critSect);
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             " Timed out stopping the shared IOProc. "
-                             "We may have failed to detect a device removal.");
-
-                // We assume rendering on a shared device has stopped as well if
-                // the IOProc times out.
-                WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
-                                                   _deviceIOProcID));
-                WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                              _deviceIOProcID));
-            }
-            _critSect.Enter();
-            _doStop = false;
-            WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                         " Recording stopped (shared)");
-        }
+  OSStatus err = noErr;
+
+  // Stop device
+  int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
+  if (_twoDevices) {
+    if (_recording && captureDeviceIsAlive == 1) {
+      _recording = false;
+      _doStopRec = true;  // Signal to io proc to stop audio device
+      _critSect.Leave();  // Cannot be under lock, risk of deadlock
+      if (kEventTimeout == _stopEventRec.Wait(2000)) {
+        CriticalSectionScoped critScoped(&_critSect);
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     " Timed out stopping the capture IOProc. "
+                     "We may have failed to detect a device removal.");
+
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      }
+      _critSect.Enter();
+      _doStopRec = false;
+      WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " Recording stopped");
+    }
+  } else {
+    // We signal a stop for a shared device even when rendering has
+    // not yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking |_recording|) before accessing
+    // resources we free below (e.g. the capture converter).
+    //
+    // In the case of a shared devcie, the IOProc will verify
+    // rendering has ended before stopping itself.
+    if (_recording && captureDeviceIsAlive == 1) {
+      _recording = false;
+      _doStop = true;     // Signal to io proc to stop audio device
+      _critSect.Leave();  // Cannot be under lock, risk of deadlock
+      if (kEventTimeout == _stopEvent.Wait(2000)) {
+        CriticalSectionScoped critScoped(&_critSect);
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     " Timed out stopping the shared IOProc. "
+                     "We may have failed to detect a device removal.");
+
+        // We assume rendering on a shared device has stopped as well if
+        // the IOProc times out.
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+      }
+      _critSect.Enter();
+      _doStop = false;
+      WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                   " Recording stopped (shared)");
     }
+  }
 
-    // Setting this signal will allow the worker thread to be stopped.
-    AtomicSet32(&_captureDeviceIsAlive, 0);
+  // Setting this signal will allow the worker thread to be stopped.
+  AtomicSet32(&_captureDeviceIsAlive, 0);
 
-    if (capture_worker_thread_.get()) {
-        _critSect.Leave();
-        capture_worker_thread_->Stop();
-        capture_worker_thread_.reset();
-        _critSect.Enter();
-    }
+  if (capture_worker_thread_.get()) {
+    _critSect.Leave();
+    capture_worker_thread_->Stop();
+    capture_worker_thread_.reset();
+    _critSect.Enter();
+  }
 
-    WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
 
-    // Remove listeners.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyStreamFormat,
-                kAudioDevicePropertyScopeInput, 0 };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    _recIsInitialized = false;
-    _recording = false;
+  _recIsInitialized = false;
+  _recording = false;
 
-    return 0;
+  return 0;
 }
 
-bool AudioDeviceMac::RecordingIsInitialized() const
-{
-    return (_recIsInitialized);
+bool AudioDeviceMac::RecordingIsInitialized() const {
+  return (_recIsInitialized);
 }
 
-bool AudioDeviceMac::Recording() const
-{
-    return (_recording);
+bool AudioDeviceMac::Recording() const {
+  return (_recording);
 }
 
-bool AudioDeviceMac::PlayoutIsInitialized() const
-{
-    return (_playIsInitialized);
+bool AudioDeviceMac::PlayoutIsInitialized() const {
+  return (_playIsInitialized);
 }
 
-int32_t AudioDeviceMac::StartPlayout()
-{
+int32_t AudioDeviceMac::StartPlayout() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized)
-    {
-        return -1;
-    }
+  if (!_playIsInitialized) {
+    return -1;
+  }
 
-    if (_playing)
-    {
-        return 0;
-    }
+  if (_playing) {
+    return 0;
+  }
 
-    RTC_DCHECK(!render_worker_thread_.get());
-    render_worker_thread_ =
-        ThreadWrapper::CreateThread(RunRender, this, "RenderWorkerThread");
-    render_worker_thread_->Start();
-    render_worker_thread_->SetPriority(kRealtimePriority);
+  RTC_DCHECK(!render_worker_thread_.get());
+  render_worker_thread_.reset(
+      new rtc::PlatformThread(RunRender, this, "RenderWorkerThread"));
+  render_worker_thread_->Start();
+  render_worker_thread_->SetPriority(rtc::kRealtimePriority);
 
-    if (_twoDevices || !_recording)
-    {
-        OSStatus err = noErr;
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
-    }
-    _playing = true;
+  if (_twoDevices || !_recording) {
+    OSStatus err = noErr;
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
+  }
+  _playing = true;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StopPlayout()
-{
+int32_t AudioDeviceMac::StopPlayout() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  if (!_playIsInitialized) {
+    return 0;
+  }
 
-    if (!_playIsInitialized)
-    {
-        return 0;
+  OSStatus err = noErr;
+
+  int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
+  if (_playing && renderDeviceIsAlive == 1) {
+    // We signal a stop for a shared device even when capturing has not
+    // yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking |_playing|) before accessing resources we
+    // free below (e.g. the render converter).
+    //
+    // In the case of a shared device, the IOProc will verify capturing
+    // has ended before stopping itself.
+    _playing = false;
+    _doStop = true;     // Signal to io proc to stop audio device
+    _critSect.Leave();  // Cannot be under lock, risk of deadlock
+    if (kEventTimeout == _stopEvent.Wait(2000)) {
+      CriticalSectionScoped critScoped(&_critSect);
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Timed out stopping the render IOProc. "
+                   "We may have failed to detect a device removal.");
+
+      // We assume capturing on a shared device has stopped as well if the
+      // IOProc times out.
+      WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
     }
+    _critSect.Enter();
+    _doStop = false;
+    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Playout stopped");
+  }
 
-    OSStatus err = noErr;
+  // Setting this signal will allow the worker thread to be stopped.
+  AtomicSet32(&_renderDeviceIsAlive, 0);
+  if (render_worker_thread_.get()) {
+    _critSect.Leave();
+    render_worker_thread_->Stop();
+    render_worker_thread_.reset();
+    _critSect.Enter();
+  }
 
-    int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
-    if (_playing && renderDeviceIsAlive == 1)
-    {
-        // We signal a stop for a shared device even when capturing has not
-        // yet ended. This is to ensure the IOProc will return early as
-        // intended (by checking |_playing|) before accessing resources we
-        // free below (e.g. the render converter).
-        //
-        // In the case of a shared device, the IOProc will verify capturing
-        // has ended before stopping itself.
-        _playing = false;
-        _doStop = true; // Signal to io proc to stop audio device
-        _critSect.Leave(); // Cannot be under lock, risk of deadlock
-        if (kEventTimeout == _stopEvent.Wait(2000))
-        {
-            CriticalSectionScoped critScoped(&_critSect);
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Timed out stopping the render IOProc. "
-                         "We may have failed to detect a device removal.");
-
-            // We assume capturing on a shared device has stopped as well if the
-            // IOProc times out.
-            WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
-                                               _deviceIOProcID));
-            WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                          _deviceIOProcID));
-        }
-        _critSect.Enter();
-        _doStop = false;
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "Playout stopped");
-    }
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
 
-    // Setting this signal will allow the worker thread to be stopped.
-    AtomicSet32(&_renderDeviceIsAlive, 0);
-    if (render_worker_thread_.get()) {
-        _critSect.Leave();
-        render_worker_thread_->Stop();
-        render_worker_thread_.reset();
-        _critSect.Enter();
-    }
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    // Remove listeners.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput,
-            0 };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    if (_macBookPro)
-    {
-        Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                     &propertyAddress);
-        if (hasProperty)
-        {
-            propertyAddress.mSelector = kAudioDevicePropertyDataSource;
-            WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-                    &propertyAddress, &objectListenerProc, this));
-        }
+  if (_macBookPro) {
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      propertyAddress.mSelector = kAudioDevicePropertyDataSource;
+      WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
     }
+  }
 
-    _playIsInitialized = false;
-    _playing = false;
+  _playIsInitialized = false;
+  _playing = false;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const
-{
-    int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
-    delayMS = static_cast<uint16_t> (1e-3 * (renderDelayUs + _renderLatencyUs) +
-                                     0.5);
-    return 0;
+int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const {
+  int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
+  delayMS =
+      static_cast<uint16_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+  return 0;
 }
 
-int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const
-{
-    int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
-    delayMS = static_cast<uint16_t> (1e-3 * (captureDelayUs +
-                                             _captureLatencyUs) + 0.5);
-    return 0;
+int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const {
+  int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+  delayMS =
+      static_cast<uint16_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
+  return 0;
 }
 
-bool AudioDeviceMac::Playing() const
-{
-    return (_playing);
+bool AudioDeviceMac::Playing() const {
+  return (_playing);
 }
 
 int32_t AudioDeviceMac::SetPlayoutBuffer(
     const AudioDeviceModule::BufferType type,
-    uint16_t sizeMS)
-{
-
-    if (type != AudioDeviceModule::kFixedBufferSize)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Adaptive buffer size not supported on this platform");
-        return -1;
-    }
+    uint16_t sizeMS) {
+  if (type != AudioDeviceModule::kFixedBufferSize) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Adaptive buffer size not supported on this platform");
+    return -1;
+  }
 
-    _playBufType = type;
-    _playBufDelayFixed = sizeMS;
-    return 0;
+  _playBufType = type;
+  _playBufDelayFixed = sizeMS;
+  return 0;
 }
 
-int32_t AudioDeviceMac::PlayoutBuffer(
-    AudioDeviceModule::BufferType& type,
-    uint16_t& sizeMS) const
-{
-
-    type = _playBufType;
-    sizeMS = _playBufDelayFixed;
+int32_t AudioDeviceMac::PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                      uint16_t& sizeMS) const {
+  type = _playBufType;
+  sizeMS = _playBufDelayFixed;
 
-    return 0;
+  return 0;
 }
 
 // Not implemented for Mac.
-int32_t AudioDeviceMac::CPULoad(uint16_t& /*load*/) const
-{
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
+int32_t AudioDeviceMac::CPULoad(uint16_t& /*load*/) const {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
 
-    return -1;
+  return -1;
 }
 
-bool AudioDeviceMac::PlayoutWarning() const
-{
-    return (_playWarning > 0);
+bool AudioDeviceMac::PlayoutWarning() const {
+  return (_playWarning > 0);
 }
 
-bool AudioDeviceMac::PlayoutError() const
-{
-    return (_playError > 0);
+bool AudioDeviceMac::PlayoutError() const {
+  return (_playError > 0);
 }
 
-bool AudioDeviceMac::RecordingWarning() const
-{
-    return (_recWarning > 0);
+bool AudioDeviceMac::RecordingWarning() const {
+  return (_recWarning > 0);
 }
 
-bool AudioDeviceMac::RecordingError() const
-{
-    return (_recError > 0);
+bool AudioDeviceMac::RecordingError() const {
+  return (_recError > 0);
 }
 
-void AudioDeviceMac::ClearPlayoutWarning()
-{
-    _playWarning = 0;
+void AudioDeviceMac::ClearPlayoutWarning() {
+  _playWarning = 0;
 }
 
-void AudioDeviceMac::ClearPlayoutError()
-{
-    _playError = 0;
+void AudioDeviceMac::ClearPlayoutError() {
+  _playError = 0;
 }
 
-void AudioDeviceMac::ClearRecordingWarning()
-{
-    _recWarning = 0;
+void AudioDeviceMac::ClearRecordingWarning() {
+  _recWarning = 0;
 }
 
-void AudioDeviceMac::ClearRecordingError()
-{
-    _recError = 0;
+void AudioDeviceMac::ClearRecordingError() {
+  _recError = 0;
 }
 
 // ============================================================================
 //                                 Private Methods
 // ============================================================================
 
-int32_t
-AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
-                                 AudioDeviceID scopedDeviceIds[],
-                                 const uint32_t deviceListLength)
-{
-    OSStatus err = noErr;
-
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    UInt32 size = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, &size));
-    if (size == 0)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "No devices");
-        return 0;
-    }
-
-    AudioDeviceID* deviceIds = (AudioDeviceID*) malloc(size);
-    UInt32 numberDevices = size / sizeof(AudioDeviceID);
-    AudioBufferList* bufferList = NULL;
-    UInt32 numberScopedDevices = 0;
+int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
+                                         AudioDeviceID scopedDeviceIds[],
+                                         const uint32_t deviceListLength) {
+  OSStatus err = noErr;
+
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  UInt32 size = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size));
+  if (size == 0) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "No devices");
+    return 0;
+  }
 
-    // First check if there is a default device and list it
-    UInt32 hardwareProperty = 0;
-    if (scope == kAudioDevicePropertyScopeOutput)
-    {
-        hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
-    } else
-    {
-        hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
-    }
+  AudioDeviceID* deviceIds = (AudioDeviceID*)malloc(size);
+  UInt32 numberDevices = size / sizeof(AudioDeviceID);
+  AudioBufferList* bufferList = NULL;
+  UInt32 numberScopedDevices = 0;
+
+  // First check if there is a default device and list it
+  UInt32 hardwareProperty = 0;
+  if (scope == kAudioDevicePropertyScopeOutput) {
+    hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+  } else {
+    hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+  }
 
-    AudioObjectPropertyAddress
-        propertyAddressDefault = { hardwareProperty,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
-
-    AudioDeviceID usedID;
-    UInt32 uintSize = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-            &propertyAddressDefault, 0, NULL, &uintSize, &usedID));
-    if (usedID != kAudioDeviceUnknown)
-    {
-        scopedDeviceIds[numberScopedDevices] = usedID;
-        numberScopedDevices++;
-    } else
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "GetNumberDevices(): Default device unknown");
-    }
+  AudioObjectPropertyAddress propertyAddressDefault = {
+      hardwareProperty, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+
+  AudioDeviceID usedID;
+  UInt32 uintSize = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+                                                     &propertyAddressDefault, 0,
+                                                     NULL, &uintSize, &usedID));
+  if (usedID != kAudioDeviceUnknown) {
+    scopedDeviceIds[numberScopedDevices] = usedID;
+    numberScopedDevices++;
+  } else {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "GetNumberDevices(): Default device unknown");
+  }
 
-    // Then list the rest of the devices
-    bool listOK = true;
+  // Then list the rest of the devices
+  bool listOK = true;
 
-    WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, &size, deviceIds));
-    if (err != noErr)
-    {
+  WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds));
+  if (err != noErr) {
+    listOK = false;
+  } else {
+    propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
+    propertyAddress.mScope = scope;
+    propertyAddress.mElement = 0;
+    for (UInt32 i = 0; i < numberDevices; i++) {
+      // Check for input channels
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(
+          deviceIds[i], &propertyAddress, 0, NULL, &size));
+      if (err == kAudioHardwareBadDeviceError) {
+        // This device doesn't actually exist; continue iterating.
+        continue;
+      } else if (err != noErr) {
         listOK = false;
-    } else
-    {
-        propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
-        propertyAddress.mScope = scope;
-        propertyAddress.mElement = 0;
-        for (UInt32 i = 0; i < numberDevices; i++)
-        {
-            // Check for input channels
-            WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(deviceIds[i],
-                    &propertyAddress, 0, NULL, &size));
-            if (err == kAudioHardwareBadDeviceError)
-            {
-                // This device doesn't actually exist; continue iterating.
-                continue;
-            } else if (err != noErr)
-            {
-                listOK = false;
-                break;
-            }
-
-            bufferList = (AudioBufferList*) malloc(size);
-            WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(deviceIds[i],
-                    &propertyAddress, 0, NULL, &size, bufferList));
-            if (err != noErr)
-            {
-                listOK = false;
-                break;
-            }
-
-            if (bufferList->mNumberBuffers > 0)
-            {
-                if (numberScopedDevices >= deviceListLength)
-                {
-                    WEBRTC_TRACE(kTraceError,
-                                 kTraceAudioDevice, _id,
-                                 "Device list is not long enough");
-                    listOK = false;
-                    break;
-                }
-
-                scopedDeviceIds[numberScopedDevices] = deviceIds[i];
-                numberScopedDevices++;
-            }
-
-            free(bufferList);
-            bufferList = NULL;
-        }  // for
-    }
+        break;
+      }
 
-    if (!listOK)
-    {
-        if (deviceIds)
-        {
-            free(deviceIds);
-            deviceIds = NULL;
+      bufferList = (AudioBufferList*)malloc(size);
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
+          deviceIds[i], &propertyAddress, 0, NULL, &size, bufferList));
+      if (err != noErr) {
+        listOK = false;
+        break;
+      }
+
+      if (bufferList->mNumberBuffers > 0) {
+        if (numberScopedDevices >= deviceListLength) {
+          WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                       "Device list is not long enough");
+          listOK = false;
+          break;
         }
 
-        if (bufferList)
-        {
-            free(bufferList);
-            bufferList = NULL;
-        }
+        scopedDeviceIds[numberScopedDevices] = deviceIds[i];
+        numberScopedDevices++;
+      }
 
-        return -1;
+      free(bufferList);
+      bufferList = NULL;
+    }  // for
+  }
+
+  if (!listOK) {
+    if (deviceIds) {
+      free(deviceIds);
+      deviceIds = NULL;
     }
 
-    // Happy ending
-    if (deviceIds)
-    {
-        free(deviceIds);
-        deviceIds = NULL;
+    if (bufferList) {
+      free(bufferList);
+      bufferList = NULL;
     }
 
-    return numberScopedDevices;
+    return -1;
+  }
+
+  // Happy ending
+  if (deviceIds) {
+    free(deviceIds);
+    deviceIds = NULL;
+  }
+
+  return numberScopedDevices;
 }
 
-int32_t
-AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
-                              const uint16_t index,
-                              char* name)
-{
-    OSStatus err = noErr;
-    UInt32 len = kAdmMaxDeviceNameSize;
-    AudioDeviceID deviceIds[MaxNumberDevices];
-
-    int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
-    if (numberDevices < 0)
-    {
-        return -1;
-    } else if (numberDevices == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "No devices");
-        return -1;
-    }
+int32_t AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
+                                      const uint16_t index,
+                                      char* name) {
+  OSStatus err = noErr;
+  UInt32 len = kAdmMaxDeviceNameSize;
+  AudioDeviceID deviceIds[MaxNumberDevices];
 
-    // If the number is below the number of devices, assume it's "WEBRTC ID"
-    // otherwise assume it's a CoreAudio ID
-    AudioDeviceID usedID;
-
-    // Check if there is a default device
-    bool isDefaultDevice = false;
-    if (index == 0)
-    {
-        UInt32 hardwareProperty = 0;
-        if (scope == kAudioDevicePropertyScopeOutput)
-        {
-            hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
-        } else
-        {
-            hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
-        }
-        AudioObjectPropertyAddress propertyAddress = { hardwareProperty,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
-        UInt32 size = sizeof(UInt32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                &propertyAddress, 0, NULL, &size, &usedID));
-        if (usedID == kAudioDeviceUnknown)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "GetDeviceName(): Default device unknown");
-        } else
-        {
-            isDefaultDevice = true;
-        }
-    }
+  int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "No devices");
+    return -1;
+  }
+
+  // If the number is below the number of devices, assume it's "WEBRTC ID"
+  // otherwise assume it's a CoreAudio ID
+  AudioDeviceID usedID;
 
+  // Check if there is a default device
+  bool isDefaultDevice = false;
+  if (index == 0) {
+    UInt32 hardwareProperty = 0;
+    if (scope == kAudioDevicePropertyScopeOutput) {
+      hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+    } else {
+      hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+    }
     AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyDeviceName, scope, 0 };
-
-    if (isDefaultDevice)
-    {
-        char devName[len];
-
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
-                &propertyAddress, 0, NULL, &len, devName));
-
-        sprintf(name, "default (%s)", devName);
-    } else
-    {
-        if (index < numberDevices)
-        {
-            usedID = deviceIds[index];
-        } else
-        {
-            usedID = index;
-        }
+        hardwareProperty, kAudioObjectPropertyScopeGlobal,
+        kAudioObjectPropertyElementMaster};
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &usedID));
+    if (usedID == kAudioDeviceUnknown) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "GetDeviceName(): Default device unknown");
+    } else {
+      isDefaultDevice = true;
+    }
+  }
 
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
-                &propertyAddress, 0, NULL, &len, name));
+  AudioObjectPropertyAddress propertyAddress = {kAudioDevicePropertyDeviceName,
+                                                scope, 0};
+
+  if (isDefaultDevice) {
+    char devName[len];
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID, &propertyAddress,
+                                                       0, NULL, &len, devName));
+
+    sprintf(name, "default (%s)", devName);
+  } else {
+    if (index < numberDevices) {
+      usedID = deviceIds[index];
+    } else {
+      usedID = index;
     }
 
-    return 0;
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID, &propertyAddress,
+                                                       0, NULL, &len, name));
+  }
+
+  return 0;
 }
 
 int32_t AudioDeviceMac::InitDevice(const uint16_t userDeviceIndex,
                                    AudioDeviceID& deviceId,
-                                   const bool isInput)
-{
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    AudioObjectPropertyScope deviceScope;
-    AudioObjectPropertySelector defaultDeviceSelector;
-    AudioDeviceID deviceIds[MaxNumberDevices];
-
-    if (isInput)
-    {
-        deviceScope = kAudioDevicePropertyScopeInput;
-        defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
-    } else
-    {
-        deviceScope = kAudioDevicePropertyScopeOutput;
-        defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
-    }
+                                   const bool isInput) {
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  AudioObjectPropertyScope deviceScope;
+  AudioObjectPropertySelector defaultDeviceSelector;
+  AudioDeviceID deviceIds[MaxNumberDevices];
+
+  if (isInput) {
+    deviceScope = kAudioDevicePropertyScopeInput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
+  } else {
+    deviceScope = kAudioDevicePropertyScopeOutput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
+  }
 
-    AudioObjectPropertyAddress
-        propertyAddress = { defaultDeviceSelector,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
-
-    // Get the actual device IDs
-    int numberDevices = GetNumberDevices(deviceScope, deviceIds,
-                                         MaxNumberDevices);
-    if (numberDevices < 0)
-    {
-        return -1;
-    } else if (numberDevices == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "InitDevice(): No devices");
-        return -1;
-    }
+  AudioObjectPropertyAddress propertyAddress = {
+      defaultDeviceSelector, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
 
-    bool isDefaultDevice = false;
-    deviceId = kAudioDeviceUnknown;
-    if (userDeviceIndex == 0)
-    {
-        // Try to use default system device
-        size = sizeof(AudioDeviceID);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                &propertyAddress, 0, NULL, &size, &deviceId));
-        if (deviceId == kAudioDeviceUnknown)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " No default device exists");
-        } else
-        {
-            isDefaultDevice = true;
-        }
-    }
+  // Get the actual device IDs
+  int numberDevices =
+      GetNumberDevices(deviceScope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "InitDevice(): No devices");
+    return -1;
+  }
 
-    if (!isDefaultDevice)
-    {
-        deviceId = deviceIds[userDeviceIndex];
+  bool isDefaultDevice = false;
+  deviceId = kAudioDeviceUnknown;
+  if (userDeviceIndex == 0) {
+    // Try to use default system device
+    size = sizeof(AudioDeviceID);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &deviceId));
+    if (deviceId == kAudioDeviceUnknown) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " No default device exists");
+    } else {
+      isDefaultDevice = true;
     }
+  }
 
-    // Obtain device name and manufacturer for logging.
-    // Also use this as a test to ensure a user-set device ID is valid.
-    char devName[128];
-    char devManf[128];
-    memset(devName, 0, sizeof(devName));
-    memset(devManf, 0, sizeof(devManf));
-
-    propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
-    propertyAddress.mScope = deviceScope;
-    propertyAddress.mElement = 0;
-    size = sizeof(devName);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
-            &propertyAddress, 0, NULL, &size, devName));
-
-    propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
-    size = sizeof(devManf);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
-            &propertyAddress, 0, NULL, &size, devManf));
+  if (!isDefaultDevice) {
+    deviceId = deviceIds[userDeviceIndex];
+  }
 
-    if (isInput)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Input device: %s %s", devManf, devName);
-    } else
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Output device: %s %s", devManf, devName);
-    }
+  // Obtain device name and manufacturer for logging.
+  // Also use this as a test to ensure a user-set device ID is valid.
+  char devName[128];
+  char devManf[128];
+  memset(devName, 0, sizeof(devName));
+  memset(devManf, 0, sizeof(devManf));
+
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
+  propertyAddress.mScope = deviceScope;
+  propertyAddress.mElement = 0;
+  size = sizeof(devName);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devName));
+
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
+  size = sizeof(devManf);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devManf));
+
+  if (isInput) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Input device: %s %s",
+                 devManf, devName);
+  } else {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Output device: %s %s",
+                 devManf, devName);
+  }
 
-    return 0;
+  return 0;
 }
 
-OSStatus AudioDeviceMac::SetDesiredPlayoutFormat()
-{
-    // Our preferred format to work with.
-    _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
-    _outDesiredFormat.mChannelsPerFrame = _playChannels;
+OSStatus AudioDeviceMac::SetDesiredPlayoutFormat() {
+  // Our preferred format to work with.
+  _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
+  _outDesiredFormat.mChannelsPerFrame = _playChannels;
 
-    if (_ptrAudioBuffer)
-    {
-        // Update audio buffer with the selected parameters.
-        _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
-        _ptrAudioBuffer->SetPlayoutChannels((uint8_t) _playChannels);
-    }
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters.
+    _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+  }
 
-    _renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT *
-        ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
+  _renderDelayOffsetSamples = _renderBufSizeSamples -
+                              N_BUFFERS_OUT * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES *
+                                  _outDesiredFormat.mChannelsPerFrame;
 
-    _outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame *
-                                        sizeof(SInt16);
-    // In uncompressed audio, a packet is one frame.
-    _outDesiredFormat.mFramesPerPacket = 1;
-    _outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame *
-                                       sizeof(SInt16);
-    _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+  _outDesiredFormat.mBytesPerPacket =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  // In uncompressed audio, a packet is one frame.
+  _outDesiredFormat.mFramesPerPacket = 1;
+  _outDesiredFormat.mBytesPerFrame =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
 
-    _outDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |
-                                     kLinearPCMFormatFlagIsPacked;
+  _outDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+  _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
-    _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
-
-    OSStatus err = noErr;
-    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat,
-                                              &_outStreamFormat,
-                                              &_renderConverter));
-
-    // Try to set buffer size to desired value (_playBufDelayFixed).
-    UInt32 bufByteCount = static_cast<UInt32> ((_outStreamFormat.mSampleRate /
-                          1000.0) *
-                          _playBufDelayFixed *
-                          _outStreamFormat.mChannelsPerFrame *
-                          sizeof(Float32));
-    if (_outStreamFormat.mFramesPerPacket != 0)
-    {
-        if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0)
-        {
-            bufByteCount = (static_cast<UInt32> (bufByteCount /
-                           _outStreamFormat.mFramesPerPacket) + 1) *
-                           _outStreamFormat.mFramesPerPacket;
-        }
-    }
-
-    // Ensure the buffer size is within the range provided by the device.
-    AudioObjectPropertyAddress propertyAddress =
-        {kAudioDevicePropertyDataSource,
-         kAudioDevicePropertyScopeOutput,
-         0};
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
-    AudioValueRange range;
-    UInt32 size = sizeof(range);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &range));
-    if (range.mMinimum > bufByteCount)
-    {
-        bufByteCount = range.mMinimum;
-    } else if (range.mMaximum < bufByteCount)
-    {
-        bufByteCount = range.mMaximum;
+  _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+  OSStatus err = noErr;
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(
+      &_outDesiredFormat, &_outStreamFormat, &_renderConverter));
+
+  // Try to set buffer size to desired value (_playBufDelayFixed).
+  UInt32 bufByteCount = static_cast<UInt32>(
+      (_outStreamFormat.mSampleRate / 1000.0) * _playBufDelayFixed *
+      _outStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_outStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount = (static_cast<UInt32>(bufByteCount /
+                                          _outStreamFormat.mFramesPerPacket) +
+                      1) *
+                     _outStreamFormat.mFramesPerPacket;
     }
+  }
 
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
-    size = sizeof(bufByteCount);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       size,
-                                                       &bufByteCount));
-
-    // Get render device latency.
-    propertyAddress.mSelector = kAudioDevicePropertyLatency;
-    UInt32 latency = 0;
-    size = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &latency));
-    _renderLatencyUs = static_cast<uint32_t> ((1.0e6 * latency) /
-                       _outStreamFormat.mSampleRate);
-
-    // Get render stream latency.
-    propertyAddress.mSelector = kAudioDevicePropertyStreams;
-    AudioStreamID stream = 0;
-    size = sizeof(AudioStreamID);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &stream));
-    propertyAddress.mSelector = kAudioStreamPropertyLatency;
-    size = sizeof(UInt32);
-    latency = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &latency));
-    _renderLatencyUs += static_cast<uint32_t> ((1.0e6 * latency) /
-                        _outStreamFormat.mSampleRate);
+  // Ensure the buffer size is within the range provided by the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  UInt32 size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
 
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  initial playout status: _renderDelayOffsetSamples=%d,"
-                 " _renderDelayUs=%d, _renderLatencyUs=%d",
-                 _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
-    return 0;
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
+
+  // Get render device latency.
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs =
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
+
+  // Get render stream latency.
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs +=
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  initial playout status: _renderDelayOffsetSamples=%d,"
+               " _renderDelayUs=%d, _renderLatencyUs=%d",
+               _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
+  return 0;
 }
 
 OSStatus AudioDeviceMac::objectListenerProc(
     AudioObjectID objectId,
     UInt32 numberAddresses,
     const AudioObjectPropertyAddress addresses[],
-    void* clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    RTC_DCHECK(ptrThis != NULL);
+    void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
+  ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
 
-    // AudioObjectPropertyListenerProc functions are supposed to return 0
-    return 0;
+  // AudioObjectPropertyListenerProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::implObjectListenerProc(
     const AudioObjectID objectId,
     const UInt32 numberAddresses,
-    const AudioObjectPropertyAddress addresses[])
-{
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "AudioDeviceMac::implObjectListenerProc()");
-
-    for (UInt32 i = 0; i < numberAddresses; i++)
-    {
-        if (addresses[i].mSelector == kAudioHardwarePropertyDevices)
-        {
-            HandleDeviceChange();
-        } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat)
-        {
-            HandleStreamFormatChange(objectId, addresses[i]);
-        } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource)
-        {
-            HandleDataSourceChange(objectId, addresses[i]);
-        } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload)
-        {
-            HandleProcessorOverload(addresses[i]);
-        }
+    const AudioObjectPropertyAddress addresses[]) {
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+               "AudioDeviceMac::implObjectListenerProc()");
+
+  for (UInt32 i = 0; i < numberAddresses; i++) {
+    if (addresses[i].mSelector == kAudioHardwarePropertyDevices) {
+      HandleDeviceChange();
+    } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat) {
+      HandleStreamFormatChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource) {
+      HandleDataSourceChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload) {
+      HandleProcessorOverload(addresses[i]);
     }
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::HandleDeviceChange()
-{
-    OSStatus err = noErr;
+int32_t AudioDeviceMac::HandleDeviceChange() {
+  OSStatus err = noErr;
 
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "kAudioHardwarePropertyDevices");
-
-    // A device has changed. Check if our registered devices have been removed.
-    // Ensure the devices have been initialized, meaning the IDs are valid.
-    if (MicrophoneIsInitialized())
-    {
-        AudioObjectPropertyAddress propertyAddress = {
-                kAudioDevicePropertyDeviceIsAlive,
-                kAudioDevicePropertyScopeInput, 0 };
-        UInt32 deviceIsAlive = 1;
-        UInt32 size = sizeof(UInt32);
-        err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0,
-                                         NULL, &size, &deviceIsAlive);
-
-        if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "Capture device is not alive (probably removed)");
-            AtomicSet32(&_captureDeviceIsAlive, 0);
-            _mixerManager.CloseMicrophone();
-            if (_recError == 1)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                             _id, "  pending recording error exists");
-            }
-            _recError = 1; // triggers callback from module process thread
-        } else if (err != noErr)
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioDeviceGetProperty()", (const char*) &err);
-            return -1;
-        }
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+               "kAudioHardwarePropertyDevices");
+
+  // A device has changed. Check if our registered devices have been removed.
+  // Ensure the devices have been initialized, meaning the IDs are valid.
+  if (MicrophoneIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeInput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
+
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "Capture device is not alive (probably removed)");
+      AtomicSet32(&_captureDeviceIsAlive, 0);
+      _mixerManager.CloseMicrophone();
+      if (_recError == 1) {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     "  pending recording error exists");
+      }
+      _recError = 1;  // triggers callback from module process thread
+    } else if (err != noErr) {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioDeviceGetProperty()", (const char*)&err);
+      return -1;
     }
+  }
 
-    if (SpeakerIsInitialized())
-    {
-        AudioObjectPropertyAddress propertyAddress = {
-                kAudioDevicePropertyDeviceIsAlive,
-                kAudioDevicePropertyScopeOutput, 0 };
-        UInt32 deviceIsAlive = 1;
-        UInt32 size = sizeof(UInt32);
-        err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0,
-                                         NULL, &size, &deviceIsAlive);
-
-        if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "Render device is not alive (probably removed)");
-            AtomicSet32(&_renderDeviceIsAlive, 0);
-            _mixerManager.CloseSpeaker();
-            if (_playError == 1)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                             _id, "  pending playout error exists");
-            }
-            _playError = 1; // triggers callback from module process thread
-        } else if (err != noErr)
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioDeviceGetProperty()", (const char*) &err);
-            return -1;
-        }
+  if (SpeakerIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeOutput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
+
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "Render device is not alive (probably removed)");
+      AtomicSet32(&_renderDeviceIsAlive, 0);
+      _mixerManager.CloseSpeaker();
+      if (_playError == 1) {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     "  pending playout error exists");
+      }
+      _playError = 1;  // triggers callback from module process thread
+    } else if (err != noErr) {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioDeviceGetProperty()", (const char*)&err);
+      return -1;
     }
+  }
 
-    return 0;
+  return 0;
 }
 
 int32_t AudioDeviceMac::HandleStreamFormatChange(
     const AudioObjectID objectId,
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    OSStatus err = noErr;
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
 
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "Stream format changed");
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Stream format changed");
 
-    if (objectId != _inputDeviceID && objectId != _outputDeviceID)
-    {
-        return 0;
-    }
-
-    // Get the new device format
-    AudioStreamBasicDescription streamFormat;
-    UInt32 size = sizeof(streamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
-            &propertyAddress, 0, NULL, &size, &streamFormat));
-
-    if (streamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable input stream format -> mFormatID",
-                 (const char *) &streamFormat.mFormatID);
-        return -1;
-    }
-
-    if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Too many channels on device (mChannelsPerFrame = %d)",
-                     streamFormat.mChannelsPerFrame);
-        return -1;
-    }
+  if (objectId != _inputDeviceID && objectId != _outputDeviceID) {
+    return 0;
+  }
 
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "Stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mSampleRate = %f, mChannelsPerFrame = %u",
-                 streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mFormatFlags = %u",
-                 streamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &streamFormat.mFormatID);
-
-    if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
-    {
-        const int io_block_size_samples = streamFormat.mChannelsPerFrame *
-            streamFormat.mSampleRate / 100 * N_BLOCKS_IO;
-        if (io_block_size_samples > _captureBufSizeSamples)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                "Input IO block size (%d) is larger than ring buffer (%u)",
-                io_block_size_samples, _captureBufSizeSamples);
-            return -1;
+  // Get the new device format
+  AudioStreamBasicDescription streamFormat;
+  UInt32 size = sizeof(streamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      objectId, &propertyAddress, 0, NULL, &size, &streamFormat));
 
-        }
+  if (streamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable input stream format -> mFormatID",
+             (const char*)&streamFormat.mFormatID);
+    return -1;
+  }
 
-        memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
-
-        if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
-        {
-            _inDesiredFormat.mChannelsPerFrame = 2;
-        } else
-        {
-            // Disable stereo recording when we only have one channel on the device.
-            _inDesiredFormat.mChannelsPerFrame = 1;
-            _recChannels = 1;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "Stereo recording unavailable on this device");
-        }
+  if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on device (mChannelsPerFrame = %d)",
+                 streamFormat.mChannelsPerFrame);
+    return -1;
+  }
 
-        if (_ptrAudioBuffer)
-        {
-            // Update audio buffer with the selected parameters
-            _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-            _ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
-        }
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mSampleRate = %f, mChannelsPerFrame = %u",
+               streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerPacket = %u, mFramesPerPacket = %u",
+               streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerFrame = %u, mBitsPerChannel = %u",
+               streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "mFormatFlags = %u",
+               streamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&streamFormat.mFormatID);
+
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    const int io_block_size_samples = streamFormat.mChannelsPerFrame *
+                                      streamFormat.mSampleRate / 100 *
+                                      N_BLOCKS_IO;
+    if (io_block_size_samples > _captureBufSizeSamples) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   "Input IO block size (%d) is larger than ring buffer (%u)",
+                   io_block_size_samples, _captureBufSizeSamples);
+      return -1;
+    }
+
+    memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
+
+    if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+      _inDesiredFormat.mChannelsPerFrame = 2;
+    } else {
+      // Disable stereo recording when we only have one channel on the device.
+      _inDesiredFormat.mChannelsPerFrame = 1;
+      _recChannels = 1;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "Stereo recording unavailable on this device");
+    }
+
+    if (_ptrAudioBuffer) {
+      // Update audio buffer with the selected parameters
+      _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+      _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+    }
+
+    // Recreate the converter with the new format
+    // TODO(xians): make this thread safe
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
+                                              &_captureConverter));
+  } else {
+    memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
 
-        // Recreate the converter with the new format
-        // TODO(xians): make this thread safe
-        WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
-
-        WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
-                &_captureConverter));
-    } else
-    {
-        memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
-
-        // Our preferred format to work with
-        if (_outStreamFormat.mChannelsPerFrame < 2)
-        {
-            _playChannels = 1;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "Stereo playout unavailable on this device");
-        }
-        WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+    // Our preferred format to work with
+    if (_outStreamFormat.mChannelsPerFrame < 2) {
+      _playChannels = 1;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "Stereo playout unavailable on this device");
     }
-    return 0;
+    WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+  }
+  return 0;
 }
 
 int32_t AudioDeviceMac::HandleDataSourceChange(
     const AudioObjectID objectId,
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    OSStatus err = noErr;
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
 
-    if (_macBookPro && propertyAddress.mScope
-        == kAudioDevicePropertyScopeOutput)
-    {
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "Data source changed");
-
-        _macBookProPanRight = false;
-        UInt32 dataSource = 0;
-        UInt32 size = sizeof(UInt32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
-                &propertyAddress, 0, NULL, &size, &dataSource));
-        if (dataSource == 'ispk')
-        {
-            _macBookProPanRight = true;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "MacBook Pro using internal speakers; stereo panning right");
-        } else
-        {
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "MacBook Pro not using internal speakers");
-        }
+  if (_macBookPro &&
+      propertyAddress.mScope == kAudioDevicePropertyScopeOutput) {
+    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Data source changed");
+
+    _macBookProPanRight = false;
+    UInt32 dataSource = 0;
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        objectId, &propertyAddress, 0, NULL, &size, &dataSource));
+    if (dataSource == 'ispk') {
+      _macBookProPanRight = true;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "MacBook Pro using internal speakers; stereo panning right");
+    } else {
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "MacBook Pro not using internal speakers");
     }
+  }
 
-    return 0;
+  return 0;
 }
 int32_t AudioDeviceMac::HandleProcessorOverload(
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    // TODO(xians): we probably want to notify the user in some way of the
-    // overload. However, the Windows interpretations of these errors seem to
-    // be more severe than what ProcessorOverload is thrown for.
-    //
-    // We don't log the notification, as it's sent from the HAL's IO thread. We
-    // don't want to slow it down even further.
-    if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
-    {
-        //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
-        // overload");
-        //_callback->ProblemIsReported(
-        // SndCardStreamObserver::ERecordingProblem);
-    } else
-    {
-        //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-        // "Render processor overload");
-        //_callback->ProblemIsReported(
-        // SndCardStreamObserver::EPlaybackProblem);
-    }
+    const AudioObjectPropertyAddress propertyAddress) {
+  // TODO(xians): we probably want to notify the user in some way of the
+  // overload. However, the Windows interpretations of these errors seem to
+  // be more severe than what ProcessorOverload is thrown for.
+  //
+  // We don't log the notification, as it's sent from the HAL's IO thread. We
+  // don't want to slow it down even further.
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    // WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
+    // overload");
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::ERecordingProblem);
+  } else {
+    // WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+    // "Render processor overload");
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::EPlaybackProblem);
+  }
 
-    return 0;
+  return 0;
 }
 
 // ============================================================================
 //                                  Thread Methods
 // ============================================================================
 
-OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID,
+                                      const AudioTimeStamp*,
                                       const AudioBufferList* inputData,
                                       const AudioTimeStamp* inputTime,
                                       AudioBufferList* outputData,
                                       const AudioTimeStamp* outputTime,
-                                      void *clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    RTC_DCHECK(ptrThis != NULL);
+                                      void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
+  ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
 
-    // AudioDeviceIOProc functions are supposed to return 0
-    return 0;
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::outConverterProc(AudioConverterRef,
-                                          UInt32 *numberDataPackets,
-                                          AudioBufferList *data,
-                                          AudioStreamPacketDescription **,
-                                          void *userData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) userData;
-    RTC_DCHECK(ptrThis != NULL);
+                                          UInt32* numberDataPackets,
+                                          AudioBufferList* data,
+                                          AudioStreamPacketDescription**,
+                                          void* userData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)userData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    return ptrThis->implOutConverterProc(numberDataPackets, data);
+  return ptrThis->implOutConverterProc(numberDataPackets, data);
 }
 
-OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID,
+                                        const AudioTimeStamp*,
                                         const AudioBufferList* inputData,
                                         const AudioTimeStamp* inputTime,
                                         AudioBufferList*,
-                                        const AudioTimeStamp*, void* clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    RTC_DCHECK(ptrThis != NULL);
+                                        const AudioTimeStamp*,
+                                        void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    ptrThis->implInDeviceIOProc(inputData, inputTime);
+  ptrThis->implInDeviceIOProc(inputData, inputTime);
 
-    // AudioDeviceIOProc functions are supposed to return 0
-    return 0;
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::inConverterProc(
     AudioConverterRef,
-    UInt32 *numberDataPackets,
-    AudioBufferList *data,
-    AudioStreamPacketDescription ** /*dataPacketDescription*/,
-    void *userData)
-{
-    AudioDeviceMac *ptrThis = static_cast<AudioDeviceMac*> (userData);
-    RTC_DCHECK(ptrThis != NULL);
-
-    return ptrThis->implInConverterProc(numberDataPackets, data);
-}
-
-OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList *inputData,
-                                          const AudioTimeStamp *inputTime,
-                                          AudioBufferList *outputData,
-                                          const AudioTimeStamp *outputTime)
-{
-    OSStatus err = noErr;
-    UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
-    UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
-    if (!_twoDevices && _recording)
-    {
-        implInDeviceIOProc(inputData, inputTime);
-    }
+    UInt32* numberDataPackets,
+    AudioBufferList* data,
+    AudioStreamPacketDescription** /*dataPacketDescription*/,
+    void* userData) {
+  AudioDeviceMac* ptrThis = static_cast<AudioDeviceMac*>(userData);
+  RTC_DCHECK(ptrThis != NULL);
+
+  return ptrThis->implInConverterProc(numberDataPackets, data);
+}
+
+OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList* inputData,
+                                          const AudioTimeStamp* inputTime,
+                                          AudioBufferList* outputData,
+                                          const AudioTimeStamp* outputTime) {
+  OSStatus err = noErr;
+  UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+  if (!_twoDevices && _recording) {
+    implInDeviceIOProc(inputData, inputTime);
+  }
 
-    // Check if we should close down audio device
-    // Double-checked locking optimization to remove locking overhead
-    if (_doStop)
-    {
-        _critSect.Enter();
-        if (_doStop)
-        {
-            if (_twoDevices || (!_recording && !_playing))
-            {
-               // In the case of a shared device, the single driving ioProc
-               // is stopped here
-               WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID,
-                                                 _deviceIOProcID));
-               WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                             _deviceIOProcID));
-               if (err == noErr)
-               {
-                  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                               _id, " Playout or shared device stopped");
-               }
-            }
-
-            _doStop = false;
-            _stopEvent.Set();
-            _critSect.Leave();
-            return 0;
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStop) {
+    _critSect.Enter();
+    if (_doStop) {
+      if (_twoDevices || (!_recording && !_playing)) {
+        // In the case of a shared device, the single driving ioProc
+        // is stopped here
+        WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+        if (err == noErr) {
+          WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                       " Playout or shared device stopped");
         }
-        _critSect.Leave();
-    }
+      }
 
-    if (!_playing)
-    {
-        // This can be the case when a shared device is capturing but not
-        // rendering. We allow the checks above before returning to avoid a
-        // timeout when capturing is stopped.
-        return 0;
+      _doStop = false;
+      _stopEvent.Set();
+      _critSect.Leave();
+      return 0;
     }
+    _critSect.Leave();
+  }
 
-    RTC_DCHECK(_outStreamFormat.mBytesPerFrame != 0);
-    UInt32 size = outputData->mBuffers->mDataByteSize
-        / _outStreamFormat.mBytesPerFrame;
-
-    // TODO(xians): signal an error somehow?
-    err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
-                                          this, &size, outputData, NULL);
-    if (err != noErr)
-    {
-        if (err == 1)
-        {
-            // This is our own error.
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " Error in AudioConverterFillComplexBuffer()");
-            return 1;
-        } else
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioConverterFillComplexBuffer()",
-                     (const char *) &err);
-            return 1;
-        }
+  if (!_playing) {
+    // This can be the case when a shared device is capturing but not
+    // rendering. We allow the checks above before returning to avoid a
+    // timeout when capturing is stopped.
+    return 0;
+  }
+
+  RTC_DCHECK(_outStreamFormat.mBytesPerFrame != 0);
+  UInt32 size =
+      outputData->mBuffers->mDataByteSize / _outStreamFormat.mBytesPerFrame;
+
+  // TODO(xians): signal an error somehow?
+  err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
+                                        this, &size, outputData, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " Error in AudioConverterFillComplexBuffer()");
+      return 1;
+    } else {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+      return 1;
     }
+  }
 
-    PaRingBufferSize bufSizeSamples =
-        PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
+  PaRingBufferSize bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
 
-    int32_t renderDelayUs = static_cast<int32_t> (1e-3 * (outputTimeNs - nowNs)
-        + 0.5);
-    renderDelayUs += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
-        / _outDesiredFormat.mChannelsPerFrame / _outDesiredFormat.mSampleRate
-        + 0.5);
+  int32_t renderDelayUs =
+      static_cast<int32_t>(1e-3 * (outputTimeNs - nowNs) + 0.5);
+  renderDelayUs += static_cast<int32_t>(
+      (1.0e6 * bufSizeSamples) / _outDesiredFormat.mChannelsPerFrame /
+          _outDesiredFormat.mSampleRate +
+      0.5);
 
-    AtomicSet32(&_renderDelayUs, renderDelayUs);
+  AtomicSet32(&_renderDelayUs, renderDelayUs);
 
-    return 0;
+  return 0;
 }
 
-OSStatus AudioDeviceMac::implOutConverterProc(UInt32 *numberDataPackets,
-                                              AudioBufferList *data)
-{
+OSStatus AudioDeviceMac::implOutConverterProc(UInt32* numberDataPackets,
+                                              AudioBufferList* data) {
   RTC_DCHECK(data->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = *numberDataPackets
-        * _outDesiredFormat.mChannelsPerFrame;
-
-    data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
-    // Always give the converter as much as it wants, zero padding as required.
-    data->mBuffers->mDataByteSize = *numberDataPackets
-        * _outDesiredFormat.mBytesPerPacket;
-    data->mBuffers->mData = _renderConvertData;
-    memset(_renderConvertData, 0, sizeof(_renderConvertData));
-
-    PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
-
-    kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_signal_all() error: %d", kernErr);
-        return 1;
-    }
+  PaRingBufferSize numSamples =
+      *numberDataPackets * _outDesiredFormat.mChannelsPerFrame;
 
-    return 0;
+  data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
+  // Always give the converter as much as it wants, zero padding as required.
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _outDesiredFormat.mBytesPerPacket;
+  data->mBuffers->mData = _renderConvertData;
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
+
+  PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
+
+  kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_signal_all() error: %d", kernErr);
+    return 1;
+  }
+
+  return 0;
 }
 
-OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList *inputData,
-                                            const AudioTimeStamp *inputTime)
-{
-    OSStatus err = noErr;
-    UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
-    UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
-    // Check if we should close down audio device
-    // Double-checked locking optimization to remove locking overhead
-    if (_doStopRec)
-    {
-        _critSect.Enter();
-        if (_doStopRec)
-        {
-            // This will be signalled only when a shared device is not in use.
-            WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
-            WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_inputDeviceID,
-                                                          _inDeviceIOProcID));
-            if (err == noErr)
-            {
-                WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                             _id, " Recording device stopped");
-            }
-
-            _doStopRec = false;
-            _stopEventRec.Set();
-            _critSect.Leave();
-            return 0;
-        }
-        _critSect.Leave();
-    }
+OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList* inputData,
+                                            const AudioTimeStamp* inputTime) {
+  OSStatus err = noErr;
+  UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
 
-    if (!_recording)
-    {
-        // Allow above checks to avoid a timeout on stopping capture.
-        return 0;
-    }
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStopRec) {
+    _critSect.Enter();
+    if (_doStopRec) {
+      // This will be signalled only when a shared device is not in use.
+      WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      if (err == noErr) {
+        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                     " Recording device stopped");
+      }
 
-    PaRingBufferSize bufSizeSamples =
-        PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
-
-    int32_t captureDelayUs = static_cast<int32_t> (1e-3 * (nowNs - inputTimeNs)
-        + 0.5);
-    captureDelayUs
-        += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
-            / _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mSampleRate
-            + 0.5);
-
-    AtomicSet32(&_captureDelayUs, captureDelayUs);
-
-    RTC_DCHECK(inputData->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = inputData->mBuffers->mDataByteSize
-        * _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mBytesPerPacket;
-    PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
-                           numSamples);
-
-    kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_signal_all() error: %d", kernErr);
+      _doStopRec = false;
+      _stopEventRec.Set();
+      _critSect.Leave();
+      return 0;
     }
+    _critSect.Leave();
+  }
 
-    return err;
-}
+  if (!_recording) {
+    // Allow above checks to avoid a timeout on stopping capture.
+    return 0;
+  }
 
-OSStatus AudioDeviceMac::implInConverterProc(UInt32 *numberDataPackets,
-                                             AudioBufferList *data)
-{
-  RTC_DCHECK(data->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = *numberDataPackets
-        * _inStreamFormat.mChannelsPerFrame;
-
-    while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples)
-    {
-        mach_timespec_t timeout;
-        timeout.tv_sec = 0;
-        timeout.tv_nsec = TIMER_PERIOD_MS;
-
-        kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
-        if (kernErr == KERN_OPERATION_TIMED_OUT)
-        {
-            int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
-            if (signal == 0)
-            {
-                // The capture device is no longer alive; stop the worker thread.
-                *numberDataPackets = 0;
-                return 1;
-            }
-        } else if (kernErr != KERN_SUCCESS)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " semaphore_wait() error: %d", kernErr);
-        }
-    }
+  PaRingBufferSize bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
 
-    // Pass the read pointer directly to the converter to avoid a memcpy.
-    void* dummyPtr;
-    PaRingBufferSize dummySize;
-    PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
-                                    &data->mBuffers->mData, &numSamples,
-                                    &dummyPtr, &dummySize);
-    PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
+  int32_t captureDelayUs =
+      static_cast<int32_t>(1e-3 * (nowNs - inputTimeNs) + 0.5);
+  captureDelayUs += static_cast<int32_t>((1.0e6 * bufSizeSamples) /
+                                             _inStreamFormat.mChannelsPerFrame /
+                                             _inStreamFormat.mSampleRate +
+                                         0.5);
 
-    data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
-    *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
-    data->mBuffers->mDataByteSize = *numberDataPackets
-        * _inStreamFormat.mBytesPerPacket;
+  AtomicSet32(&_captureDelayUs, captureDelayUs);
 
-    return 0;
+  RTC_DCHECK(inputData->mNumberBuffers == 1);
+  PaRingBufferSize numSamples = inputData->mBuffers->mDataByteSize *
+                                _inStreamFormat.mChannelsPerFrame /
+                                _inStreamFormat.mBytesPerPacket;
+  PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
+                         numSamples);
+
+  kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_signal_all() error: %d", kernErr);
+  }
+
+  return err;
 }
 
-bool AudioDeviceMac::RunRender(void* ptrThis)
-{
-    return static_cast<AudioDeviceMac*> (ptrThis)->RenderWorkerThread();
-}
-
-bool AudioDeviceMac::RenderWorkerThread()
-{
-    PaRingBufferSize numSamples = ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
-        * _outDesiredFormat.mChannelsPerFrame;
-    while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer)
-        - _renderDelayOffsetSamples < numSamples)
-    {
-        mach_timespec_t timeout;
-        timeout.tv_sec = 0;
-        timeout.tv_nsec = TIMER_PERIOD_MS;
-
-        kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
-        if (kernErr == KERN_OPERATION_TIMED_OUT)
-        {
-            int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
-            if (signal == 0)
-            {
-                // The render device is no longer alive; stop the worker thread.
-                return false;
-            }
-        } else if (kernErr != KERN_SUCCESS)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " semaphore_timedwait() error: %d", kernErr);
-        }
+OSStatus AudioDeviceMac::implInConverterProc(UInt32* numberDataPackets,
+                                             AudioBufferList* data) {
+  RTC_DCHECK(data->mNumberBuffers == 1);
+  PaRingBufferSize numSamples =
+      *numberDataPackets * _inStreamFormat.mChannelsPerFrame;
+
+  while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
+
+    kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
+      if (signal == 0) {
+        // The capture device is no longer alive; stop the worker thread.
+        *numberDataPackets = 0;
+        return 1;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " semaphore_wait() error: %d", kernErr);
     }
+  }
 
-    int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
-
-    if (!_ptrAudioBuffer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  capture AudioBuffer is invalid");
+  // Pass the read pointer directly to the converter to avoid a memcpy.
+  void* dummyPtr;
+  PaRingBufferSize dummySize;
+  PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
+                                  &data->mBuffers->mData, &numSamples,
+                                  &dummyPtr, &dummySize);
+  PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
+
+  data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
+  *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _inStreamFormat.mBytesPerPacket;
+
+  return 0;
+}
+
+bool AudioDeviceMac::RunRender(void* ptrThis) {
+  return static_cast<AudioDeviceMac*>(ptrThis)->RenderWorkerThread();
+}
+
+bool AudioDeviceMac::RenderWorkerThread() {
+  PaRingBufferSize numSamples =
+      ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
+  while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer) -
+             _renderDelayOffsetSamples <
+         numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
+
+    kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
+      if (signal == 0) {
+        // The render device is no longer alive; stop the worker thread.
         return false;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " semaphore_timedwait() error: %d", kernErr);
     }
+  }
 
-    // Ask for new PCM data to be played out using the AudioDeviceBuffer.
-    uint32_t nSamples =
-        _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
+  int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
 
-    nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
-    if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  invalid number of output samples(%d)", nSamples);
-    }
+  if (!_ptrAudioBuffer) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  capture AudioBuffer is invalid");
+    return false;
+  }
 
-    uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
-
-    SInt16 *pPlayBuffer = (SInt16 *) &playBuffer;
-    if (_macBookProPanRight && (_playChannels == 2))
-    {
-        // Mix entirely into the right channel and zero the left channel.
-        SInt32 sampleInt32 = 0;
-        for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx
-            += 2)
-        {
-            sampleInt32 = pPlayBuffer[sampleIdx];
-            sampleInt32 += pPlayBuffer[sampleIdx + 1];
-            sampleInt32 /= 2;
-
-            if (sampleInt32 > 32767)
-            {
-                sampleInt32 = 32767;
-            } else if (sampleInt32 < -32768)
-            {
-                sampleInt32 = -32768;
-            }
-
-            pPlayBuffer[sampleIdx] = 0;
-            pPlayBuffer[sampleIdx + 1] = static_cast<SInt16> (sampleInt32);
-        }
+  // Ask for new PCM data to be played out using the AudioDeviceBuffer.
+  uint32_t nSamples =
+      _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
+
+  nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+  if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  invalid number of output samples(%d)", nSamples);
+  }
+
+  uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
+
+  SInt16* pPlayBuffer = (SInt16*)&playBuffer;
+  if (_macBookProPanRight && (_playChannels == 2)) {
+    // Mix entirely into the right channel and zero the left channel.
+    SInt32 sampleInt32 = 0;
+    for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx += 2) {
+      sampleInt32 = pPlayBuffer[sampleIdx];
+      sampleInt32 += pPlayBuffer[sampleIdx + 1];
+      sampleInt32 /= 2;
+
+      if (sampleInt32 > 32767) {
+        sampleInt32 = 32767;
+      } else if (sampleInt32 < -32768) {
+        sampleInt32 = -32768;
+      }
+
+      pPlayBuffer[sampleIdx] = 0;
+      pPlayBuffer[sampleIdx + 1] = static_cast<SInt16>(sampleInt32);
     }
+  }
 
-    PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
+  PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
 
-    return true;
+  return true;
 }
 
-bool AudioDeviceMac::RunCapture(void* ptrThis)
-{
-    return static_cast<AudioDeviceMac*> (ptrThis)->CaptureWorkerThread();
+bool AudioDeviceMac::RunCapture(void* ptrThis) {
+  return static_cast<AudioDeviceMac*>(ptrThis)->CaptureWorkerThread();
 }
 
-bool AudioDeviceMac::CaptureWorkerThread()
-{
-    OSStatus err = noErr;
-    UInt32 noRecSamples = ENGINE_REC_BUF_SIZE_IN_SAMPLES
-        * _inDesiredFormat.mChannelsPerFrame;
-    SInt16 recordBuffer[noRecSamples];
-    UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
-
-    AudioBufferList engineBuffer;
-    engineBuffer.mNumberBuffers = 1; // Interleaved channels.
-    engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
-    engineBuffer.mBuffers->mDataByteSize = _inDesiredFormat.mBytesPerPacket
-        * noRecSamples;
-    engineBuffer.mBuffers->mData = recordBuffer;
-
-    err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
-                                          this, &size, &engineBuffer, NULL);
-    if (err != noErr)
-    {
-        if (err == 1)
-        {
-            // This is our own error.
-            return false;
-        } else
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioConverterFillComplexBuffer()",
-                     (const char *) &err);
-            return false;
-        }
+bool AudioDeviceMac::CaptureWorkerThread() {
+  OSStatus err = noErr;
+  UInt32 noRecSamples =
+      ENGINE_REC_BUF_SIZE_IN_SAMPLES * _inDesiredFormat.mChannelsPerFrame;
+  SInt16 recordBuffer[noRecSamples];
+  UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
+
+  AudioBufferList engineBuffer;
+  engineBuffer.mNumberBuffers = 1;  // Interleaved channels.
+  engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
+  engineBuffer.mBuffers->mDataByteSize =
+      _inDesiredFormat.mBytesPerPacket * noRecSamples;
+  engineBuffer.mBuffers->mData = recordBuffer;
+
+  err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
+                                        this, &size, &engineBuffer, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      return false;
+    } else {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+      return false;
     }
+  }
 
-    // TODO(xians): what if the returned size is incorrect?
-    if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES)
-    {
-        uint32_t currentMicLevel(0);
-        uint32_t newMicLevel(0);
-        int32_t msecOnPlaySide;
-        int32_t msecOnRecordSide;
-
-        int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
-        int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
-
-        msecOnPlaySide = static_cast<int32_t> (1e-3 * (renderDelayUs +
-                                                       _renderLatencyUs) + 0.5);
-        msecOnRecordSide = static_cast<int32_t> (1e-3 * (captureDelayUs +
-                                                         _captureLatencyUs) +
-                                                 0.5);
-
-        if (!_ptrAudioBuffer)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         "  capture AudioBuffer is invalid");
-            return false;
-        }
+  // TODO(xians): what if the returned size is incorrect?
+  if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES) {
+    uint32_t currentMicLevel(0);
+    uint32_t newMicLevel(0);
+    int32_t msecOnPlaySide;
+    int32_t msecOnRecordSide;
 
-        // store the recorded buffer (no action will be taken if the
-        // #recorded samples is not a full buffer)
-        _ptrAudioBuffer->SetRecordedBuffer((int8_t*) &recordBuffer,
-                                           (uint32_t) size);
-
-        if (AGC())
-        {
-            // store current mic level in the audio buffer if AGC is enabled
-            if (MicrophoneVolume(currentMicLevel) == 0)
-            {
-                // this call does not affect the actual microphone volume
-                _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
-            }
-        }
+    int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+    int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
 
-        _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
-
-        _ptrAudioBuffer->SetTypingStatus(KeyPressed());
-
-        // deliver recorded samples at specified sample rate, mic level etc.
-        // to the observer using callback
-        _ptrAudioBuffer->DeliverRecordedData();
-
-        if (AGC())
-        {
-            newMicLevel = _ptrAudioBuffer->NewMicLevel();
-            if (newMicLevel != 0)
-            {
-                // The VQE will only deliver non-zero microphone levels when
-                // a change is needed.
-                // Set this new mic level (received from the observer as return
-                // value in the callback).
-                WEBRTC_TRACE(kTraceStream, kTraceAudioDevice,
-                             _id, "  AGC change of volume: old=%u => new=%u",
-                             currentMicLevel, newMicLevel);
-                if (SetMicrophoneVolume(newMicLevel) == -1)
-                {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  the required modification of the microphone "
-                                 "volume failed");
-                }
-            }
+    msecOnPlaySide =
+        static_cast<int32_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+    msecOnRecordSide =
+        static_cast<int32_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
+
+    if (!_ptrAudioBuffer) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   "  capture AudioBuffer is invalid");
+      return false;
+    }
+
+    // store the recorded buffer (no action will be taken if the
+    // #recorded samples is not a full buffer)
+    _ptrAudioBuffer->SetRecordedBuffer((int8_t*)&recordBuffer, (uint32_t)size);
+
+    if (AGC()) {
+      // Use mod to ensure we check the volume on the first pass.
+      if (get_mic_volume_counter_ms_ % kGetMicVolumeIntervalMs == 0) {
+        get_mic_volume_counter_ms_ = 0;
+        // store current mic level in the audio buffer if AGC is enabled
+        if (MicrophoneVolume(currentMicLevel) == 0) {
+          // this call does not affect the actual microphone volume
+          _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
         }
+      }
+      get_mic_volume_counter_ms_ += kBufferSizeMs;
+    }
+
+    _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
+
+    _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+    // deliver recorded samples at specified sample rate, mic level etc.
+    // to the observer using callback
+    _ptrAudioBuffer->DeliverRecordedData();
+
+    if (AGC()) {
+      newMicLevel = _ptrAudioBuffer->NewMicLevel();
+      if (newMicLevel != 0) {
+        // The VQE will only deliver non-zero microphone levels when
+        // a change is needed.
+        // Set this new mic level (received from the observer as return
+        // value in the callback).
+        WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
+                     "  AGC change of volume: old=%u => new=%u",
+                     currentMicLevel, newMicLevel);
+        if (SetMicrophoneVolume(newMicLevel) == -1) {
+          WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                       "  the required modification of the microphone "
+                       "volume failed");
+        }
+      }
     }
+  }
 
-    return true;
+  return true;
 }
 
 bool AudioDeviceMac::KeyPressed() {
   bool key_down = false;
   // Loop through all Mac virtual key constant values.
-  for (unsigned int key_index = 0;
-                    key_index < arraysize(prev_key_state_);
-                    ++key_index) {
-    bool keyState = CGEventSourceKeyState(
-                             kCGEventSourceStateHIDSystemState,
-                             key_index);
+  for (unsigned int key_index = 0; key_index < arraysize(prev_key_state_);
+       ++key_index) {
+    bool keyState =
+        CGEventSourceKeyState(kCGEventSourceStateHIDSystemState, key_index);
     // A false -> true change in keymap means a key is pressed.
     key_down |= (keyState && !prev_key_state_[key_index]);
     // Save current state.
diff --git a/webrtc/modules/audio_device/mac/audio_device_mac.h b/webrtc/modules/audio_device/mac/audio_device_mac.h
index f2b66b4d06..ca3a51997d 100644
--- a/webrtc/modules/audio_device/mac/audio_device_mac.h
+++ b/webrtc/modules/audio_device/mac/audio_device_mac.h
@@ -23,350 +23,359 @@
 
 struct PaUtilRingBuffer;
 
-namespace webrtc
-{
+namespace rtc {
+class PlatformThread;
+}  // namespace rtc
+
+namespace webrtc {
 class EventWrapper;
-class ThreadWrapper;
 
 const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
 const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;
 
-const uint32_t N_REC_CHANNELS = 1; // default is mono recording
-const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout
+const uint32_t N_REC_CHANNELS = 1;   // default is mono recording
+const uint32_t N_PLAY_CHANNELS = 2;  // default is stereo playout
 const uint32_t N_DEVICE_CHANNELS = 64;
 
-const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100);
-const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);
+const int kBufferSizeMs = 10;
+
+const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES =
+    N_REC_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
+const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES =
+    N_PLAY_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
 
 const int N_BLOCKS_IO = 2;
-const int N_BUFFERS_IN = 2;  // Must be at least N_BLOCKS_IO.
+const int N_BUFFERS_IN = 2;   // Must be at least N_BLOCKS_IO.
 const int N_BUFFERS_OUT = 3;  // Must be at least N_BLOCKS_IO.
 
-const uint32_t TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
+const uint32_t TIMER_PERIOD_MS = 2 * 10 * N_BLOCKS_IO * 1000000;
 
 const uint32_t REC_BUF_SIZE_IN_SAMPLES =
     ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
 const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
     ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;
 
-class AudioDeviceMac: public AudioDeviceGeneric
-{
-public:
-    AudioDeviceMac(const int32_t id);
-    ~AudioDeviceMac();
-
-    // Retrieve the currently utilized audio layer
-    virtual int32_t
-        ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
-
-    // Main initializaton and termination
-    virtual int32_t Init();
-    virtual int32_t Terminate();
-    virtual bool Initialized() const;
-
-    // Device enumeration
-    virtual int16_t PlayoutDevices();
-    virtual int16_t RecordingDevices();
-    virtual int32_t PlayoutDeviceName(
-        uint16_t index,
-        char name[kAdmMaxDeviceNameSize],
-        char guid[kAdmMaxGuidSize]);
-    virtual int32_t RecordingDeviceName(
-        uint16_t index,
-        char name[kAdmMaxDeviceNameSize],
-        char guid[kAdmMaxGuidSize]);
-
-    // Device selection
-    virtual int32_t SetPlayoutDevice(uint16_t index);
-    virtual int32_t SetPlayoutDevice(
-        AudioDeviceModule::WindowsDeviceType device);
-    virtual int32_t SetRecordingDevice(uint16_t index);
-    virtual int32_t SetRecordingDevice(
-        AudioDeviceModule::WindowsDeviceType device);
-
-    // Audio transport initialization
-    virtual int32_t PlayoutIsAvailable(bool& available);
-    virtual int32_t InitPlayout();
-    virtual bool PlayoutIsInitialized() const;
-    virtual int32_t RecordingIsAvailable(bool& available);
-    virtual int32_t InitRecording();
-    virtual bool RecordingIsInitialized() const;
-
-    // Audio transport control
-    virtual int32_t StartPlayout();
-    virtual int32_t StopPlayout();
-    virtual bool Playing() const;
-    virtual int32_t StartRecording();
-    virtual int32_t StopRecording();
-    virtual bool Recording() const;
-
-    // Microphone Automatic Gain Control (AGC)
-    virtual int32_t SetAGC(bool enable);
-    virtual bool AGC() const;
-
-    // Volume control based on the Windows Wave API (Windows only)
-    virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
-    virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
-                                  uint16_t& volumeRight) const;
-
-    // Audio mixer initialization
-    virtual int32_t InitSpeaker();
-    virtual bool SpeakerIsInitialized() const;
-    virtual int32_t InitMicrophone();
-    virtual bool MicrophoneIsInitialized() const;
-
-    // Speaker volume controls
-    virtual int32_t SpeakerVolumeIsAvailable(bool& available);
-    virtual int32_t SetSpeakerVolume(uint32_t volume);
-    virtual int32_t SpeakerVolume(uint32_t& volume) const;
-    virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
-    virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
-    virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
-
-    // Microphone volume controls
-    virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
-    virtual int32_t SetMicrophoneVolume(uint32_t volume);
-    virtual int32_t MicrophoneVolume(uint32_t& volume) const;
-    virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
-    virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
-    virtual int32_t
-        MicrophoneVolumeStepSize(uint16_t& stepSize) const;
-
-    // Microphone mute control
-    virtual int32_t MicrophoneMuteIsAvailable(bool& available);
-    virtual int32_t SetMicrophoneMute(bool enable);
-    virtual int32_t MicrophoneMute(bool& enabled) const;
-
-    // Speaker mute control
-    virtual int32_t SpeakerMuteIsAvailable(bool& available);
-    virtual int32_t SetSpeakerMute(bool enable);
-    virtual int32_t SpeakerMute(bool& enabled) const;
-
-    // Microphone boost control
-    virtual int32_t MicrophoneBoostIsAvailable(bool& available);
-    virtual int32_t SetMicrophoneBoost(bool enable);
-    virtual int32_t MicrophoneBoost(bool& enabled) const;
-
-    // Stereo support
-    virtual int32_t StereoPlayoutIsAvailable(bool& available);
-    virtual int32_t SetStereoPlayout(bool enable);
-    virtual int32_t StereoPlayout(bool& enabled) const;
-    virtual int32_t StereoRecordingIsAvailable(bool& available);
-    virtual int32_t SetStereoRecording(bool enable);
-    virtual int32_t StereoRecording(bool& enabled) const;
-
-    // Delay information and control
-    virtual int32_t
-        SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
-                         uint16_t sizeMS);
-    virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
-                                  uint16_t& sizeMS) const;
-    virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
-    virtual int32_t RecordingDelay(uint16_t& delayMS) const;
-
-    // CPU load
-    virtual int32_t CPULoad(uint16_t& load) const;
-
-    virtual bool PlayoutWarning() const;
-    virtual bool PlayoutError() const;
-    virtual bool RecordingWarning() const;
-    virtual bool RecordingError() const;
-    virtual void ClearPlayoutWarning();
-    virtual void ClearPlayoutError();
-    virtual void ClearRecordingWarning();
-    virtual void ClearRecordingError();
-
-    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-private:
-    virtual int32_t MicrophoneIsAvailable(bool& available);
-    virtual int32_t SpeakerIsAvailable(bool& available);
-
-    static void AtomicSet32(int32_t* theValue, int32_t newValue);
-    static int32_t AtomicGet32(int32_t* theValue);
-
-    static void logCAMsg(const TraceLevel level,
-                         const TraceModule module,
-                         const int32_t id, const char *msg,
-                         const char *err);
-
-    int32_t GetNumberDevices(const AudioObjectPropertyScope scope,
-                             AudioDeviceID scopedDeviceIds[],
-                             const uint32_t deviceListLength);
-
-    int32_t GetDeviceName(const AudioObjectPropertyScope scope,
-                          const uint16_t index, char* name);
-
-    int32_t InitDevice(uint16_t userDeviceIndex,
-                       AudioDeviceID& deviceId, bool isInput);
-
-    // Always work with our preferred playout format inside VoE.
-    // Then convert the output to the OS setting using an AudioConverter.
-    OSStatus SetDesiredPlayoutFormat();
-
-    static OSStatus
-        objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
-                           const AudioObjectPropertyAddress addresses[],
-                           void* clientData);
-
-    OSStatus
-        implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
-                               const AudioObjectPropertyAddress addresses[]);
-
-    int32_t HandleDeviceChange();
-
-    int32_t
-        HandleStreamFormatChange(AudioObjectID objectId,
+const int kGetMicVolumeIntervalMs = 1000;
+
+class AudioDeviceMac : public AudioDeviceGeneric {
+ public:
+  AudioDeviceMac(const int32_t id);
+  ~AudioDeviceMac();
+
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const;
+
+  // Main initializaton and termination
+  virtual int32_t Init();
+  virtual int32_t Terminate();
+  virtual bool Initialized() const;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices();
+  virtual int16_t RecordingDevices();
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]);
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]);
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index);
+  virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+  virtual int32_t SetRecordingDevice(uint16_t index);
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device);
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available);
+  virtual int32_t InitPlayout();
+  virtual bool PlayoutIsInitialized() const;
+  virtual int32_t RecordingIsAvailable(bool& available);
+  virtual int32_t InitRecording();
+  virtual bool RecordingIsInitialized() const;
+
+  // Audio transport control
+  virtual int32_t StartPlayout();
+  virtual int32_t StopPlayout();
+  virtual bool Playing() const;
+  virtual int32_t StartRecording();
+  virtual int32_t StopRecording();
+  virtual bool Recording() const;
+
+  // Microphone Automatic Gain Control (AGC)
+  virtual int32_t SetAGC(bool enable);
+  virtual bool AGC() const;
+
+  // Volume control based on the Windows Wave API (Windows only)
+  virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
+  virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
+                                uint16_t& volumeRight) const;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker();
+  virtual bool SpeakerIsInitialized() const;
+  virtual int32_t InitMicrophone();
+  virtual bool MicrophoneIsInitialized() const;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available);
+  virtual int32_t SetSpeakerVolume(uint32_t volume);
+  virtual int32_t SpeakerVolume(uint32_t& volume) const;
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
+  virtual int32_t SetMicrophoneVolume(uint32_t volume);
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const;
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available);
+  virtual int32_t SetMicrophoneMute(bool enable);
+  virtual int32_t MicrophoneMute(bool& enabled) const;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available);
+  virtual int32_t SetSpeakerMute(bool enable);
+  virtual int32_t SpeakerMute(bool& enabled) const;
+
+  // Microphone boost control
+  virtual int32_t MicrophoneBoostIsAvailable(bool& available);
+  virtual int32_t SetMicrophoneBoost(bool enable);
+  virtual int32_t MicrophoneBoost(bool& enabled) const;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available);
+  virtual int32_t SetStereoPlayout(bool enable);
+  virtual int32_t StereoPlayout(bool& enabled) const;
+  virtual int32_t StereoRecordingIsAvailable(bool& available);
+  virtual int32_t SetStereoRecording(bool enable);
+  virtual int32_t StereoRecording(bool& enabled) const;
+
+  // Delay information and control
+  virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+                                   uint16_t sizeMS);
+  virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                uint16_t& sizeMS) const;
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
+  virtual int32_t RecordingDelay(uint16_t& delayMS) const;
+
+  // CPU load
+  virtual int32_t CPULoad(uint16_t& load) const;
+
+  virtual bool PlayoutWarning() const;
+  virtual bool PlayoutError() const;
+  virtual bool RecordingWarning() const;
+  virtual bool RecordingError() const;
+  virtual void ClearPlayoutWarning();
+  virtual void ClearPlayoutError();
+  virtual void ClearRecordingWarning();
+  virtual void ClearRecordingError();
+
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ private:
+  virtual int32_t MicrophoneIsAvailable(bool& available);
+  virtual int32_t SpeakerIsAvailable(bool& available);
+
+  static void AtomicSet32(int32_t* theValue, int32_t newValue);
+  static int32_t AtomicGet32(int32_t* theValue);
+
+  static void logCAMsg(const TraceLevel level,
+                       const TraceModule module,
+                       const int32_t id,
+                       const char* msg,
+                       const char* err);
+
+  int32_t GetNumberDevices(const AudioObjectPropertyScope scope,
+                           AudioDeviceID scopedDeviceIds[],
+                           const uint32_t deviceListLength);
+
+  int32_t GetDeviceName(const AudioObjectPropertyScope scope,
+                        const uint16_t index,
+                        char* name);
+
+  int32_t InitDevice(uint16_t userDeviceIndex,
+                     AudioDeviceID& deviceId,
+                     bool isInput);
+
+  // Always work with our preferred playout format inside VoE.
+  // Then convert the output to the OS setting using an AudioConverter.
+  OSStatus SetDesiredPlayoutFormat();
+
+  static OSStatus objectListenerProc(
+      AudioObjectID objectId,
+      UInt32 numberAddresses,
+      const AudioObjectPropertyAddress addresses[],
+      void* clientData);
+
+  OSStatus implObjectListenerProc(AudioObjectID objectId,
+                                  UInt32 numberAddresses,
+                                  const AudioObjectPropertyAddress addresses[]);
+
+  int32_t HandleDeviceChange();
+
+  int32_t HandleStreamFormatChange(AudioObjectID objectId,
+                                   AudioObjectPropertyAddress propertyAddress);
+
+  int32_t HandleDataSourceChange(AudioObjectID objectId,
                                  AudioObjectPropertyAddress propertyAddress);
 
-    int32_t
-        HandleDataSourceChange(AudioObjectID objectId,
-                               AudioObjectPropertyAddress propertyAddress);
-
-    int32_t
-        HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
-
-    static OSStatus deviceIOProc(AudioDeviceID device,
-                                 const AudioTimeStamp *now,
-                                 const AudioBufferList *inputData,
-                                 const AudioTimeStamp *inputTime,
-                                 AudioBufferList *outputData,
+  int32_t HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
+
+  static OSStatus deviceIOProc(AudioDeviceID device,
+                               const AudioTimeStamp* now,
+                               const AudioBufferList* inputData,
+                               const AudioTimeStamp* inputTime,
+                               AudioBufferList* outputData,
+                               const AudioTimeStamp* outputTime,
+                               void* clientData);
+
+  static OSStatus outConverterProc(
+      AudioConverterRef audioConverter,
+      UInt32* numberDataPackets,
+      AudioBufferList* data,
+      AudioStreamPacketDescription** dataPacketDescription,
+      void* userData);
+
+  static OSStatus inDeviceIOProc(AudioDeviceID device,
+                                 const AudioTimeStamp* now,
+                                 const AudioBufferList* inputData,
+                                 const AudioTimeStamp* inputTime,
+                                 AudioBufferList* outputData,
                                  const AudioTimeStamp* outputTime,
-                                 void *clientData);
+                                 void* clientData);
 
-    static OSStatus
-        outConverterProc(AudioConverterRef audioConverter,
-                         UInt32 *numberDataPackets, AudioBufferList *data,
-                         AudioStreamPacketDescription **dataPacketDescription,
-                         void *userData);
+  static OSStatus inConverterProc(
+      AudioConverterRef audioConverter,
+      UInt32* numberDataPackets,
+      AudioBufferList* data,
+      AudioStreamPacketDescription** dataPacketDescription,
+      void* inUserData);
 
-    static OSStatus inDeviceIOProc(AudioDeviceID device,
-                                   const AudioTimeStamp *now,
-                                   const AudioBufferList *inputData,
-                                   const AudioTimeStamp *inputTime,
-                                   AudioBufferList *outputData,
-                                   const AudioTimeStamp *outputTime,
-                                   void *clientData);
+  OSStatus implDeviceIOProc(const AudioBufferList* inputData,
+                            const AudioTimeStamp* inputTime,
+                            AudioBufferList* outputData,
+                            const AudioTimeStamp* outputTime);
 
-    static OSStatus
-        inConverterProc(AudioConverterRef audioConverter,
-                        UInt32 *numberDataPackets, AudioBufferList *data,
-                        AudioStreamPacketDescription **dataPacketDescription,
-                        void *inUserData);
+  OSStatus implOutConverterProc(UInt32* numberDataPackets,
+                                AudioBufferList* data);
 
-    OSStatus implDeviceIOProc(const AudioBufferList *inputData,
-                              const AudioTimeStamp *inputTime,
-                              AudioBufferList *outputData,
-                              const AudioTimeStamp *outputTime);
+  OSStatus implInDeviceIOProc(const AudioBufferList* inputData,
+                              const AudioTimeStamp* inputTime);
 
-    OSStatus implOutConverterProc(UInt32 *numberDataPackets,
-                                  AudioBufferList *data);
+  OSStatus implInConverterProc(UInt32* numberDataPackets,
+                               AudioBufferList* data);
 
-    OSStatus implInDeviceIOProc(const AudioBufferList *inputData,
-                                const AudioTimeStamp *inputTime);
+  static bool RunCapture(void*);
+  static bool RunRender(void*);
+  bool CaptureWorkerThread();
+  bool RenderWorkerThread();
 
-    OSStatus implInConverterProc(UInt32 *numberDataPackets,
-                                 AudioBufferList *data);
+  bool KeyPressed();
 
-    static bool RunCapture(void*);
-    static bool RunRender(void*);
-    bool CaptureWorkerThread();
-    bool RenderWorkerThread();
+  AudioDeviceBuffer* _ptrAudioBuffer;
 
-    bool KeyPressed();
+  CriticalSectionWrapper& _critSect;
 
-    AudioDeviceBuffer* _ptrAudioBuffer;
+  EventWrapper& _stopEventRec;
+  EventWrapper& _stopEvent;
 
-    CriticalSectionWrapper& _critSect;
+  // TODO(pbos): Replace with direct members, just start/stop, no need to
+  // recreate the thread.
+  // Only valid/running between calls to StartRecording and StopRecording.
+  rtc::scoped_ptr<rtc::PlatformThread> capture_worker_thread_;
 
-    EventWrapper& _stopEventRec;
-    EventWrapper& _stopEvent;
+  // Only valid/running between calls to StartPlayout and StopPlayout.
+  rtc::scoped_ptr<rtc::PlatformThread> render_worker_thread_;
 
-    // Only valid/running between calls to StartRecording and StopRecording.
-    rtc::scoped_ptr<ThreadWrapper> capture_worker_thread_;
+  int32_t _id;
 
-    // Only valid/running between calls to StartPlayout and StopPlayout.
-    rtc::scoped_ptr<ThreadWrapper> render_worker_thread_;
+  AudioMixerManagerMac _mixerManager;
 
-    int32_t _id;
-
-    AudioMixerManagerMac _mixerManager;
-
-    uint16_t _inputDeviceIndex;
-    uint16_t _outputDeviceIndex;
-    AudioDeviceID _inputDeviceID;
-    AudioDeviceID _outputDeviceID;
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  AudioDeviceID _inputDeviceID;
+  AudioDeviceID _outputDeviceID;
 #if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
-    AudioDeviceIOProcID _inDeviceIOProcID;
-    AudioDeviceIOProcID _deviceIOProcID;
+  AudioDeviceIOProcID _inDeviceIOProcID;
+  AudioDeviceIOProcID _deviceIOProcID;
 #endif
-    bool _inputDeviceIsSpecified;
-    bool _outputDeviceIsSpecified;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  uint8_t _recChannels;
+  uint8_t _playChannels;
 
-    uint8_t _recChannels;
-    uint8_t _playChannels;
+  Float32* _captureBufData;
+  SInt16* _renderBufData;
 
-    Float32* _captureBufData;
-    SInt16* _renderBufData;
+  SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
 
-    SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
+  AudioDeviceModule::BufferType _playBufType;
 
-    AudioDeviceModule::BufferType _playBufType;
+  bool _initialized;
+  bool _isShutDown;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+  bool _AGC;
 
-    bool _initialized;
-    bool _isShutDown;
-    bool _recording;
-    bool _playing;
-    bool _recIsInitialized;
-    bool _playIsInitialized;
-    bool _AGC;
+  // Atomically set varaibles
+  int32_t _renderDeviceIsAlive;
+  int32_t _captureDeviceIsAlive;
 
-    // Atomically set varaibles
-    int32_t _renderDeviceIsAlive;
-    int32_t _captureDeviceIsAlive;
+  bool _twoDevices;
+  bool _doStop;  // For play if not shared device or play+rec if shared device
+  bool _doStopRec;  // For rec if not shared device
+  bool _macBookPro;
+  bool _macBookProPanRight;
 
-    bool _twoDevices;
-    bool _doStop; // For play if not shared device or play+rec if shared device
-    bool _doStopRec; // For rec if not shared device
-    bool _macBookPro;
-    bool _macBookProPanRight;
+  AudioConverterRef _captureConverter;
+  AudioConverterRef _renderConverter;
 
-    AudioConverterRef _captureConverter;
-    AudioConverterRef _renderConverter;
+  AudioStreamBasicDescription _outStreamFormat;
+  AudioStreamBasicDescription _outDesiredFormat;
+  AudioStreamBasicDescription _inStreamFormat;
+  AudioStreamBasicDescription _inDesiredFormat;
 
-    AudioStreamBasicDescription _outStreamFormat;
-    AudioStreamBasicDescription _outDesiredFormat;
-    AudioStreamBasicDescription _inStreamFormat;
-    AudioStreamBasicDescription _inDesiredFormat;
+  uint32_t _captureLatencyUs;
+  uint32_t _renderLatencyUs;
 
-    uint32_t _captureLatencyUs;
-    uint32_t _renderLatencyUs;
+  // Atomically set variables
+  mutable int32_t _captureDelayUs;
+  mutable int32_t _renderDelayUs;
 
-    // Atomically set variables
-    mutable int32_t _captureDelayUs;
-    mutable int32_t _renderDelayUs;
+  int32_t _renderDelayOffsetSamples;
 
-    int32_t _renderDelayOffsetSamples;
+  uint16_t _playBufDelayFixed;  // fixed playback delay
 
-    uint16_t _playBufDelayFixed; // fixed playback delay
+  uint16_t _playWarning;
+  uint16_t _playError;
+  uint16_t _recWarning;
+  uint16_t _recError;
 
-    uint16_t _playWarning;
-    uint16_t _playError;
-    uint16_t _recWarning;
-    uint16_t _recError;
+  PaUtilRingBuffer* _paCaptureBuffer;
+  PaUtilRingBuffer* _paRenderBuffer;
 
-    PaUtilRingBuffer* _paCaptureBuffer;
-    PaUtilRingBuffer* _paRenderBuffer;
+  semaphore_t _renderSemaphore;
+  semaphore_t _captureSemaphore;
 
-    semaphore_t _renderSemaphore;
-    semaphore_t _captureSemaphore;
+  int _captureBufSizeSamples;
+  int _renderBufSizeSamples;
 
-    int _captureBufSizeSamples;
-    int _renderBufSizeSamples;
+  // Typing detection
+  // 0x5c is key "9", after that comes function keys.
+  bool prev_key_state_[0x5d];
 
-    // Typing detection
-    // 0x5c is key "9", after that comes function keys.
-    bool prev_key_state_[0x5d];
+  int get_mic_volume_counter_ms_;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
index 9016ffe508..e7e0754695 100644
--- a/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
+++ b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
@@ -11,1134 +11,989 @@
 #include "webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
-#include <unistd.h>             // getpid()
+#include <unistd.h>  // getpid()
 
 namespace webrtc {
-	
-#define WEBRTC_CA_RETURN_ON_ERR(expr)                                     \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,    \
-                "Error in " #expr, (const char *)&err);                 \
-            return -1;                                                  \
-        }                                                               \
-    } while(0)
-
-#define WEBRTC_CA_LOG_ERR(expr)                                           \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,    \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
-
-#define WEBRTC_CA_LOG_WARN(expr)                                           \
-    do {                                                                 \
-        err = expr;                                                      \
-        if (err != noErr) {                                              \
-            logCAMsg(kTraceWarning, kTraceAudioDevice, _id,  \
-                "Error in " #expr, (const char *)&err);                  \
-        }                                                                \
-    } while(0)
-
-AudioMixerManagerMac::AudioMixerManagerMac(const int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id),
-    _inputDeviceID(kAudioObjectUnknown),
-    _outputDeviceID(kAudioObjectUnknown),
-    _noInputChannels(0),
-    _noOutputChannels(0)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s constructed", __FUNCTION__);
-}
-
-AudioMixerManagerMac::~AudioMixerManagerMac()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destructed", __FUNCTION__);
 
-    Close();
-
-    delete &_critSect;
+#define WEBRTC_CA_RETURN_ON_ERR(expr)                                  \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+      return -1;                                                       \
+    }                                                                  \
+  } while (0)
+
+#define WEBRTC_CA_LOG_ERR(expr)                                        \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+    }                                                                  \
+  } while (0)
+
+#define WEBRTC_CA_LOG_WARN(expr)                                         \
+  do {                                                                   \
+    err = expr;                                                          \
+    if (err != noErr) {                                                  \
+      logCAMsg(kTraceWarning, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                     \
+    }                                                                    \
+  } while (0)
+
+AudioMixerManagerMac::AudioMixerManagerMac(const int32_t id)
+    : _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      _id(id),
+      _inputDeviceID(kAudioObjectUnknown),
+      _outputDeviceID(kAudioObjectUnknown),
+      _noInputChannels(0),
+      _noOutputChannels(0) {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s constructed",
+               __FUNCTION__);
+}
+
+AudioMixerManagerMac::~AudioMixerManagerMac() {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destructed",
+               __FUNCTION__);
+
+  Close();
+
+  delete &_critSect;
 }
 
 // ============================================================================
 //	                                PUBLIC METHODS
 // ============================================================================
 
-int32_t AudioMixerManagerMac::Close()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+int32_t AudioMixerManagerMac::Close() {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    CloseSpeaker();
-    CloseMicrophone();
-
-    return 0;
+  CloseSpeaker();
+  CloseMicrophone();
 
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::CloseSpeaker()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+int32_t AudioMixerManagerMac::CloseSpeaker() {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    _outputDeviceID = kAudioObjectUnknown;
-    _noOutputChannels = 0;
+  _outputDeviceID = kAudioObjectUnknown;
+  _noOutputChannels = 0;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::CloseMicrophone()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+int32_t AudioMixerManagerMac::CloseMicrophone() {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    _inputDeviceID = kAudioObjectUnknown;
-    _noInputChannels = 0;
+  _inputDeviceID = kAudioObjectUnknown;
+  _noInputChannels = 0;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::OpenSpeaker(AudioDeviceID deviceID)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::OpenSpeaker(id=%d)", deviceID);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    pid_t hogPid = -1;
+int32_t AudioMixerManagerMac::OpenSpeaker(AudioDeviceID deviceID) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::OpenSpeaker(id=%d)", deviceID);
 
-    _outputDeviceID = deviceID;
+  CriticalSectionScoped lock(&_critSect);
 
-    // Check which process, if any, has hogged the device. 
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
-            kAudioDevicePropertyScopeOutput, 0 };
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  pid_t hogPid = -1;
 
-    size = sizeof(hogPid);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-            &propertyAddress, 0, NULL, &size, &hogPid));
-
-    if (hogPid == -1)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " No process has hogged the input device");
-    }
-    // getpid() is apparently "always successful"
-    else if (hogPid == getpid())
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Our process has hogged the input device");
-    } else
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Another process (pid = %d) has hogged the input device",
-                     static_cast<int> (hogPid));
-
-        return -1;
-    }
+  _outputDeviceID = deviceID;
 
-    // get number of channels from stream format
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  // Check which process, if any, has hogged the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyHogMode, kAudioDevicePropertyScopeOutput, 0};
 
-    // Get the stream format, to be able to read the number of channels.
-    AudioStreamBasicDescription streamFormat;
-    size = sizeof(AudioStreamBasicDescription);
-    memset(&streamFormat, 0, size);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-            &propertyAddress, 0, NULL, &size, &streamFormat));
+  size = sizeof(hogPid);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &hogPid));
 
-    _noOutputChannels = streamFormat.mChannelsPerFrame;
-
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::OpenMicrophone(AudioDeviceID deviceID)
-{
+  if (hogPid == -1) {
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::OpenMicrophone(id=%d)", deviceID);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    pid_t hogPid = -1;
-
-    _inputDeviceID = deviceID;
-
-    // Check which process, if any, has hogged the device. 
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
-            kAudioDevicePropertyScopeInput, 0 };
-    size = sizeof(hogPid);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &hogPid));
-    if (hogPid == -1)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " No process has hogged the input device");
-    }
-    // getpid() is apparently "always successful"
-    else if (hogPid == getpid())
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Our process has hogged the input device");
-    } else
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Another process (pid = %d) has hogged the input device",
-                     static_cast<int> (hogPid));
-
-        return -1;
-    }
-
-    // get number of channels from stream format
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-
-    // Get the stream format, to be able to read the number of channels.
-    AudioStreamBasicDescription streamFormat;
-    size = sizeof(AudioStreamBasicDescription);
-    memset(&streamFormat, 0, size);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &streamFormat));
-
-    _noInputChannels = streamFormat.mChannelsPerFrame;
+                 " No process has hogged the input device");
+  }
+  // getpid() is apparently "always successful"
+  else if (hogPid == getpid()) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 " Our process has hogged the input device");
+  } else {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Another process (pid = %d) has hogged the input device",
+                 static_cast<int>(hogPid));
 
-    return 0;
-}
+    return -1;
+  }
 
-bool AudioMixerManagerMac::SpeakerIsInitialized() const
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+  // get number of channels from stream format
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
 
-    return (_outputDeviceID != kAudioObjectUnknown);
-}
+  // Get the stream format, to be able to read the number of channels.
+  AudioStreamBasicDescription streamFormat;
+  size = sizeof(AudioStreamBasicDescription);
+  memset(&streamFormat, 0, size);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &streamFormat));
 
-bool AudioMixerManagerMac::MicrophoneIsInitialized() const
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+  _noOutputChannels = streamFormat.mChannelsPerFrame;
 
-    return (_inputDeviceID != kAudioObjectUnknown);
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::SetSpeakerVolume(uint32_t volume)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetSpeakerVolume(volume=%u)", volume);
+int32_t AudioMixerManagerMac::OpenMicrophone(AudioDeviceID deviceID) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::OpenMicrophone(id=%d)", deviceID);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  pid_t hogPid = -1;
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    bool success = false;
+  _inputDeviceID = deviceID;
 
-    // volume range is 0.0 - 1.0, convert from 0 -255
-    const Float32 vol = (Float32)(volume / 255.0);
+  // Check which process, if any, has hogged the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyHogMode, kAudioDevicePropertyScopeInput, 0};
+  size = sizeof(hogPid);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &hogPid));
+  if (hogPid == -1) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 " No process has hogged the input device");
+  }
+  // getpid() is apparently "always successful"
+  else if (hogPid == getpid()) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 " Our process has hogged the input device");
+  } else {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Another process (pid = %d) has hogged the input device",
+                 static_cast<int>(hogPid));
 
-    assert(vol <= 1.0 && vol >= 0.0);
+    return -1;
+  }
 
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
-            0 };
-    Boolean isSettable = false;
-    err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                        &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(vol);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, size, &vol));
+  // get number of channels from stream format
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
 
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(vol);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                    &propertyAddress, 0, NULL, size, &vol));
-        }
-        success = true;
-    }
+  // Get the stream format, to be able to read the number of channels.
+  AudioStreamBasicDescription streamFormat;
+  size = sizeof(AudioStreamBasicDescription);
+  memset(&streamFormat, 0, size);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &streamFormat));
 
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set a volume on any output channel");
-        return -1;
-    }
+  _noInputChannels = streamFormat.mChannelsPerFrame;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::SpeakerVolume(uint32_t& volume) const
-{
+bool AudioMixerManagerMac::SpeakerIsInitialized() const {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    Float32 channelVol = 0;
-    Float32 vol = 0;
-
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
-            0 };
-    Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(vol);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, &size, &vol));
-
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> (vol * 255 + 0.5);
-    } else
-    {
-        // Otherwise get the average volume across channels.
-        vol = 0;
-        for (UInt32 i = 1; i <= _noOutputChannels; i++)
-        {
-            channelVol = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelVol);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelVol));
-
-                vol += channelVol;
-                channels++;
-            }
-        }
-
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get a volume on any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> (255 * vol / channels + 0.5);
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::SpeakerVolume() => vol=%i", vol);
-
-    return 0;
+  return (_outputDeviceID != kAudioObjectUnknown);
 }
 
-int32_t
-AudioMixerManagerMac::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
+bool AudioMixerManagerMac::MicrophoneIsInitialized() const {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    maxVolume = 255;
-
-    return 0;
+  return (_inputDeviceID != kAudioObjectUnknown);
 }
 
-int32_t
-AudioMixerManagerMac::MinSpeakerVolume(uint32_t& minVolume) const
-{
+int32_t AudioMixerManagerMac::SetSpeakerVolume(uint32_t volume) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetSpeakerVolume(volume=%u)", volume);
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  CriticalSectionScoped lock(&_critSect);
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    minVolume = 0;
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    return 0;
-}
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  bool success = false;
 
-int32_t
-AudioMixerManagerMac::SpeakerVolumeStepSize(uint16_t& stepSize) const
-{
+  // volume range is 0.0 - 1.0, convert from 0 -255
+  const Float32 vol = (Float32)(volume / 255.0);
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  assert(vol <= 1.0 && vol >= 0.0);
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    stepSize = 1;
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, size, &vol));
 
     return 0;
-}
+  }
 
-int32_t AudioMixerManagerMac::SpeakerVolumeIsAvailable(bool& available)
-{
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
-            0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Volume cannot be set for output channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
-    }
-
+    if (err == noErr && isSettable) {
+      size = sizeof(vol);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, size, &vol));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set a volume on any output channel");
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerVolume(uint32_t& volume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  Float32 channelVol = 0;
+  Float32 vol = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, &size, &vol));
+
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(vol * 255 + 0.5);
+  } else {
+    // Otherwise get the average volume across channels.
+    vol = 0;
+    for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+      channelVol = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelVol);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _outputDeviceID, &propertyAddress, 0, NULL, &size, &channelVol));
+
+        vol += channelVol;
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get a volume on any channel");
+      return -1;
+    }
+
+    assert(channels > 0);
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(255 * vol / channels + 0.5);
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "     AudioMixerManagerMac::SpeakerVolume() => vol=%i", vol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  maxVolume = 255;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MinSpeakerVolume(uint32_t& minVolume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  minVolume = 0;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerVolumeStepSize(uint16_t& stepSize) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  stepSize = 1;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerVolumeIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
     available = true;
     return 0;
-}
-
-int32_t AudioMixerManagerMac::SpeakerMuteIsAvailable(bool& available)
-{
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  }
 
-    OSStatus err = noErr;
-
-    // Does the capture device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeOutput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Mute cannot be set for output channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
-    }
-
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Volume cannot be set for output channel %d, err=%d", i,
+                   err);
+      return -1;
+    }
+  }
+
+  available = true;
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerMuteIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
     available = true;
     return 0;
-}
-
-int32_t AudioMixerManagerMac::SetSpeakerMute(bool enable)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetSpeakerMute(enable=%u)", enable);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    UInt32 mute = enable ? 1 : 0;
-    bool success = false;
+  }
 
-    // Does the render device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeOutput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(mute);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, size, &mute));
-
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(mute);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                    &propertyAddress, 0, NULL, size, &mute));
-        }
-        success = true;
-    }
-
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set mute on any input channel");
-        return -1;
-    }
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Mute cannot be set for output channel %d, err=%d", i, err);
+      return -1;
+    }
+  }
+
+  available = true;
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SetSpeakerMute(bool enable) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetSpeakerMute(enable=%u)", enable);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  UInt32 mute = enable ? 1 : 0;
+  bool success = false;
+
+  // Does the render device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(mute);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, size, &mute));
 
     return 0;
-}
+  }
 
-int32_t AudioMixerManagerMac::SpeakerMute(bool& enabled) const
-{
-
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    UInt32 channelMuted = 0;
-    UInt32 muted = 0;
-
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeOutput, 0 };
-    Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(muted);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, &size, &muted));
-
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
-    } else
-    {
-        // Otherwise check if all channels are muted.
-        for (UInt32 i = 1; i <= _noOutputChannels; i++)
-        {
-            muted = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelMuted);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelMuted));
-
-                muted = (muted && channelMuted);
-                channels++;
-            }
-        }
-
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get mute for any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::SpeakerMute() => enabled=%d, enabled");
-
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::StereoPlayoutIsAvailable(bool& available)
-{
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    available = (_noOutputChannels == 2);
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::StereoRecordingIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    available = (_noInputChannels == 2);
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::MicrophoneMuteIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-
-    // Does the capture device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
-    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Mute cannot be set for output channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
-    }
-
+    if (err == noErr && isSettable) {
+      size = sizeof(mute);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set mute on any input channel");
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerMute(bool& enabled) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  UInt32 channelMuted = 0;
+  UInt32 muted = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(muted);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, &size, &muted));
+
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  } else {
+    // Otherwise check if all channels are muted.
+    for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+      muted = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelMuted);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _outputDeviceID, &propertyAddress, 0, NULL, &size, &channelMuted));
+
+        muted = (muted && channelMuted);
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get mute for any channel");
+      return -1;
+    }
+
+    assert(channels > 0);
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  }
+
+  WEBRTC_TRACE(
+      kTraceInfo, kTraceAudioDevice, _id,
+      "     AudioMixerManagerMac::SpeakerMute() => enabled=%d, enabled");
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::StereoPlayoutIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  available = (_noOutputChannels == 2);
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::StereoRecordingIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  available = (_noInputChannels == 2);
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneMuteIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
     available = true;
     return 0;
-}
+  }
 
-int32_t AudioMixerManagerMac::SetMicrophoneMute(bool enable)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetMicrophoneMute(enable=%u)", enable);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    UInt32 mute = enable ? 1 : 0;
-    bool success = false;
-
-    // Does the capture device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(mute);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, size, &mute));
-
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(mute);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                    &propertyAddress, 0, NULL, size, &mute));
-        }
-        success = true;
-    }
-
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set mute on any input channel");
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::MicrophoneMute(bool& enabled) const
-{
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    UInt32 channelMuted = 0;
-    UInt32 muted = 0;
-
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeInput, 0 };
-    Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(muted);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, &size, &muted));
-
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
-    } else
-    {
-        // Otherwise check if all channels are muted.
-        for (UInt32 i = 1; i <= _noInputChannels; i++)
-        {
-            muted = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelMuted);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelMuted));
-
-                muted = (muted && channelMuted);
-                channels++;
-            }
-        }
-
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get mute for any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::MicrophoneMute() => enabled=%d",
-                 enabled);
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Mute cannot be set for output channel %d, err=%d", i, err);
+      return -1;
+    }
+  }
+
+  available = true;
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SetMicrophoneMute(bool enable) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetMicrophoneMute(enable=%u)", enable);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  UInt32 mute = enable ? 1 : 0;
+  bool success = false;
+
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(mute);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, size, &mute));
 
     return 0;
-}
-
-int32_t AudioMixerManagerMac::MicrophoneBoostIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    available = false; // No AudioObjectPropertySelector value for Mic Boost
+  }
 
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::SetMicrophoneBoost(bool enable)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetMicrophoneBoost(enable=%u)", enable);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    // Ensure that the selected microphone has a valid boost control.
-    bool available(false);
-    MicrophoneBoostIsAvailable(available);
-    if (!available)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  it is not possible to enable microphone boost");
-        return -1;
-    }
-
-    // It is assumed that the call above fails!
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::MicrophoneBoost(bool& enabled) const
-{
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    // Microphone boost cannot be enabled on this platform!
-    enabled = false;
-
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::MicrophoneVolumeIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyVolumeScalar,
-                kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Volume cannot be set for input channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
-    }
-
+    if (err == noErr && isSettable) {
+      size = sizeof(mute);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _inputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set mute on any input channel");
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneMute(bool& enabled) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  UInt32 channelMuted = 0;
+  UInt32 muted = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(muted);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, &size, &muted));
+
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  } else {
+    // Otherwise check if all channels are muted.
+    for (UInt32 i = 1; i <= _noInputChannels; i++) {
+      muted = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelMuted);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _inputDeviceID, &propertyAddress, 0, NULL, &size, &channelMuted));
+
+        muted = (muted && channelMuted);
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get mute for any channel");
+      return -1;
+    }
+
+    assert(channels > 0);
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "     AudioMixerManagerMac::MicrophoneMute() => enabled=%d",
+               enabled);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneBoostIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  available = false;  // No AudioObjectPropertySelector value for Mic Boost
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SetMicrophoneBoost(bool enable) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetMicrophoneBoost(enable=%u)", enable);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // Ensure that the selected microphone has a valid boost control.
+  bool available(false);
+  MicrophoneBoostIsAvailable(available);
+  if (!available) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  it is not possible to enable microphone boost");
+    return -1;
+  }
+
+  // It is assumed that the call above fails!
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneBoost(bool& enabled) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // Microphone boost cannot be enabled on this platform!
+  enabled = false;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneVolumeIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
     available = true;
     return 0;
-}
-
-int32_t AudioMixerManagerMac::SetMicrophoneVolume(uint32_t volume)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetMicrophoneVolume(volume=%u)", volume);
-
-    CriticalSectionScoped lock(&_critSect);
+  }
 
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    bool success = false;
-
-    // volume range is 0.0 - 1.0, convert from 0 - 255
-    const Float32 vol = (Float32)(volume / 255.0);
-
-    assert(vol <= 1.0 && vol >= 0.0);
-
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyVolumeScalar,
-                kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(vol);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, size, &vol));
-
-        return 0;
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Volume cannot be set for input channel %d, err=%d", i,
+                   err);
+      return -1;
     }
+  }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(vol);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                    &propertyAddress, 0, NULL, size, &vol));
-        }
-        success = true;
-    }
-
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set a level on any input channel");
-        return -1;
-    }
-
-    return 0;
+  available = true;
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MicrophoneVolume(uint32_t& volume) const
-{
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+int32_t AudioMixerManagerMac::SetMicrophoneVolume(uint32_t volume) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetMicrophoneVolume(volume=%u)", volume);
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    Float32 channelVol = 0;
-    Float32 volFloat32 = 0;
-
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyVolumeScalar,
-                kAudioDevicePropertyScopeInput, 0 };
-    Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(volFloat32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, &size, &volFloat32));
-
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> (volFloat32 * 255 + 0.5);
-    } else
-    {
-        // Otherwise get the average volume across channels.
-        volFloat32 = 0;
-        for (UInt32 i = 1; i <= _noInputChannels; i++)
-        {
-            channelVol = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelVol);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelVol));
-
-                volFloat32 += channelVol;
-                channels++;
-            }
-        }
-
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get a level on any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> 
-            (255 * volFloat32 / channels + 0.5);
-    }
+  CriticalSectionScoped lock(&_critSect);
 
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::MicrophoneVolume() => vol=%u",
-                 volume);
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    return 0;
-}
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  bool success = false;
 
-int32_t
-AudioMixerManagerMac::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
+  // volume range is 0.0 - 1.0, convert from 0 - 255
+  const Float32 vol = (Float32)(volume / 255.0);
 
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  assert(vol <= 1.0 && vol >= 0.0);
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    maxVolume = 255;
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, size, &vol));
 
     return 0;
-}
-
-int32_t
-AudioMixerManagerMac::MinMicrophoneVolume(uint32_t& minVolume) const
-{
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  }
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 10
-    minVolume = 0;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err == noErr && isSettable) {
+      size = sizeof(vol);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _inputDeviceID, &propertyAddress, 0, NULL, size, &vol));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set a level on any input channel");
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneVolume(uint32_t& volume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  Float32 channelVol = 0;
+  Float32 volFloat32 = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(volFloat32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, &size, &volFloat32));
+
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(volFloat32 * 255 + 0.5);
+  } else {
+    // Otherwise get the average volume across channels.
+    volFloat32 = 0;
+    for (UInt32 i = 1; i <= _noInputChannels; i++) {
+      channelVol = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelVol);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _inputDeviceID, &propertyAddress, 0, NULL, &size, &channelVol));
+
+        volFloat32 += channelVol;
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get a level on any channel");
+      return -1;
+    }
+
+    assert(channels > 0);
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(255 * volFloat32 / channels + 0.5);
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "     AudioMixerManagerMac::MicrophoneVolume() => vol=%u",
+               volume);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  maxVolume = 255;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MinMicrophoneVolume(uint32_t& minVolume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 10
+  minVolume = 0;
 
-    return 0;
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
-{
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 10
-    stepSize = 1;
-
-    return 0;
+int32_t AudioMixerManagerMac::MicrophoneVolumeStepSize(
+    uint16_t& stepSize) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 10
+  stepSize = 1;
+
+  return 0;
 }
 
 // ============================================================================
@@ -1148,18 +1003,18 @@ AudioMixerManagerMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
 // CoreAudio errors are best interpreted as four character strings.
 void AudioMixerManagerMac::logCAMsg(const TraceLevel level,
                                     const TraceModule module,
-                                    const int32_t id, const char *msg,
-                                    const char *err)
-{
-    assert(msg != NULL);
-    assert(err != NULL);
+                                    const int32_t id,
+                                    const char* msg,
+                                    const char* err) {
+  assert(msg != NULL);
+  assert(err != NULL);
 
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
+  WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
 #else
-    // We need to flip the characters in this case.
-    WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
-        + 2, err + 1, err);
+  // We need to flip the characters in this case.
+  WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err + 2,
+               err + 1, err);
 #endif
 }
 
diff --git a/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
index 94d6928921..9cbfe2deb4 100644
--- a/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
+++ b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
@@ -18,63 +18,62 @@
 #include <CoreAudio/CoreAudio.h>
 
 namespace webrtc {
-	
-class AudioMixerManagerMac
-{
-public:
-    int32_t OpenSpeaker(AudioDeviceID deviceID);
-    int32_t OpenMicrophone(AudioDeviceID deviceID);
-    int32_t SetSpeakerVolume(uint32_t volume);
-    int32_t SpeakerVolume(uint32_t& volume) const;
-    int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
-    int32_t MinSpeakerVolume(uint32_t& minVolume) const;
-    int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
-    int32_t SpeakerVolumeIsAvailable(bool& available);
-    int32_t SpeakerMuteIsAvailable(bool& available);
-    int32_t SetSpeakerMute(bool enable);
-    int32_t SpeakerMute(bool& enabled) const;
-    int32_t StereoPlayoutIsAvailable(bool& available);
-    int32_t StereoRecordingIsAvailable(bool& available);
-    int32_t MicrophoneMuteIsAvailable(bool& available);
-    int32_t SetMicrophoneMute(bool enable);
-    int32_t MicrophoneMute(bool& enabled) const;
-    int32_t MicrophoneBoostIsAvailable(bool& available);
-    int32_t SetMicrophoneBoost(bool enable);
-    int32_t MicrophoneBoost(bool& enabled) const;
-    int32_t MicrophoneVolumeIsAvailable(bool& available);
-    int32_t SetMicrophoneVolume(uint32_t volume);
-    int32_t MicrophoneVolume(uint32_t& volume) const;
-    int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
-    int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
-    int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
-    int32_t Close();
-    int32_t CloseSpeaker();
-    int32_t CloseMicrophone();
-    bool SpeakerIsInitialized() const;
-    bool MicrophoneIsInitialized() const;
 
-public:
-    AudioMixerManagerMac(const int32_t id);
-    ~AudioMixerManagerMac();
+class AudioMixerManagerMac {
+ public:
+  int32_t OpenSpeaker(AudioDeviceID deviceID);
+  int32_t OpenMicrophone(AudioDeviceID deviceID);
+  int32_t SetSpeakerVolume(uint32_t volume);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t StereoPlayoutIsAvailable(bool& available);
+  int32_t StereoRecordingIsAvailable(bool& available);
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneBoostIsAvailable(bool& available);
+  int32_t SetMicrophoneBoost(bool enable);
+  int32_t MicrophoneBoost(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
+  int32_t Close();
+  int32_t CloseSpeaker();
+  int32_t CloseMicrophone();
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
 
-private:
-    static void logCAMsg(const TraceLevel level,
-                         const TraceModule module,
-                         const int32_t id, const char *msg,
-                         const char *err);
+ public:
+  AudioMixerManagerMac(const int32_t id);
+  ~AudioMixerManagerMac();
 
-private:
-    CriticalSectionWrapper& _critSect;
-    int32_t _id;
+ private:
+  static void logCAMsg(const TraceLevel level,
+                       const TraceModule module,
+                       const int32_t id,
+                       const char* msg,
+                       const char* err);
 
-    AudioDeviceID _inputDeviceID;
-    AudioDeviceID _outputDeviceID;
+ private:
+  CriticalSectionWrapper& _critSect;
+  int32_t _id;
 
-    uint16_t _noInputChannels;
-    uint16_t _noOutputChannels;
+  AudioDeviceID _inputDeviceID;
+  AudioDeviceID _outputDeviceID;
 
+  uint16_t _noInputChannels;
+  uint16_t _noOutputChannels;
 };
-	
+
 }  // namespace webrtc
 
 #endif  // AUDIO_MIXER_MAC_H
diff --git a/webrtc/modules/audio_device/main/interface/audio_device.h b/webrtc/modules/audio_device/main/interface/audio_device.h
deleted file mode 100644
index 71f16b685a..0000000000
--- a/webrtc/modules/audio_device/main/interface/audio_device.h
+++ /dev/null
@@ -1,16 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
-#define MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
-
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-#endif  // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
diff --git a/webrtc/modules/audio_device/main/source/OWNERS b/webrtc/modules/audio_device/main/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/webrtc/modules/audio_device/main/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/webrtc/modules/audio_device/main/source/audio_device.gypi b/webrtc/modules/audio_device/main/source/audio_device.gypi
deleted file mode 100644
index 126eb98ac1..0000000000
--- a/webrtc/modules/audio_device/main/source/audio_device.gypi
+++ /dev/null
@@ -1,14 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../audio_device.gypi',
-  ],
-}
-
diff --git a/webrtc/modules/audio_device/test/audio_device_test_api.cc b/webrtc/modules/audio_device/test/audio_device_test_api.cc
index f36a049ffc..26a2dcd8c4 100644
--- a/webrtc/modules/audio_device/test/audio_device_test_api.cc
+++ b/webrtc/modules/audio_device/test/audio_device_test_api.cc
@@ -85,7 +85,7 @@ class AudioTransportAPI: public AudioTransport {
   int32_t RecordedDataIsAvailable(const void* audioSamples,
                                   const size_t nSamples,
                                   const size_t nBytesPerSample,
-                                  const uint8_t nChannels,
+                                  const size_t nChannels,
                                   const uint32_t sampleRate,
                                   const uint32_t totalDelay,
                                   const int32_t clockSkew,
@@ -110,7 +110,7 @@ class AudioTransportAPI: public AudioTransport {
 
   int32_t NeedMorePlayData(const size_t nSamples,
                            const size_t nBytesPerSample,
-                           const uint8_t nChannels,
+                           const size_t nChannels,
                            const uint32_t sampleRate,
                            void* audioSamples,
                            size_t& nSamplesOut,
@@ -128,29 +128,6 @@ class AudioTransportAPI: public AudioTransport {
     return 0;
   }
 
-  int OnDataAvailable(const int voe_channels[],
-                      int number_of_voe_channels,
-                      const int16_t* audio_data,
-                      int sample_rate,
-                      int number_of_channels,
-                      size_t number_of_frames,
-                      int audio_delay_milliseconds,
-                      int current_volume,
-                      bool key_pressed,
-                      bool need_audio_processing) override {
-    return 0;
-  }
-
-  void PushCaptureData(int voe_channel, const void* audio_data,
-                       int bits_per_sample, int sample_rate,
-                       int number_of_channels,
-                       size_t number_of_frames) override {}
-
-  void PullRenderData(int bits_per_sample, int sample_rate,
-                      int number_of_channels, size_t number_of_frames,
-                      void* audio_data,
-                      int64_t* elapsed_time_ms,
-                      int64_t* ntp_time_ms) override {}
  private:
   uint32_t rec_count_;
   uint32_t play_count_;
@@ -1040,9 +1017,15 @@ TEST_F(AudioDeviceAPITest, MicrophoneVolumeIsAvailable) {
 // MicrophoneVolume
 // MaxMicrophoneVolume
 // MinMicrophoneVolume
-// NOTE: Disabled on mac due to issue 257.
-#ifndef WEBRTC_MAC
-TEST_F(AudioDeviceAPITest, MicrophoneVolumeTests) {
+
+// Disabled on Mac and Linux,
+// see https://bugs.chromium.org/p/webrtc/issues/detail?id=5414
+#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
+#define MAYBE_MicrophoneVolumeTests DISABLED_MicrophoneVolumeTests
+#else
+#define MAYBE_MicrophoneVolumeTests MicrophoneVolumeTests
+#endif
+TEST_F(AudioDeviceAPITest, MAYBE_MicrophoneVolumeTests) {
   uint32_t vol(0);
   uint32_t volume(0);
   uint32_t maxVolume(0);
@@ -1135,7 +1118,6 @@ TEST_F(AudioDeviceAPITest, MicrophoneVolumeTests) {
     EXPECT_EQ(0, audio_device_->SetMicrophoneVolume(maxVolume/10));
   }
 }
-#endif  // !WEBRTC_MAC
 
 TEST_F(AudioDeviceAPITest, SpeakerMuteIsAvailable) {
   bool available;
diff --git a/webrtc/modules/audio_device/test/audio_device_test_defines.h b/webrtc/modules/audio_device/test/audio_device_test_defines.h
index 5068646bdd..cc8e3e3aef 100644
--- a/webrtc/modules/audio_device/test/audio_device_test_defines.h
+++ b/webrtc/modules/audio_device/test/audio_device_test_defines.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 #ifdef _WIN32
diff --git a/webrtc/modules/audio_device/test/func_test_manager.cc b/webrtc/modules/audio_device/test/func_test_manager.cc
index 0ebfc8395f..0a2963e9ae 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.cc
+++ b/webrtc/modules/audio_device/test/func_test_manager.cc
@@ -194,7 +194,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
     const void* audioSamples,
     const size_t nSamples,
     const size_t nBytesPerSample,
-    const uint8_t nChannels,
+    const size_t nChannels,
     const uint32_t samplesPerSec,
     const uint32_t totalDelayMS,
     const int32_t clockDrift,
@@ -339,7 +339,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
 int32_t AudioTransportImpl::NeedMorePlayData(
     const size_t nSamples,
     const size_t nBytesPerSample,
-    const uint8_t nChannels,
+    const size_t nChannels,
     const uint32_t samplesPerSec,
     void* audioSamples,
     size_t& nSamplesOut,
@@ -365,7 +365,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
                 int16_t* ptr16Out = NULL;
 
                 const size_t nSamplesIn = packet->nSamples;
-                const uint8_t nChannelsIn = packet->nChannels;
+                const size_t nChannelsIn = packet->nChannels;
                 const uint32_t samplesPerSecIn = packet->samplesPerSec;
                 const size_t nBytesPerSampleIn = packet->nBytesPerSample;
 
@@ -573,32 +573,6 @@ int32_t AudioTransportImpl::NeedMorePlayData(
     return 0;
 }
 
-int AudioTransportImpl::OnDataAvailable(const int voe_channels[],
-                                        int number_of_voe_channels,
-                                        const int16_t* audio_data,
-                                        int sample_rate,
-                                        int number_of_channels,
-                                        size_t number_of_frames,
-                                        int audio_delay_milliseconds,
-                                        int current_volume,
-                                        bool key_pressed,
-                                        bool need_audio_processing) {
-  return 0;
-}
-
-void AudioTransportImpl::PushCaptureData(int voe_channel,
-                                         const void* audio_data,
-                                         int bits_per_sample, int sample_rate,
-                                         int number_of_channels,
-                                         size_t number_of_frames) {}
-
-void AudioTransportImpl::PullRenderData(int bits_per_sample, int sample_rate,
-                                        int number_of_channels,
-                                        size_t number_of_frames,
-                                        void* audio_data,
-                                        int64_t* elapsed_time_ms,
-                                        int64_t* ntp_time_ms) {}
-
 FuncTestManager::FuncTestManager() :
     _audioDevice(NULL),
     _audioEventObserver(NULL),
@@ -686,7 +660,7 @@ int32_t FuncTestManager::Close()
         _audioDevice = NULL;
     }
 
-    // return the ThreadWrapper (singleton)
+    // return the PlatformThread (singleton)
     Trace::ReturnTrace();
 
     // PRINT_TEST_RESULTS;
diff --git a/webrtc/modules/audio_device/test/func_test_manager.h b/webrtc/modules/audio_device/test/func_test_manager.h
index 6ef13490d8..b7cc81cc1a 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.h
+++ b/webrtc/modules/audio_device/test/func_test_manager.h
@@ -49,7 +49,7 @@ struct AudioPacket
     uint8_t dataBuffer[4 * 960];
     size_t nSamples;
     size_t nBytesPerSample;
-    uint8_t nChannels;
+    size_t nChannels;
     uint32_t samplesPerSec;
 };
 
@@ -88,7 +88,7 @@ public:
     int32_t RecordedDataIsAvailable(const void* audioSamples,
                                     const size_t nSamples,
                                     const size_t nBytesPerSample,
-                                    const uint8_t nChannels,
+                                    const size_t nChannels,
                                     const uint32_t samplesPerSec,
                                     const uint32_t totalDelayMS,
                                     const int32_t clockDrift,
@@ -98,35 +98,13 @@ public:
 
     int32_t NeedMorePlayData(const size_t nSamples,
                              const size_t nBytesPerSample,
-                             const uint8_t nChannels,
+                             const size_t nChannels,
                              const uint32_t samplesPerSec,
                              void* audioSamples,
                              size_t& nSamplesOut,
                              int64_t* elapsed_time_ms,
                              int64_t* ntp_time_ms) override;
 
-    int OnDataAvailable(const int voe_channels[],
-                        int number_of_voe_channels,
-                        const int16_t* audio_data,
-                        int sample_rate,
-                        int number_of_channels,
-                        size_t number_of_frames,
-                        int audio_delay_milliseconds,
-                        int current_volume,
-                        bool key_pressed,
-                        bool need_audio_processing) override;
-
-    void PushCaptureData(int voe_channel, const void* audio_data,
-                         int bits_per_sample, int sample_rate,
-                         int number_of_channels,
-                         size_t number_of_frames) override;
-
-    void PullRenderData(int bits_per_sample, int sample_rate,
-                        int number_of_channels, size_t number_of_frames,
-                        void* audio_data,
-                        int64_t* elapsed_time_ms,
-                        int64_t* ntp_time_ms) override;
-
     AudioTransportImpl(AudioDeviceModule* audioDevice);
     ~AudioTransportImpl();
 
diff --git a/webrtc/modules/audio_device/win/audio_device_wave_win.cc b/webrtc/modules/audio_device/win/audio_device_wave_win.cc
index 96bee7425a..6f4d7df397 100644
--- a/webrtc/modules/audio_device/win/audio_device_wave_win.cc
+++ b/webrtc/modules/audio_device/win/audio_device_wave_win.cc
@@ -228,15 +228,9 @@ int32_t AudioDeviceWindowsWave::Init()
     }
 
     const char* threadName = "webrtc_audio_module_thread";
-    _ptrThread = ThreadWrapper::CreateThread(ThreadFunc, this, threadName);
-    if (!_ptrThread->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "failed to start the audio thread");
-        _ptrThread.reset();
-        return -1;
-    }
-    _ptrThread->SetPriority(kRealtimePriority);
+    _ptrThread.reset(new rtc::PlatformThread(ThreadFunc, this, threadName));
+    _ptrThread->Start();
+    _ptrThread->SetPriority(rtc::kRealtimePriority);
 
     const bool periodic(true);
     if (!_timeEvent.StartTimer(periodic, TIMER_PERIOD_MS))
@@ -250,12 +244,8 @@ int32_t AudioDeviceWindowsWave::Init()
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "periodic timer (dT=%d) is now active", TIMER_PERIOD_MS);
 
-    _hGetCaptureVolumeThread = CreateThread(NULL,
-                                            0,
-                                            GetCaptureVolumeThread,
-                                            this,
-                                            0,
-                                            NULL);
+    _hGetCaptureVolumeThread =
+        CreateThread(NULL, 0, GetCaptureVolumeThread, this, 0, NULL);
     if (_hGetCaptureVolumeThread == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -265,12 +255,8 @@ int32_t AudioDeviceWindowsWave::Init()
 
     SetThreadPriority(_hGetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
 
-    _hSetCaptureVolumeThread = CreateThread(NULL,
-                                            0,
-                                            SetCaptureVolumeThread,
-                                            this,
-                                            0,
-                                            NULL);
+    _hSetCaptureVolumeThread =
+        CreateThread(NULL, 0, SetCaptureVolumeThread, this, 0, NULL);
     if (_hSetCaptureVolumeThread == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -303,7 +289,7 @@ int32_t AudioDeviceWindowsWave::Terminate()
 
     if (_ptrThread)
     {
-        ThreadWrapper* tmpThread = _ptrThread.release();
+        rtc::PlatformThread* tmpThread = _ptrThread.release();
         _critSect.Leave();
 
         _timeEvent.Set();
diff --git a/webrtc/modules/audio_device/win/audio_device_wave_win.h b/webrtc/modules/audio_device/win/audio_device_wave_win.h
index c99185c3ab..a1cfc6acbf 100644
--- a/webrtc/modules/audio_device/win/audio_device_wave_win.h
+++ b/webrtc/modules/audio_device/win/audio_device_wave_win.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/win/audio_mixer_manager_win.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 #pragma comment( lib, "winmm.lib" )
 
@@ -222,7 +222,8 @@ private:
     HANDLE                                  _hShutdownSetVolumeEvent;
     HANDLE                                  _hSetCaptureVolumeEvent;
 
-    rtc::scoped_ptr<ThreadWrapper>          _ptrThread;
+    // TODO(pbos): Remove scoped_ptr usage and use PlatformThread directly
+    rtc::scoped_ptr<rtc::PlatformThread>    _ptrThread;
 
     CriticalSectionWrapper&                 _critSectCb;
 
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 2d0c602ef0..9d91911bc2 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -30,12 +30,12 @@ source_set("audio_processing") {
     "aec/aec_resampler.c",
     "aec/aec_resampler.h",
     "aec/echo_cancellation.c",
+    "aec/echo_cancellation.h",
     "aec/echo_cancellation_internal.h",
-    "aec/include/echo_cancellation.h",
     "aecm/aecm_core.c",
     "aecm/aecm_core.h",
     "aecm/echo_control_mobile.c",
-    "aecm/include/echo_control_mobile.h",
+    "aecm/echo_control_mobile.h",
     "agc/agc.cc",
     "agc/agc.h",
     "agc/agc_manager_direct.cc",
@@ -161,8 +161,8 @@ source_set("audio_processing") {
   if (rtc_prefer_fixed_point) {
     defines += [ "WEBRTC_NS_FIXED" ]
     sources += [
-      "ns/include/noise_suppression_x.h",
       "ns/noise_suppression_x.c",
+      "ns/noise_suppression_x.h",
       "ns/nsx_core.c",
       "ns/nsx_core.h",
       "ns/nsx_defines.h",
@@ -176,8 +176,8 @@ source_set("audio_processing") {
     defines += [ "WEBRTC_NS_FLOAT" ]
     sources += [
       "ns/defines.h",
-      "ns/include/noise_suppression.h",
       "ns/noise_suppression.c",
+      "ns/noise_suppression.h",
       "ns/ns_core.c",
       "ns/ns_core.h",
       "ns/windows_private.h",
diff --git a/webrtc/modules/audio_processing/OWNERS b/webrtc/modules/audio_processing/OWNERS
index 7b760682b0..d14f7f8614 100644
--- a/webrtc/modules/audio_processing/OWNERS
+++ b/webrtc/modules/audio_processing/OWNERS
@@ -1,5 +1,4 @@
 aluebs@webrtc.org
-andrew@webrtc.org
 henrik.lundin@webrtc.org
 peah@webrtc.org
 
diff --git a/webrtc/modules/audio_processing/aec/aec_core.c b/webrtc/modules/audio_processing/aec/aec_core.c
index f8eed32372..901e0fde0b 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.c
+++ b/webrtc/modules/audio_processing/aec/aec_core.c
@@ -44,7 +44,6 @@ static const int countLen = 50;
 static const int kDelayMetricsAggregationWindow = 1250;  // 5 seconds at 16 kHz.
 
 // Quantities to control H band scaling for SWB input
-static const int flagHbandCn = 1;  // flag for adding comfort noise in H band
 static const float cnScaleHband =
     (float)0.4;  // scale for comfort noise in H band
 // Initial bin for averaging nlp gain in low band
@@ -135,6 +134,9 @@ WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
 WebRtcAecOverdriveAndSuppress WebRtcAec_OverdriveAndSuppress;
 WebRtcAecComfortNoise WebRtcAec_ComfortNoise;
 WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
+WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
+WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
+WebRtcAecWindowData WebRtcAec_WindowData;
 
 __inline static float MulRe(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bRe - aIm * bIm;
@@ -151,40 +153,49 @@ static int CmpFloat(const void* a, const void* b) {
   return (*da > *db) - (*da < *db);
 }
 
-static void FilterFar(AecCore* aec, float yf[2][PART_LEN1]) {
+static void FilterFar(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
   int i;
-  for (i = 0; i < aec->num_partitions; i++) {
+  for (i = 0; i < num_partitions; i++) {
     int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= aec->num_partitions * (PART_LEN1);
+    if (i + x_fft_buf_block_pos >= num_partitions) {
+      xPos -= num_partitions * (PART_LEN1);
     }
 
     for (j = 0; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
+      y_fft[0][j] += MulRe(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
+      y_fft[1][j] += MulIm(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
     }
   }
 }
 
-static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1]) {
-  const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-  const float error_threshold = aec->extended_filter_enabled
+static void ScaleErrorSignal(int extended_filter_enabled,
+                             float normal_mu,
+                             float normal_error_threshold,
+                             float x_pow[PART_LEN1],
+                             float ef[2][PART_LEN1]) {
+  const float mu = extended_filter_enabled ? kExtendedMu : normal_mu;
+  const float error_threshold = extended_filter_enabled
                                     ? kExtendedErrorThreshold
-                                    : aec->normal_error_threshold;
+                                    : normal_error_threshold;
   int i;
   float abs_ef;
   for (i = 0; i < (PART_LEN1); i++) {
-    ef[0][i] /= (aec->xPow[i] + 1e-10f);
-    ef[1][i] /= (aec->xPow[i] + 1e-10f);
+    ef[0][i] /= (x_pow[i] + 1e-10f);
+    ef[1][i] /= (x_pow[i] + 1e-10f);
     abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
     if (abs_ef > error_threshold) {
@@ -199,59 +210,40 @@ static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1]) {
   }
 }
 
-// Time-unconstrined filter adaptation.
-// TODO(andrew): consider for a low-complexity mode.
-// static void FilterAdaptationUnconstrained(AecCore* aec, float *fft,
-//                                          float ef[2][PART_LEN1]) {
-//  int i, j;
-//  for (i = 0; i < aec->num_partitions; i++) {
-//    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
-//    int pos;
-//    // Check for wrap
-//    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-//      xPos -= aec->num_partitions * PART_LEN1;
-//    }
-//
-//    pos = i * PART_LEN1;
-//
-//    for (j = 0; j < PART_LEN1; j++) {
-//      aec->wfBuf[0][pos + j] += MulRe(aec->xfBuf[0][xPos + j],
-//                                      -aec->xfBuf[1][xPos + j],
-//                                      ef[0][j], ef[1][j]);
-//      aec->wfBuf[1][pos + j] += MulIm(aec->xfBuf[0][xPos + j],
-//                                      -aec->xfBuf[1][xPos + j],
-//                                      ef[0][j], ef[1][j]);
-//    }
-//  }
-//}
-
-static void FilterAdaptation(AecCore* aec, float* fft, float ef[2][PART_LEN1]) {
+
+static void FilterAdaptation(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
   int i, j;
-  for (i = 0; i < aec->num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * (PART_LEN1);
+  float fft[PART_LEN2];
+  for (i = 0; i < num_partitions; i++) {
+    int xPos = (i + x_fft_buf_block_pos) * (PART_LEN1);
     int pos;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= aec->num_partitions * PART_LEN1;
+    if (i + x_fft_buf_block_pos >= num_partitions) {
+      xPos -= num_partitions * PART_LEN1;
     }
 
     pos = i * PART_LEN1;
 
     for (j = 0; j < PART_LEN; j++) {
 
-      fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
-                         -aec->xfBuf[1][xPos + j],
-                         ef[0][j],
-                         ef[1][j]);
-      fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
-                             -aec->xfBuf[1][xPos + j],
-                             ef[0][j],
-                             ef[1][j]);
+      fft[2 * j] = MulRe(x_fft_buf[0][xPos + j],
+                         -x_fft_buf[1][xPos + j],
+                         e_fft[0][j],
+                         e_fft[1][j]);
+      fft[2 * j + 1] = MulIm(x_fft_buf[0][xPos + j],
+                             -x_fft_buf[1][xPos + j],
+                             e_fft[0][j],
+                             e_fft[1][j]);
     }
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN],
-                   ef[1][PART_LEN]);
+    fft[1] = MulRe(x_fft_buf[0][xPos + PART_LEN],
+                   -x_fft_buf[1][xPos + PART_LEN],
+                   e_fft[0][PART_LEN],
+                   e_fft[1][PART_LEN]);
 
     aec_rdft_inverse_128(fft);
     memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
@@ -265,12 +257,12 @@ static void FilterAdaptation(AecCore* aec, float* fft, float ef[2][PART_LEN1]) {
     }
     aec_rdft_forward_128(fft);
 
-    aec->wfBuf[0][pos] += fft[0];
-    aec->wfBuf[0][pos + PART_LEN] += fft[1];
+    h_fft_buf[0][pos] += fft[0];
+    h_fft_buf[0][pos + PART_LEN] += fft[1];
 
     for (j = 1; j < PART_LEN; j++) {
-      aec->wfBuf[0][pos + j] += fft[2 * j];
-      aec->wfBuf[1][pos + j] += fft[2 * j + 1];
+      h_fft_buf[0][pos + j] += fft[2 * j];
+      h_fft_buf[1][pos + j] += fft[2 * j + 1];
     }
   }
 }
@@ -334,12 +326,13 @@ const float WebRtcAec_kMinFarendPSD = 15;
 //  - sde : cross-PSD of near-end and residual echo
 //  - sxd : cross-PSD of near-end and far-end
 //
-// In addition to updating the PSDs, also the filter diverge state is determined
-// upon actions are taken.
+// In addition to updating the PSDs, also the filter diverge state is
+// determined.
 static void SmoothedPSD(AecCore* aec,
                         float efw[2][PART_LEN1],
                         float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1]) {
+                        float xfw[2][PART_LEN1],
+                        int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh = aec->extended_filter_enabled
       ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
@@ -380,15 +373,12 @@ static void SmoothedPSD(AecCore* aec,
     seSum += aec->se[i];
   }
 
-  // Divergent filter safeguard.
+  // Divergent filter safeguard update.
   aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
 
-  if (aec->divergeState)
-    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
-
-  // Reset if error is significantly larger than nearend (13 dB).
-  if (!aec->extended_filter_enabled && seSum > (19.95f * sdSum))
-    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+  // Signal extreme filter divergence if the error is significantly larger
+  // than the nearend (13 dB).
+  *extreme_filter_divergence = (seSum > (19.95f * sdSum));
 }
 
 // Window time domain data to be used by the fft.
@@ -417,32 +407,15 @@ __inline static void StoreAsComplex(const float* data,
 
 static void SubbandCoherence(AecCore* aec,
                              float efw[2][PART_LEN1],
+                             float dfw[2][PART_LEN1],
                              float xfw[2][PART_LEN1],
                              float* fft,
                              float* cohde,
-                             float* cohxd) {
-  float dfw[2][PART_LEN1];
+                             float* cohxd,
+                             int* extreme_filter_divergence) {
   int i;
 
-  if (aec->delayEstCtr == 0)
-    aec->delayIdx = PartitionDelay(aec);
-
-  // Use delayed far.
-  memcpy(xfw,
-         aec->xfwBuf + aec->delayIdx * PART_LEN1,
-         sizeof(xfw[0][0]) * 2 * PART_LEN1);
-
-  // Windowed near fft
-  WindowData(fft, aec->dBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, dfw);
-
-  // Windowed error fft
-  WindowData(fft, aec->eBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, efw);
-
-  SmoothedPSD(aec, efw, dfw, xfw);
+  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
 
   // Subband coherence
   for (i = 0; i < PART_LEN1; i++) {
@@ -458,23 +431,23 @@ static void SubbandCoherence(AecCore* aec,
 static void GetHighbandGain(const float* lambda, float* nlpGainHband) {
   int i;
 
-  nlpGainHband[0] = (float)0.0;
+  *nlpGainHband = (float)0.0;
   for (i = freqAvgIc; i < PART_LEN1 - 1; i++) {
-    nlpGainHband[0] += lambda[i];
+    *nlpGainHband += lambda[i];
   }
-  nlpGainHband[0] /= (float)(PART_LEN1 - 1 - freqAvgIc);
+  *nlpGainHband /= (float)(PART_LEN1 - 1 - freqAvgIc);
 }
 
 static void ComfortNoise(AecCore* aec,
                          float efw[2][PART_LEN1],
-                         complex_t* comfortNoiseHband,
+                         float comfortNoiseHband[2][PART_LEN1],
                          const float* noisePow,
                          const float* lambda) {
   int i, num;
   float rand[PART_LEN];
   float noise, noiseAvg, tmp, tmpAvg;
   int16_t randW16[PART_LEN];
-  complex_t u[PART_LEN1];
+  float u[2][PART_LEN1];
 
   const float pi2 = 6.28318530717959f;
 
@@ -486,22 +459,22 @@ static void ComfortNoise(AecCore* aec,
 
   // Reject LF noise
   u[0][0] = 0;
-  u[0][1] = 0;
+  u[1][0] = 0;
   for (i = 1; i < PART_LEN1; i++) {
     tmp = pi2 * rand[i - 1];
 
     noise = sqrtf(noisePow[i]);
-    u[i][0] = noise * cosf(tmp);
-    u[i][1] = -noise * sinf(tmp);
+    u[0][i] = noise * cosf(tmp);
+    u[1][i] = -noise * sinf(tmp);
   }
-  u[PART_LEN][1] = 0;
+  u[1][PART_LEN] = 0;
 
   for (i = 0; i < PART_LEN1; i++) {
     // This is the proper weighting to match the background noise power
     tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
     // tmp = 1 - lambda[i];
-    efw[0][i] += tmp * u[i][0];
-    efw[1][i] += tmp * u[i][1];
+    efw[0][i] += tmp * u[0][i];
+    efw[1][i] += tmp * u[1][i];
   }
 
   // For H band comfort noise
@@ -509,7 +482,7 @@ static void ComfortNoise(AecCore* aec,
   noiseAvg = 0.0;
   tmpAvg = 0.0;
   num = 0;
-  if (aec->num_bands > 1 && flagHbandCn == 1) {
+  if (aec->num_bands > 1) {
 
     // average noise scale
     // average over second half of freq spectrum (i.e., 4->8khz)
@@ -534,21 +507,24 @@ static void ComfortNoise(AecCore* aec,
     // TODO: we should probably have a new random vector here.
     // Reject LF noise
     u[0][0] = 0;
-    u[0][1] = 0;
+    u[1][0] = 0;
     for (i = 1; i < PART_LEN1; i++) {
       tmp = pi2 * rand[i - 1];
 
       // Use average noise for H band
-      u[i][0] = noiseAvg * (float)cos(tmp);
-      u[i][1] = -noiseAvg * (float)sin(tmp);
+      u[0][i] = noiseAvg * (float)cos(tmp);
+      u[1][i] = -noiseAvg * (float)sin(tmp);
     }
-    u[PART_LEN][1] = 0;
+    u[1][PART_LEN] = 0;
 
     for (i = 0; i < PART_LEN1; i++) {
       // Use average NLP weight for H band
-      comfortNoiseHband[i][0] = tmpAvg * u[i][0];
-      comfortNoiseHband[i][1] = tmpAvg * u[i][1];
+      comfortNoiseHband[0][i] = tmpAvg * u[0][i];
+      comfortNoiseHband[1][i] = tmpAvg * u[1][i];
     }
+  } else {
+    memset(comfortNoiseHband, 0,
+           2 * PART_LEN1 * sizeof(comfortNoiseHband[0][0]));
   }
 }
 
@@ -837,21 +813,29 @@ static void UpdateDelayMetrics(AecCore* self) {
   return;
 }
 
-static void TimeToFrequency(float time_data[PART_LEN2],
-                            float freq_data[2][PART_LEN1],
-                            int window) {
-  int i = 0;
-
-  // TODO(bjornv): Should we have a different function/wrapper for windowed FFT?
-  if (window) {
-    for (i = 0; i < PART_LEN; i++) {
-      time_data[i] *= WebRtcAec_sqrtHanning[i];
-      time_data[PART_LEN + i] *= WebRtcAec_sqrtHanning[PART_LEN - i];
-    }
+static void ScaledInverseFft(float freq_data[2][PART_LEN1],
+                             float time_data[PART_LEN2],
+                             float scale,
+                             int conjugate) {
+  int i;
+  const float normalization = scale / ((float)PART_LEN2);
+  const float sign = (conjugate ? -1 : 1);
+  time_data[0] = freq_data[0][0] * normalization;
+  time_data[1] = freq_data[0][PART_LEN] * normalization;
+  for (i = 1; i < PART_LEN; i++) {
+    time_data[2 * i] = freq_data[0][i] * normalization;
+    time_data[2 * i + 1] = sign * freq_data[1][i] * normalization;
   }
+  aec_rdft_inverse_128(time_data);
+}
+
 
+static void Fft(float time_data[PART_LEN2],
+                float freq_data[2][PART_LEN1]) {
+  int i;
   aec_rdft_forward_128(time_data);
-  // Reorder.
+
+  // Reorder fft output data.
   freq_data[1][0] = 0;
   freq_data[1][PART_LEN] = 0;
   freq_data[0][0] = time_data[0];
@@ -862,13 +846,6 @@ static void TimeToFrequency(float time_data[PART_LEN2],
   }
 }
 
-static int MoveFarReadPtrWithoutSystemDelayUpdate(AecCore* self, int elements) {
-  WebRtc_MoveReadPtr(self->far_buf_windowed, elements);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  WebRtc_MoveReadPtr(self->far_time_buf, elements);
-#endif
-  return WebRtc_MoveReadPtr(self->far_buf, elements);
-}
 
 static int SignalBasedDelayCorrection(AecCore* self) {
   int delay_correction = 0;
@@ -909,7 +886,7 @@ static int SignalBasedDelayCorrection(AecCore* self) {
     const int upper_bound = self->num_partitions * 3 / 4;
     const int do_correction = delay <= lower_bound || delay > upper_bound;
     if (do_correction == 1) {
-      int available_read = (int)WebRtc_available_read(self->far_buf);
+      int available_read = (int)WebRtc_available_read(self->far_time_buf);
       // With |shift_offset| we gradually rely on the delay estimates.  For
       // positive delays we reduce the correction by |shift_offset| to lower the
       // risk of pushing the AEC into a non causal state.  For negative delays
@@ -942,13 +919,94 @@ static int SignalBasedDelayCorrection(AecCore* self) {
   return delay_correction;
 }
 
-static void NonLinearProcessing(AecCore* aec,
-                                float* output,
-                                float* const* outputH) {
-  float efw[2][PART_LEN1], xfw[2][PART_LEN1];
-  complex_t comfortNoiseHband[PART_LEN1];
+static void EchoSubtraction(
+    AecCore* aec,
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    int metrics_mode,
+    int extended_filter_enabled,
+    float normal_mu,
+    float normal_error_threshold,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float* const y,
+    float x_pow[PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    PowerLevel* linout_level,
+    float echo_subtractor_output[PART_LEN]) {
+  float s_fft[2][PART_LEN1];
+  float e_extended[PART_LEN2];
+  float s_extended[PART_LEN2];
+  float *s;
+  float e[PART_LEN];
+  float e_fft[2][PART_LEN1];
+  int i;
+  memset(s_fft, 0, sizeof(s_fft));
+
+  // Conditionally reset the echo subtraction filter if the filter has diverged
+  // significantly.
+  if (!aec->extended_filter_enabled &&
+      aec->extreme_filter_divergence) {
+    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+    aec->extreme_filter_divergence = 0;
+  }
+
+  // Produce echo estimate s_fft.
+  WebRtcAec_FilterFar(num_partitions,
+                      x_fft_buf_block_pos,
+                      x_fft_buf,
+                      h_fft_buf,
+                      s_fft);
+
+  // Compute the time-domain echo estimate s.
+  ScaledInverseFft(s_fft, s_extended, 2.0f, 0);
+  s = &s_extended[PART_LEN];
+
+  // Compute the time-domain echo prediction error.
+  for (i = 0; i < PART_LEN; ++i) {
+    e[i] = y[i] - s[i];
+  }
+
+  // Compute the frequency domain echo prediction error.
+  memset(e_extended, 0, sizeof(float) * PART_LEN);
+  memcpy(e_extended + PART_LEN, e, sizeof(float) * PART_LEN);
+  Fft(e_extended, e_fft);
+
+  RTC_AEC_DEBUG_RAW_WRITE(aec->e_fft_file,
+                          &e_fft[0][0],
+                          sizeof(e_fft[0][0]) * PART_LEN1 * 2);
+
+  if (metrics_mode == 1) {
+    // Note that the first PART_LEN samples in fft (before transformation) are
+    // zero. Hence, the scaling by two in UpdateLevel() should not be
+    // performed. That scaling is taken care of in UpdateMetrics() instead.
+    UpdateLevel(linout_level, e_fft);
+  }
+
+  // Scale error signal inversely with far power.
+  WebRtcAec_ScaleErrorSignal(extended_filter_enabled,
+                             normal_mu,
+                             normal_error_threshold,
+                             x_pow,
+                             e_fft);
+  WebRtcAec_FilterAdaptation(num_partitions,
+                             x_fft_buf_block_pos,
+                             x_fft_buf,
+                             e_fft,
+                             h_fft_buf);
+  memcpy(echo_subtractor_output, e, sizeof(float) * PART_LEN);
+}
+
+
+static void EchoSuppression(AecCore* aec,
+                            float farend[PART_LEN2],
+                            float* echo_subtractor_output,
+                            float* output,
+                            float* const* outputH) {
+  float efw[2][PART_LEN1];
+  float xfw[2][PART_LEN1];
+  float dfw[2][PART_LEN1];
+  float comfortNoiseHband[2][PART_LEN1];
   float fft[PART_LEN2];
-  float scale, dtmp;
   float nlpGainHband;
   int i;
   size_t j;
@@ -972,27 +1030,51 @@ static void NonLinearProcessing(AecCore* aec,
 
   float* xfw_ptr = NULL;
 
-  aec->delayEstCtr++;
-  if (aec->delayEstCtr == delayEstInterval) {
-    aec->delayEstCtr = 0;
-  }
+  // Update eBuf with echo subtractor output.
+  memcpy(aec->eBuf + PART_LEN,
+         echo_subtractor_output,
+         sizeof(float) * PART_LEN);
 
-  // initialize comfort noise for H band
-  memset(comfortNoiseHband, 0, sizeof(comfortNoiseHband));
-  nlpGainHband = (float)0.0;
-  dtmp = (float)0.0;
+  // Analysis filter banks for the echo suppressor.
+  // Windowed near-end ffts.
+  WindowData(fft, aec->dBuf);
+  aec_rdft_forward_128(fft);
+  StoreAsComplex(fft, dfw);
+
+  // Windowed echo suppressor output ffts.
+  WindowData(fft, aec->eBuf);
+  aec_rdft_forward_128(fft);
+  StoreAsComplex(fft, efw);
 
-  // We should always have at least one element stored in |far_buf|.
-  assert(WebRtc_available_read(aec->far_buf_windowed) > 0);
   // NLP
-  WebRtc_ReadBuffer(aec->far_buf_windowed, (void**)&xfw_ptr, &xfw[0][0], 1);
 
-  // TODO(bjornv): Investigate if we can reuse |far_buf_windowed| instead of
-  // |xfwBuf|.
+  // Convert far-end partition to the frequency domain with windowing.
+  WindowData(fft, farend);
+  Fft(fft, xfw);
+  xfw_ptr = &xfw[0][0];
+
   // Buffer far.
   memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
 
-  WebRtcAec_SubbandCoherence(aec, efw, xfw, fft, cohde, cohxd);
+  aec->delayEstCtr++;
+  if (aec->delayEstCtr == delayEstInterval) {
+    aec->delayEstCtr = 0;
+    aec->delayIdx = WebRtcAec_PartitionDelay(aec);
+  }
+
+  // Use delayed far.
+  memcpy(xfw,
+         aec->xfwBuf + aec->delayIdx * PART_LEN1,
+         sizeof(xfw[0][0]) * 2 * PART_LEN1);
+
+  WebRtcAec_SubbandCoherence(aec, efw, dfw, xfw, fft, cohde, cohxd,
+                             &aec->extreme_filter_divergence);
+
+  // Select the microphone signal as output if the filter is deemed to have
+  // diverged.
+  if (aec->divergeState) {
+    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
+  }
 
   hNlXdAvg = 0;
   for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
@@ -1098,67 +1180,51 @@ static void NonLinearProcessing(AecCore* aec,
     // scaling only in UpdateMetrics().
     UpdateLevel(&aec->nlpoutlevel, efw);
   }
+
   // Inverse error fft.
-  fft[0] = efw[0][0];
-  fft[1] = efw[0][PART_LEN];
-  for (i = 1; i < PART_LEN; i++) {
-    fft[2 * i] = efw[0][i];
-    // Sign change required by Ooura fft.
-    fft[2 * i + 1] = -efw[1][i];
-  }
-  aec_rdft_inverse_128(fft);
+  ScaledInverseFft(efw, fft, 2.0f, 1);
 
   // Overlap and add to obtain output.
-  scale = 2.0f / PART_LEN2;
   for (i = 0; i < PART_LEN; i++) {
-    fft[i] *= scale;  // fft scaling
-    fft[i] = fft[i] * WebRtcAec_sqrtHanning[i] + aec->outBuf[i];
-
-    fft[PART_LEN + i] *= scale;  // fft scaling
-    aec->outBuf[i] = fft[PART_LEN + i] * WebRtcAec_sqrtHanning[PART_LEN - i];
+    output[i] = (fft[i] * WebRtcAec_sqrtHanning[i] +
+                 aec->outBuf[i] * WebRtcAec_sqrtHanning[PART_LEN - i]);
 
     // Saturate output to keep it in the allowed range.
     output[i] = WEBRTC_SPL_SAT(
-        WEBRTC_SPL_WORD16_MAX, fft[i], WEBRTC_SPL_WORD16_MIN);
+        WEBRTC_SPL_WORD16_MAX, output[i], WEBRTC_SPL_WORD16_MIN);
   }
+  memcpy(aec->outBuf, &fft[PART_LEN], PART_LEN * sizeof(aec->outBuf[0]));
 
   // For H band
   if (aec->num_bands > 1) {
-
     // H band gain
     // average nlp over low band: average over second half of freq spectrum
     // (4->8khz)
     GetHighbandGain(hNl, &nlpGainHband);
 
     // Inverse comfort_noise
-    if (flagHbandCn == 1) {
-      fft[0] = comfortNoiseHband[0][0];
-      fft[1] = comfortNoiseHband[PART_LEN][0];
-      for (i = 1; i < PART_LEN; i++) {
-        fft[2 * i] = comfortNoiseHband[i][0];
-        fft[2 * i + 1] = comfortNoiseHband[i][1];
-      }
-      aec_rdft_inverse_128(fft);
-      scale = 2.0f / PART_LEN2;
-    }
+    ScaledInverseFft(comfortNoiseHband, fft, 2.0f, 0);
 
     // compute gain factor
     for (j = 0; j < aec->num_bands - 1; ++j) {
       for (i = 0; i < PART_LEN; i++) {
-        dtmp = aec->dBufH[j][i];
-        dtmp = dtmp * nlpGainHband;  // for variable gain
+        outputH[j][i] = aec->dBufH[j][i] * nlpGainHband;
+      }
+    }
 
-        // add some comfort noise where Hband is attenuated
-        if (flagHbandCn == 1 && j == 0) {
-          fft[i] *= scale;  // fft scaling
-          dtmp += cnScaleHband * fft[i];
-        }
+    // Add some comfort noise where Hband is attenuated.
+    for (i = 0; i < PART_LEN; i++) {
+      outputH[0][i] += cnScaleHband * fft[i];
+    }
 
-        // Saturate output to keep it in the allowed range.
+    // Saturate output to keep it in the allowed range.
+    for (j = 0; j < aec->num_bands - 1; ++j) {
+      for (i = 0; i < PART_LEN; i++) {
         outputH[j][i] = WEBRTC_SPL_SAT(
-            WEBRTC_SPL_WORD16_MAX, dtmp, WEBRTC_SPL_WORD16_MIN);
+            WEBRTC_SPL_WORD16_MAX, outputH[j][i], WEBRTC_SPL_WORD16_MIN);
       }
     }
+
   }
 
   // Copy the current block to the old position.
@@ -1177,11 +1243,9 @@ static void NonLinearProcessing(AecCore* aec,
 
 static void ProcessBlock(AecCore* aec) {
   size_t i;
-  float y[PART_LEN], e[PART_LEN];
-  float scale;
 
   float fft[PART_LEN2];
-  float xf[2][PART_LEN1], yf[2][PART_LEN1], ef[2][PART_LEN1];
+  float xf[2][PART_LEN1];
   float df[2][PART_LEN1];
   float far_spectrum = 0.0f;
   float near_spectrum = 0.0f;
@@ -1198,15 +1262,18 @@ static void ProcessBlock(AecCore* aec) {
 
   float nearend[PART_LEN];
   float* nearend_ptr = NULL;
+  float farend[PART_LEN2];
+  float* farend_ptr = NULL;
+  float echo_subtractor_output[PART_LEN];
   float output[PART_LEN];
   float outputH[NUM_HIGH_BANDS_MAX][PART_LEN];
   float* outputH_ptr[NUM_HIGH_BANDS_MAX];
+  float* xf_ptr = NULL;
+
   for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
     outputH_ptr[i] = outputH[i];
   }
 
-  float* xf_ptr = NULL;
-
   // Concatenate old and new nearend blocks.
   for (i = 0; i < aec->num_bands - 1; ++i) {
     WebRtc_ReadBuffer(aec->nearFrBufH[i],
@@ -1218,25 +1285,28 @@ static void ProcessBlock(AecCore* aec) {
   WebRtc_ReadBuffer(aec->nearFrBuf, (void**)&nearend_ptr, nearend, PART_LEN);
   memcpy(aec->dBuf + PART_LEN, nearend_ptr, sizeof(nearend));
 
-  // ---------- Ooura fft ----------
+  // We should always have at least one element stored in |far_buf|.
+  assert(WebRtc_available_read(aec->far_time_buf) > 0);
+  WebRtc_ReadBuffer(aec->far_time_buf, (void**)&farend_ptr, farend, 1);
 
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   {
-    float farend[PART_LEN];
-    float* farend_ptr = NULL;
-    WebRtc_ReadBuffer(aec->far_time_buf, (void**)&farend_ptr, farend, 1);
-    RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, farend_ptr, PART_LEN);
+    // TODO(minyue): |farend_ptr| starts from buffered samples. This will be
+    // modified when |aec->far_time_buf| is revised.
+    RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, &farend_ptr[PART_LEN], PART_LEN);
+
     RTC_AEC_DEBUG_WAV_WRITE(aec->nearFile, nearend_ptr, PART_LEN);
   }
 #endif
 
-  // We should always have at least one element stored in |far_buf|.
-  assert(WebRtc_available_read(aec->far_buf) > 0);
-  WebRtc_ReadBuffer(aec->far_buf, (void**)&xf_ptr, &xf[0][0], 1);
+  // Convert far-end signal to the frequency domain.
+  memcpy(fft, farend_ptr, sizeof(float) * PART_LEN2);
+  Fft(fft, xf);
+  xf_ptr = &xf[0][0];
 
   // Near fft
   memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
-  TimeToFrequency(fft, df, 0);
+  Fft(fft, df);
 
   // Power smoothing
   for (i = 0; i < PART_LEN1; i++) {
@@ -1314,60 +1384,25 @@ static void ProcessBlock(AecCore* aec) {
          &xf_ptr[PART_LEN1],
          sizeof(float) * PART_LEN1);
 
-  memset(yf, 0, sizeof(yf));
-
-  // Filter far
-  WebRtcAec_FilterFar(aec, yf);
-
-  // Inverse fft to obtain echo estimate and error.
-  fft[0] = yf[0][0];
-  fft[1] = yf[0][PART_LEN];
-  for (i = 1; i < PART_LEN; i++) {
-    fft[2 * i] = yf[0][i];
-    fft[2 * i + 1] = yf[1][i];
-  }
-  aec_rdft_inverse_128(fft);
-
-  scale = 2.0f / PART_LEN2;
-  for (i = 0; i < PART_LEN; i++) {
-    y[i] = fft[PART_LEN + i] * scale;  // fft scaling
-  }
-
-  for (i = 0; i < PART_LEN; i++) {
-    e[i] = nearend_ptr[i] - y[i];
-  }
-
-  // Error fft
-  memcpy(aec->eBuf + PART_LEN, e, sizeof(float) * PART_LEN);
-  memset(fft, 0, sizeof(float) * PART_LEN);
-  memcpy(fft + PART_LEN, e, sizeof(float) * PART_LEN);
-  // TODO(bjornv): Change to use TimeToFrequency().
-  aec_rdft_forward_128(fft);
-
-  ef[1][0] = 0;
-  ef[1][PART_LEN] = 0;
-  ef[0][0] = fft[0];
-  ef[0][PART_LEN] = fft[1];
-  for (i = 1; i < PART_LEN; i++) {
-    ef[0][i] = fft[2 * i];
-    ef[1][i] = fft[2 * i + 1];
-  }
-
-  RTC_AEC_DEBUG_RAW_WRITE(aec->e_fft_file,
-                          &ef[0][0],
-                          sizeof(ef[0][0]) * PART_LEN1 * 2);
-
-  if (aec->metricsMode == 1) {
-    // Note that the first PART_LEN samples in fft (before transformation) are
-    // zero. Hence, the scaling by two in UpdateLevel() should not be
-    // performed. That scaling is taken care of in UpdateMetrics() instead.
-    UpdateLevel(&aec->linoutlevel, ef);
-  }
-
-  // Scale error signal inversely with far power.
-  WebRtcAec_ScaleErrorSignal(aec, ef);
-  WebRtcAec_FilterAdaptation(aec, fft, ef);
-  NonLinearProcessing(aec, output, outputH_ptr);
+  // Perform echo subtraction.
+  EchoSubtraction(aec,
+                  aec->num_partitions,
+                  aec->xfBufBlockPos,
+                  aec->metricsMode,
+                  aec->extended_filter_enabled,
+                  aec->normal_mu,
+                  aec->normal_error_threshold,
+                  aec->xfBuf,
+                  nearend_ptr,
+                  aec->xPow,
+                  aec->wfBuf,
+                  &aec->linoutlevel,
+                  echo_subtractor_output);
+
+  RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, echo_subtractor_output, PART_LEN);
+
+  // Perform echo suppression.
+  EchoSuppression(aec, farend_ptr, echo_subtractor_output, output, outputH_ptr);
 
   if (aec->metricsMode == 1) {
     // Update power levels and echo metrics
@@ -1383,7 +1418,6 @@ static void ProcessBlock(AecCore* aec) {
     WebRtc_WriteBuffer(aec->outFrBufH[i], outputH[i], PART_LEN);
   }
 
-  RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, e, PART_LEN);
   RTC_AEC_DEBUG_WAV_WRITE(aec->outFile, output, PART_LEN);
 }
 
@@ -1422,26 +1456,20 @@ AecCore* WebRtcAec_CreateAec() {
   }
 
   // Create far-end buffers.
-  aec->far_buf =
-      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
-  if (!aec->far_buf) {
-    WebRtcAec_FreeAec(aec);
-    return NULL;
-  }
-  aec->far_buf_windowed =
-      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
-  if (!aec->far_buf_windowed) {
-    WebRtcAec_FreeAec(aec);
-    return NULL;
-  }
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  aec->instance_index = webrtc_aec_instance_count;
+  // For bit exactness with legacy code, each element in |far_time_buf| is
+  // supposed to contain |PART_LEN2| samples with an overlap of |PART_LEN|
+  // samples from the last frame.
+  // TODO(minyue): reduce |far_time_buf| to non-overlapped |PART_LEN| samples.
   aec->far_time_buf =
-      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN);
+      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN2);
   if (!aec->far_time_buf) {
     WebRtcAec_FreeAec(aec);
     return NULL;
   }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+  aec->instance_index = webrtc_aec_instance_count;
+
   aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = NULL;
   aec->debug_dump_count = 0;
 #endif
@@ -1477,6 +1505,10 @@ AecCore* WebRtcAec_CreateAec() {
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
   WebRtcAec_ComfortNoise = ComfortNoise;
   WebRtcAec_SubbandCoherence = SubbandCoherence;
+  WebRtcAec_StoreAsComplex = StoreAsComplex;
+  WebRtcAec_PartitionDelay = PartitionDelay;
+  WebRtcAec_WindowData = WindowData;
+
 
 #if defined(WEBRTC_ARCH_X86_FAMILY)
   if (WebRtc_GetCPUInfo(kSSE2)) {
@@ -1515,11 +1547,8 @@ void WebRtcAec_FreeAec(AecCore* aec) {
     WebRtc_FreeBuffer(aec->outFrBufH[i]);
   }
 
-  WebRtc_FreeBuffer(aec->far_buf);
-  WebRtc_FreeBuffer(aec->far_buf_windowed);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
   WebRtc_FreeBuffer(aec->far_time_buf);
-#endif
+
   RTC_AEC_DEBUG_WAV_CLOSE(aec->farFile);
   RTC_AEC_DEBUG_WAV_CLOSE(aec->nearFile);
   RTC_AEC_DEBUG_WAV_CLOSE(aec->outFile);
@@ -1555,10 +1584,9 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   }
 
   // Initialize far-end buffers.
-  WebRtc_InitBuffer(aec->far_buf);
-  WebRtc_InitBuffer(aec->far_buf_windowed);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
   WebRtc_InitBuffer(aec->far_time_buf);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
   {
     int process_rate = sampFreq > 16000 ? 16000 : sampFreq;
     RTC_AEC_DEBUG_WAV_REOPEN("aec_far", aec->instance_index,
@@ -1693,6 +1721,8 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   aec->seed = 777;
   aec->delayEstCtr = 0;
 
+  aec->extreme_filter_divergence = 0;
+
   // Metrics disabled by default
   aec->metricsMode = 0;
   InitMetrics(aec);
@@ -1700,27 +1730,22 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   return 0;
 }
 
-void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend) {
-  float fft[PART_LEN2];
-  float xf[2][PART_LEN1];
 
+// For bit exactness with a legacy code, |farend| is supposed to contain
+// |PART_LEN2| samples with an overlap of |PART_LEN| samples from the last
+// frame.
+// TODO(minyue): reduce |farend| to non-overlapped |PART_LEN| samples.
+void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend) {
   // Check if the buffer is full, and in that case flush the oldest data.
-  if (WebRtc_available_write(aec->far_buf) < 1) {
+  if (WebRtc_available_write(aec->far_time_buf) < 1) {
     WebRtcAec_MoveFarReadPtr(aec, 1);
   }
-  // Convert far-end partition to the frequency domain without windowing.
-  memcpy(fft, farend, sizeof(float) * PART_LEN2);
-  TimeToFrequency(fft, xf, 0);
-  WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1);
 
-  // Convert far-end partition to the frequency domain with windowing.
-  memcpy(fft, farend, sizeof(float) * PART_LEN2);
-  TimeToFrequency(fft, xf, 1);
-  WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
+  WebRtc_WriteBuffer(aec->far_time_buf, farend, 1);
 }
 
 int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements) {
-  int elements_moved = MoveFarReadPtrWithoutSystemDelayUpdate(aec, elements);
+  int elements_moved = WebRtc_MoveReadPtr(aec->far_time_buf, elements);
   aec->system_delay -= elements_moved * PART_LEN;
   return elements_moved;
 }
@@ -1794,14 +1819,14 @@ void WebRtcAec_ProcessFrames(AecCore* aec,
       // rounding, like -16.
       int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
       int moved_elements =
-          MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
+          WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
       aec->knownDelay -= moved_elements * PART_LEN;
     } else {
       // 2 b) Apply signal based delay correction.
       int move_elements = SignalBasedDelayCorrection(aec);
       int moved_elements =
-          MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
-      int far_near_buffer_diff = WebRtc_available_read(aec->far_buf) -
+          WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+      int far_near_buffer_diff = WebRtc_available_read(aec->far_time_buf) -
           WebRtc_available_read(aec->nearFrBuf) / PART_LEN;
       WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
       WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
@@ -1880,10 +1905,6 @@ void WebRtcAec_GetEchoStats(AecCore* self,
   *a_nlp = self->aNlp;
 }
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-void* WebRtcAec_far_time_buf(AecCore* self) { return self->far_time_buf; }
-#endif
-
 void WebRtcAec_SetConfigCore(AecCore* self,
                              int nlp_mode,
                              int metrics_mode,
diff --git a/webrtc/modules/audio_processing/aec/aec_core_internal.h b/webrtc/modules/audio_processing/aec/aec_core_internal.h
index 2de028379b..3809c82567 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_internal.h
+++ b/webrtc/modules/audio_processing/aec/aec_core_internal.h
@@ -95,8 +95,8 @@ struct AecCore {
 
   int xfBufBlockPos;
 
-  RingBuffer* far_buf;
-  RingBuffer* far_buf_windowed;
+  RingBuffer* far_time_buf;
+
   int system_delay;  // Current system delay buffered in AEC.
 
   int mult;  // sampling frequency multiple
@@ -152,6 +152,10 @@ struct AecCore {
   // Runtime selection of number of filter partitions.
   int num_partitions;
 
+  // Flag that extreme filter divergence has been detected by the Echo
+  // Suppressor.
+  int extreme_filter_divergence;
+
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   // Sequence number of this AEC instance, so that different instances can
   // choose different dump file names.
@@ -161,7 +165,6 @@ struct AecCore {
   // each time.
   int debug_dump_count;
 
-  RingBuffer* far_time_buf;
   rtc_WavWriter* farFile;
   rtc_WavWriter* nearFile;
   rtc_WavWriter* outFile;
@@ -170,13 +173,25 @@ struct AecCore {
 #endif
 };
 
-typedef void (*WebRtcAecFilterFar)(AecCore* aec, float yf[2][PART_LEN1]);
+typedef void (*WebRtcAecFilterFar)(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]);
 extern WebRtcAecFilterFar WebRtcAec_FilterFar;
-typedef void (*WebRtcAecScaleErrorSignal)(AecCore* aec, float ef[2][PART_LEN1]);
-extern WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
-typedef void (*WebRtcAecFilterAdaptation)(AecCore* aec,
-                                          float* fft,
+typedef void (*WebRtcAecScaleErrorSignal)(int extended_filter_enabled,
+                                          float normal_mu,
+                                          float normal_error_threshold,
+                                          float x_pow[PART_LEN1],
                                           float ef[2][PART_LEN1]);
+extern WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
+typedef void (*WebRtcAecFilterAdaptation)(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
 extern WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
 typedef void (*WebRtcAecOverdriveAndSuppress)(AecCore* aec,
                                               float hNl[PART_LEN1],
@@ -186,17 +201,29 @@ extern WebRtcAecOverdriveAndSuppress WebRtcAec_OverdriveAndSuppress;
 
 typedef void (*WebRtcAecComfortNoise)(AecCore* aec,
                                       float efw[2][PART_LEN1],
-                                      complex_t* comfortNoiseHband,
+                                      float comfortNoiseHband[2][PART_LEN1],
                                       const float* noisePow,
                                       const float* lambda);
 extern WebRtcAecComfortNoise WebRtcAec_ComfortNoise;
 
 typedef void (*WebRtcAecSubBandCoherence)(AecCore* aec,
                                           float efw[2][PART_LEN1],
+                                          float dfw[2][PART_LEN1],
                                           float xfw[2][PART_LEN1],
                                           float* fft,
                                           float* cohde,
-                                          float* cohxd);
+                                          float* cohxd,
+                                          int* extreme_filter_divergence);
 extern WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
 
+typedef int (*WebRtcAecPartitionDelay)(const AecCore* aec);
+extern WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
+
+typedef void (*WebRtcAecStoreAsComplex)(const float* data,
+                                        float data_complex[2][PART_LEN1]);
+extern WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
+
+typedef void (*WebRtcAecWindowData)(float* x_windowed, const float* x);
+extern WebRtcAecWindowData WebRtcAec_WindowData;
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
diff --git a/webrtc/modules/audio_processing/aec/aec_core_mips.c b/webrtc/modules/audio_processing/aec/aec_core_mips.c
index bb33087aee..035a4b76af 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_mips.c
+++ b/webrtc/modules/audio_processing/aec/aec_core_mips.c
@@ -20,13 +20,12 @@
 #include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
 
-static const int flagHbandCn = 1; // flag for adding comfort noise in H band
 extern const float WebRtcAec_weightCurve[65];
 extern const float WebRtcAec_overDriveCurve[65];
 
 void WebRtcAec_ComfortNoise_mips(AecCore* aec,
                                  float efw[2][PART_LEN1],
-                                 complex_t* comfortNoiseHband,
+                                 float comfortNoiseHband[2][PART_LEN1],
                                  const float* noisePow,
                                  const float* lambda) {
   int i, num;
@@ -274,7 +273,7 @@ void WebRtcAec_ComfortNoise_mips(AecCore* aec,
   noiseAvg = 0.0;
   tmpAvg = 0.0;
   num = 0;
-  if ((aec->sampFreq == 32000 || aec->sampFreq == 48000) && flagHbandCn == 1) {
+  if (aec->num_bands > 1) {
     for (i = 0; i < PART_LEN; i++) {
       rand[i] = ((float)randW16[i]) / 32768;
     }
@@ -314,27 +313,35 @@ void WebRtcAec_ComfortNoise_mips(AecCore* aec,
 
     for (i = 0; i < PART_LEN1; i++) {
       // Use average NLP weight for H band
-      comfortNoiseHband[i][0] = tmpAvg * u[i][0];
-      comfortNoiseHband[i][1] = tmpAvg * u[i][1];
+      comfortNoiseHband[0][i] = tmpAvg * u[i][0];
+      comfortNoiseHband[1][i] = tmpAvg * u[i][1];
     }
+  } else {
+    memset(comfortNoiseHband, 0,
+           2 * PART_LEN1 * sizeof(comfortNoiseHband[0][0]));
   }
 }
 
-void WebRtcAec_FilterFar_mips(AecCore* aec, float yf[2][PART_LEN1]) {
+void WebRtcAec_FilterFar_mips(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
   int i;
-  for (i = 0; i < aec->num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+  for (i = 0; i < num_partitions; i++) {
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >=  aec->num_partitions) {
-      xPos -=  aec->num_partitions * (PART_LEN1);
+    if (i + x_fft_buf_block_pos >=  num_partitions) {
+      xPos -=  num_partitions * (PART_LEN1);
     }
-    float* yf0 = yf[0];
-    float* yf1 = yf[1];
-    float* aRe = aec->xfBuf[0] + xPos;
-    float* aIm = aec->xfBuf[1] + xPos;
-    float* bRe = aec->wfBuf[0] + pos;
-    float* bIm = aec->wfBuf[1] + pos;
+    float* yf0 = y_fft[0];
+    float* yf1 = y_fft[1];
+    float* aRe = x_fft_buf[0] + xPos;
+    float* aIm = x_fft_buf[1] + xPos;
+    float* bRe = h_fft_buf[0] + pos;
+    float* bIm = h_fft_buf[1] + pos;
     float f0, f1, f2, f3, f4, f5, f6, f7, f8, f9, f10, f11, f12, f13;
     int len = PART_LEN1 >> 1;
 
@@ -432,23 +439,27 @@ void WebRtcAec_FilterFar_mips(AecCore* aec, float yf[2][PART_LEN1]) {
   }
 }
 
-void WebRtcAec_FilterAdaptation_mips(AecCore* aec,
-                                     float* fft,
-                                     float ef[2][PART_LEN1]) {
+void WebRtcAec_FilterAdaptation_mips(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
+  float fft[PART_LEN2];
   int i;
-  for (i = 0; i < aec->num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
+  for (i = 0; i < num_partitions; i++) {
+    int xPos = (i + x_fft_buf_block_pos)*(PART_LEN1);
     int pos;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= aec->num_partitions * PART_LEN1;
+    if (i + x_fft_buf_block_pos >= num_partitions) {
+      xPos -= num_partitions * PART_LEN1;
     }
 
     pos = i * PART_LEN1;
-    float* aRe = aec->xfBuf[0] + xPos;
-    float* aIm = aec->xfBuf[1] + xPos;
-    float* bRe = ef[0];
-    float* bIm = ef[1];
+    float* aRe = x_fft_buf[0] + xPos;
+    float* aIm = x_fft_buf[1] + xPos;
+    float* bRe = e_fft[0];
+    float* bIm = e_fft[1];
     float* fft_tmp;
 
     float f0, f1, f2, f3, f4, f5, f6 ,f7, f8, f9, f10, f11, f12;
@@ -573,8 +584,8 @@ void WebRtcAec_FilterAdaptation_mips(AecCore* aec,
       );
     }
     aec_rdft_forward_128(fft);
-    aRe = aec->wfBuf[0] + pos;
-    aIm = aec->wfBuf[1] + pos;
+    aRe = h_fft_buf[0] + pos;
+    aIm = h_fft_buf[1] + pos;
     __asm __volatile (
       ".set     push                                    \n\t"
       ".set     noreorder                               \n\t"
@@ -699,15 +710,18 @@ void WebRtcAec_OverdriveAndSuppress_mips(AecCore* aec,
   }
 }
 
-void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
-  const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-  const float error_threshold = aec->extended_filter_enabled
+void WebRtcAec_ScaleErrorSignal_mips(int extended_filter_enabled,
+                                     float normal_mu,
+                                     float normal_error_threshold,
+                                     float x_pow[PART_LEN1],
+                                     float ef[2][PART_LEN1]) {
+  const float mu = extended_filter_enabled ? kExtendedMu : normal_mu;
+  const float error_threshold = extended_filter_enabled
                                     ? kExtendedErrorThreshold
-                                    : aec->normal_error_threshold;
+                                    : normal_error_threshold;
   int len = (PART_LEN1);
   float* ef0 = ef[0];
   float* ef1 = ef[1];
-  float* xPow = aec->xPow;
   float fac1 = 1e-10f;
   float err_th2 = error_threshold * error_threshold;
   float f0, f1, f2;
@@ -719,7 +733,7 @@ void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
     ".set       push                                   \n\t"
     ".set       noreorder                              \n\t"
    "1:                                                 \n\t"
-    "lwc1       %[f0],     0(%[xPow])                  \n\t"
+    "lwc1       %[f0],     0(%[x_pow])                 \n\t"
     "lwc1       %[f1],     0(%[ef0])                   \n\t"
     "lwc1       %[f2],     0(%[ef1])                   \n\t"
     "add.s      %[f0],     %[f0],       %[fac1]        \n\t"
@@ -747,7 +761,7 @@ void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
     "swc1       %[f1],     0(%[ef0])                   \n\t"
     "swc1       %[f2],     0(%[ef1])                   \n\t"
     "addiu      %[len],    %[len],      -1             \n\t"
-    "addiu      %[xPow],   %[xPow],     4              \n\t"
+    "addiu      %[x_pow],  %[x_pow],    4              \n\t"
     "addiu      %[ef0],    %[ef0],      4              \n\t"
     "bgtz       %[len],    1b                          \n\t"
     " addiu     %[ef1],    %[ef1],      4              \n\t"
@@ -756,7 +770,7 @@ void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
 #if !defined(MIPS32_R2_LE)
       [f3] "=&f" (f3),
 #endif
-      [xPow] "+r" (xPow), [ef0] "+r" (ef0), [ef1] "+r" (ef1),
+      [x_pow] "+r" (x_pow), [ef0] "+r" (ef0), [ef1] "+r" (ef1),
       [len] "+r" (len)
     : [fac1] "f" (fac1), [err_th2] "f" (err_th2), [mu] "f" (mu),
       [err_th] "f" (error_threshold)
@@ -771,4 +785,3 @@ void WebRtcAec_InitAec_mips(void) {
   WebRtcAec_ComfortNoise = WebRtcAec_ComfortNoise_mips;
   WebRtcAec_OverdriveAndSuppress = WebRtcAec_OverdriveAndSuppress_mips;
 }
-
diff --git a/webrtc/modules/audio_processing/aec/aec_core_neon.c b/webrtc/modules/audio_processing/aec/aec_core_neon.c
index 9a677aaa67..7898ab2543 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_neon.c
+++ b/webrtc/modules/audio_processing/aec/aec_core_neon.c
@@ -34,45 +34,49 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bIm + aIm * bRe;
 }
 
-static void FilterFarNEON(AecCore* aec, float yf[2][PART_LEN1]) {
+static void FilterFarNEON(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
   int i;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
     int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * PART_LEN1;
     }
 
     // vectorized code (four at once)
     for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const float32x4_t xfBuf_re = vld1q_f32(&aec->xfBuf[0][xPos + j]);
-      const float32x4_t xfBuf_im = vld1q_f32(&aec->xfBuf[1][xPos + j]);
-      const float32x4_t wfBuf_re = vld1q_f32(&aec->wfBuf[0][pos + j]);
-      const float32x4_t wfBuf_im = vld1q_f32(&aec->wfBuf[1][pos + j]);
-      const float32x4_t yf_re = vld1q_f32(&yf[0][j]);
-      const float32x4_t yf_im = vld1q_f32(&yf[1][j]);
-      const float32x4_t a = vmulq_f32(xfBuf_re, wfBuf_re);
-      const float32x4_t e = vmlsq_f32(a, xfBuf_im, wfBuf_im);
-      const float32x4_t c = vmulq_f32(xfBuf_re, wfBuf_im);
-      const float32x4_t f = vmlaq_f32(c, xfBuf_im, wfBuf_re);
-      const float32x4_t g = vaddq_f32(yf_re, e);
-      const float32x4_t h = vaddq_f32(yf_im, f);
-      vst1q_f32(&yf[0][j], g);
-      vst1q_f32(&yf[1][j], h);
+      const float32x4_t x_fft_buf_re = vld1q_f32(&x_fft_buf[0][xPos + j]);
+      const float32x4_t x_fft_buf_im = vld1q_f32(&x_fft_buf[1][xPos + j]);
+      const float32x4_t h_fft_buf_re = vld1q_f32(&h_fft_buf[0][pos + j]);
+      const float32x4_t h_fft_buf_im = vld1q_f32(&h_fft_buf[1][pos + j]);
+      const float32x4_t y_fft_re = vld1q_f32(&y_fft[0][j]);
+      const float32x4_t y_fft_im = vld1q_f32(&y_fft[1][j]);
+      const float32x4_t a = vmulq_f32(x_fft_buf_re, h_fft_buf_re);
+      const float32x4_t e = vmlsq_f32(a, x_fft_buf_im, h_fft_buf_im);
+      const float32x4_t c = vmulq_f32(x_fft_buf_re, h_fft_buf_im);
+      const float32x4_t f = vmlaq_f32(c, x_fft_buf_im, h_fft_buf_re);
+      const float32x4_t g = vaddq_f32(y_fft_re, e);
+      const float32x4_t h = vaddq_f32(y_fft_im, f);
+      vst1q_f32(&y_fft[0][j], g);
+      vst1q_f32(&y_fft[1][j], h);
     }
     // scalar code for the remaining items.
     for (; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
+      y_fft[0][j] += MulRe(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
+      y_fft[1][j] += MulIm(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
     }
   }
 }
@@ -122,20 +126,24 @@ static float32x4_t vsqrtq_f32(float32x4_t s) {
 }
 #endif  // WEBRTC_ARCH_ARM64
 
-static void ScaleErrorSignalNEON(AecCore* aec, float ef[2][PART_LEN1]) {
-  const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-  const float error_threshold = aec->extended_filter_enabled ?
-      kExtendedErrorThreshold : aec->normal_error_threshold;
+static void ScaleErrorSignalNEON(int extended_filter_enabled,
+                                 float normal_mu,
+                                 float normal_error_threshold,
+                                 float x_pow[PART_LEN1],
+                                 float ef[2][PART_LEN1]) {
+  const float mu = extended_filter_enabled ? kExtendedMu : normal_mu;
+  const float error_threshold = extended_filter_enabled ?
+      kExtendedErrorThreshold : normal_error_threshold;
   const float32x4_t k1e_10f = vdupq_n_f32(1e-10f);
   const float32x4_t kMu = vmovq_n_f32(mu);
   const float32x4_t kThresh = vmovq_n_f32(error_threshold);
   int i;
   // vectorized code (four at once)
   for (i = 0; i + 3 < PART_LEN1; i += 4) {
-    const float32x4_t xPow = vld1q_f32(&aec->xPow[i]);
+    const float32x4_t x_pow_local = vld1q_f32(&x_pow[i]);
     const float32x4_t ef_re_base = vld1q_f32(&ef[0][i]);
     const float32x4_t ef_im_base = vld1q_f32(&ef[1][i]);
-    const float32x4_t xPowPlus = vaddq_f32(xPow, k1e_10f);
+    const float32x4_t xPowPlus = vaddq_f32(x_pow_local, k1e_10f);
     float32x4_t ef_re = vdivq_f32(ef_re_base, xPowPlus);
     float32x4_t ef_im = vdivq_f32(ef_im_base, xPowPlus);
     const float32x4_t ef_re2 = vmulq_f32(ef_re, ef_re);
@@ -162,8 +170,8 @@ static void ScaleErrorSignalNEON(AecCore* aec, float ef[2][PART_LEN1]) {
   // scalar code for the remaining items.
   for (; i < PART_LEN1; i++) {
     float abs_ef;
-    ef[0][i] /= (aec->xPow[i] + 1e-10f);
-    ef[1][i] /= (aec->xPow[i] + 1e-10f);
+    ef[0][i] /= (x_pow[i] + 1e-10f);
+    ef[1][i] /= (x_pow[i] + 1e-10f);
     abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
     if (abs_ef > error_threshold) {
@@ -178,34 +186,37 @@ static void ScaleErrorSignalNEON(AecCore* aec, float ef[2][PART_LEN1]) {
   }
 }
 
-static void FilterAdaptationNEON(AecCore* aec,
-                                 float* fft,
-                                 float ef[2][PART_LEN1]) {
+static void FilterAdaptationNEON(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
+  float fft[PART_LEN2];
   int i;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     int j;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * PART_LEN1;
     }
 
     // Process the whole array...
     for (j = 0; j < PART_LEN; j += 4) {
-      // Load xfBuf and ef.
-      const float32x4_t xfBuf_re = vld1q_f32(&aec->xfBuf[0][xPos + j]);
-      const float32x4_t xfBuf_im = vld1q_f32(&aec->xfBuf[1][xPos + j]);
-      const float32x4_t ef_re = vld1q_f32(&ef[0][j]);
-      const float32x4_t ef_im = vld1q_f32(&ef[1][j]);
-      // Calculate the product of conjugate(xfBuf) by ef.
+      // Load x_fft_buf and e_fft.
+      const float32x4_t x_fft_buf_re = vld1q_f32(&x_fft_buf[0][xPos + j]);
+      const float32x4_t x_fft_buf_im = vld1q_f32(&x_fft_buf[1][xPos + j]);
+      const float32x4_t e_fft_re = vld1q_f32(&e_fft[0][j]);
+      const float32x4_t e_fft_im = vld1q_f32(&e_fft[1][j]);
+      // Calculate the product of conjugate(x_fft_buf) by e_fft.
       //   re(conjugate(a) * b) = aRe * bRe + aIm * bIm
       //   im(conjugate(a) * b)=  aRe * bIm - aIm * bRe
-      const float32x4_t a = vmulq_f32(xfBuf_re, ef_re);
-      const float32x4_t e = vmlaq_f32(a, xfBuf_im, ef_im);
-      const float32x4_t c = vmulq_f32(xfBuf_re, ef_im);
-      const float32x4_t f = vmlsq_f32(c, xfBuf_im, ef_re);
+      const float32x4_t a = vmulq_f32(x_fft_buf_re, e_fft_re);
+      const float32x4_t e = vmlaq_f32(a, x_fft_buf_im, e_fft_im);
+      const float32x4_t c = vmulq_f32(x_fft_buf_re, e_fft_im);
+      const float32x4_t f = vmlsq_f32(c, x_fft_buf_im, e_fft_re);
       // Interleave real and imaginary parts.
       const float32x4x2_t g_n_h = vzipq_f32(e, f);
       // Store
@@ -213,10 +224,10 @@ static void FilterAdaptationNEON(AecCore* aec,
       vst1q_f32(&fft[2 * j + 4], g_n_h.val[1]);
     }
     // ... and fixup the first imaginary entry.
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN],
-                   ef[1][PART_LEN]);
+    fft[1] = MulRe(x_fft_buf[0][xPos + PART_LEN],
+                   -x_fft_buf[1][xPos + PART_LEN],
+                   e_fft[0][PART_LEN],
+                   e_fft[1][PART_LEN]);
 
     aec_rdft_inverse_128(fft);
     memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
@@ -234,21 +245,21 @@ static void FilterAdaptationNEON(AecCore* aec,
     aec_rdft_forward_128(fft);
 
     {
-      const float wt1 = aec->wfBuf[1][pos];
-      aec->wfBuf[0][pos + PART_LEN] += fft[1];
+      const float wt1 = h_fft_buf[1][pos];
+      h_fft_buf[0][pos + PART_LEN] += fft[1];
       for (j = 0; j < PART_LEN; j += 4) {
-        float32x4_t wtBuf_re = vld1q_f32(&aec->wfBuf[0][pos + j]);
-        float32x4_t wtBuf_im = vld1q_f32(&aec->wfBuf[1][pos + j]);
+        float32x4_t wtBuf_re = vld1q_f32(&h_fft_buf[0][pos + j]);
+        float32x4_t wtBuf_im = vld1q_f32(&h_fft_buf[1][pos + j]);
         const float32x4_t fft0 = vld1q_f32(&fft[2 * j + 0]);
         const float32x4_t fft4 = vld1q_f32(&fft[2 * j + 4]);
         const float32x4x2_t fft_re_im = vuzpq_f32(fft0, fft4);
         wtBuf_re = vaddq_f32(wtBuf_re, fft_re_im.val[0]);
         wtBuf_im = vaddq_f32(wtBuf_im, fft_re_im.val[1]);
 
-        vst1q_f32(&aec->wfBuf[0][pos + j], wtBuf_re);
-        vst1q_f32(&aec->wfBuf[1][pos + j], wtBuf_im);
+        vst1q_f32(&h_fft_buf[0][pos + j], wtBuf_re);
+        vst1q_f32(&h_fft_buf[1][pos + j], wtBuf_im);
       }
-      aec->wfBuf[1][pos] = wt1;
+      h_fft_buf[1][pos] = wt1;
     }
   }
 }
@@ -442,7 +453,7 @@ static void OverdriveAndSuppressNEON(AecCore* aec,
   }
 }
 
-static int PartitionDelay(const AecCore* aec) {
+static int PartitionDelayNEON(const AecCore* aec) {
   // Measures the energy in each filter partition and returns the partition with
   // highest energy.
   // TODO(bjornv): Spread computational cost by computing one partition per
@@ -499,7 +510,8 @@ static int PartitionDelay(const AecCore* aec) {
 static void SmoothedPSD(AecCore* aec,
                         float efw[2][PART_LEN1],
                         float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1]) {
+                        float xfw[2][PART_LEN1],
+                        int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh = aec->extended_filter_enabled
       ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
@@ -615,19 +627,16 @@ static void SmoothedPSD(AecCore* aec,
     seSum += aec->se[i];
   }
 
-  // Divergent filter safeguard.
+  // Divergent filter safeguard update.
   aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
 
-  if (aec->divergeState)
-    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
-
-  // Reset if error is significantly larger than nearend (13 dB).
-  if (!aec->extended_filter_enabled && seSum > (19.95f * sdSum))
-    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+  // Signal extreme filter divergence if the error is significantly larger
+  // than the nearend (13 dB).
+  *extreme_filter_divergence = (seSum > (19.95f * sdSum));
 }
 
 // Window time domain data to be used by the fft.
-__inline static void WindowData(float* x_windowed, const float* x) {
+static void WindowDataNEON(float* x_windowed, const float* x) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const float32x4_t vec_Buf1 = vld1q_f32(&x[i]);
@@ -648,8 +657,8 @@ __inline static void WindowData(float* x_windowed, const float* x) {
 }
 
 // Puts fft output data into a complex valued array.
-__inline static void StoreAsComplex(const float* data,
-                                    float data_complex[2][PART_LEN1]) {
+static void StoreAsComplexNEON(const float* data,
+                               float data_complex[2][PART_LEN1]) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const float32x4x2_t vec_data = vld2q_f32(&data[2 * i]);
@@ -665,32 +674,15 @@ __inline static void StoreAsComplex(const float* data,
 
 static void SubbandCoherenceNEON(AecCore* aec,
                                  float efw[2][PART_LEN1],
+                                 float dfw[2][PART_LEN1],
                                  float xfw[2][PART_LEN1],
                                  float* fft,
                                  float* cohde,
-                                 float* cohxd) {
-  float dfw[2][PART_LEN1];
+                                 float* cohxd,
+                                 int* extreme_filter_divergence) {
   int i;
 
-  if (aec->delayEstCtr == 0)
-    aec->delayIdx = PartitionDelay(aec);
-
-  // Use delayed far.
-  memcpy(xfw,
-         aec->xfwBuf + aec->delayIdx * PART_LEN1,
-         sizeof(xfw[0][0]) * 2 * PART_LEN1);
-
-  // Windowed near fft
-  WindowData(fft, aec->dBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, dfw);
-
-  // Windowed error fft
-  WindowData(fft, aec->eBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, efw);
-
-  SmoothedPSD(aec, efw, dfw, xfw);
+  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
 
   {
     const float32x4_t vec_1eminus10 =  vdupq_n_f32(1e-10f);
@@ -732,5 +724,7 @@ void WebRtcAec_InitAec_neon(void) {
   WebRtcAec_FilterAdaptation = FilterAdaptationNEON;
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressNEON;
   WebRtcAec_SubbandCoherence = SubbandCoherenceNEON;
+  WebRtcAec_StoreAsComplex = StoreAsComplexNEON;
+  WebRtcAec_PartitionDelay = PartitionDelayNEON;
+  WebRtcAec_WindowData = WindowDataNEON;
 }
-
diff --git a/webrtc/modules/audio_processing/aec/aec_core_sse2.c b/webrtc/modules/audio_processing/aec/aec_core_sse2.c
index b1bffcbb9f..f897a4c0c7 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_sse2.c
+++ b/webrtc/modules/audio_processing/aec/aec_core_sse2.c
@@ -29,67 +29,76 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bIm + aIm * bRe;
 }
 
-static void FilterFarSSE2(AecCore* aec, float yf[2][PART_LEN1]) {
+static void FilterFarSSE2(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
+
   int i;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
     int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * (PART_LEN1);
     }
 
     // vectorized code (four at once)
     for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 wfBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-      const __m128 wfBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
-      const __m128 yf_re = _mm_loadu_ps(&yf[0][j]);
-      const __m128 yf_im = _mm_loadu_ps(&yf[1][j]);
-      const __m128 a = _mm_mul_ps(xfBuf_re, wfBuf_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, wfBuf_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, wfBuf_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, wfBuf_re);
+      const __m128 x_fft_buf_re = _mm_loadu_ps(&x_fft_buf[0][xPos + j]);
+      const __m128 x_fft_buf_im = _mm_loadu_ps(&x_fft_buf[1][xPos + j]);
+      const __m128 h_fft_buf_re = _mm_loadu_ps(&h_fft_buf[0][pos + j]);
+      const __m128 h_fft_buf_im = _mm_loadu_ps(&h_fft_buf[1][pos + j]);
+      const __m128 y_fft_re = _mm_loadu_ps(&y_fft[0][j]);
+      const __m128 y_fft_im = _mm_loadu_ps(&y_fft[1][j]);
+      const __m128 a = _mm_mul_ps(x_fft_buf_re, h_fft_buf_re);
+      const __m128 b = _mm_mul_ps(x_fft_buf_im, h_fft_buf_im);
+      const __m128 c = _mm_mul_ps(x_fft_buf_re, h_fft_buf_im);
+      const __m128 d = _mm_mul_ps(x_fft_buf_im, h_fft_buf_re);
       const __m128 e = _mm_sub_ps(a, b);
       const __m128 f = _mm_add_ps(c, d);
-      const __m128 g = _mm_add_ps(yf_re, e);
-      const __m128 h = _mm_add_ps(yf_im, f);
-      _mm_storeu_ps(&yf[0][j], g);
-      _mm_storeu_ps(&yf[1][j], h);
+      const __m128 g = _mm_add_ps(y_fft_re, e);
+      const __m128 h = _mm_add_ps(y_fft_im, f);
+      _mm_storeu_ps(&y_fft[0][j], g);
+      _mm_storeu_ps(&y_fft[1][j], h);
     }
     // scalar code for the remaining items.
     for (; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
+      y_fft[0][j] += MulRe(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
+      y_fft[1][j] += MulIm(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
     }
   }
 }
 
-static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1]) {
+static void ScaleErrorSignalSSE2(int extended_filter_enabled,
+                                 float normal_mu,
+                                 float normal_error_threshold,
+                                 float x_pow[PART_LEN1],
+                                 float ef[2][PART_LEN1]) {
   const __m128 k1e_10f = _mm_set1_ps(1e-10f);
-  const __m128 kMu = aec->extended_filter_enabled ? _mm_set1_ps(kExtendedMu)
-                                                  : _mm_set1_ps(aec->normal_mu);
-  const __m128 kThresh = aec->extended_filter_enabled
+  const __m128 kMu = extended_filter_enabled ? _mm_set1_ps(kExtendedMu)
+      : _mm_set1_ps(normal_mu);
+  const __m128 kThresh = extended_filter_enabled
                              ? _mm_set1_ps(kExtendedErrorThreshold)
-                             : _mm_set1_ps(aec->normal_error_threshold);
+                             : _mm_set1_ps(normal_error_threshold);
 
   int i;
   // vectorized code (four at once)
   for (i = 0; i + 3 < PART_LEN1; i += 4) {
-    const __m128 xPow = _mm_loadu_ps(&aec->xPow[i]);
+    const __m128 x_pow_local = _mm_loadu_ps(&x_pow[i]);
     const __m128 ef_re_base = _mm_loadu_ps(&ef[0][i]);
     const __m128 ef_im_base = _mm_loadu_ps(&ef[1][i]);
 
-    const __m128 xPowPlus = _mm_add_ps(xPow, k1e_10f);
+    const __m128 xPowPlus = _mm_add_ps(x_pow_local, k1e_10f);
     __m128 ef_re = _mm_div_ps(ef_re_base, xPowPlus);
     __m128 ef_im = _mm_div_ps(ef_im_base, xPowPlus);
     const __m128 ef_re2 = _mm_mul_ps(ef_re, ef_re);
@@ -116,14 +125,14 @@ static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1]) {
   // scalar code for the remaining items.
   {
     const float mu =
-        aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-    const float error_threshold = aec->extended_filter_enabled
+        extended_filter_enabled ? kExtendedMu : normal_mu;
+    const float error_threshold = extended_filter_enabled
                                       ? kExtendedErrorThreshold
-                                      : aec->normal_error_threshold;
+                                      : normal_error_threshold;
     for (; i < (PART_LEN1); i++) {
       float abs_ef;
-      ef[0][i] /= (aec->xPow[i] + 1e-10f);
-      ef[1][i] /= (aec->xPow[i] + 1e-10f);
+      ef[0][i] /= (x_pow[i] + 1e-10f);
+      ef[1][i] /= (x_pow[i] + 1e-10f);
       abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
       if (abs_ef > error_threshold) {
@@ -139,33 +148,36 @@ static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1]) {
   }
 }
 
-static void FilterAdaptationSSE2(AecCore* aec,
-                                 float* fft,
-                                 float ef[2][PART_LEN1]) {
+static void FilterAdaptationSSE2(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
+  float fft[PART_LEN2];
   int i, j;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * (PART_LEN1);
+    int xPos = (i + x_fft_buf_block_pos) * (PART_LEN1);
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * PART_LEN1;
     }
 
     // Process the whole array...
     for (j = 0; j < PART_LEN; j += 4) {
-      // Load xfBuf and ef.
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 ef_re = _mm_loadu_ps(&ef[0][j]);
-      const __m128 ef_im = _mm_loadu_ps(&ef[1][j]);
-      // Calculate the product of conjugate(xfBuf) by ef.
+      // Load x_fft_buf and e_fft.
+      const __m128 x_fft_buf_re = _mm_loadu_ps(&x_fft_buf[0][xPos + j]);
+      const __m128 x_fft_buf_im = _mm_loadu_ps(&x_fft_buf[1][xPos + j]);
+      const __m128 e_fft_re = _mm_loadu_ps(&e_fft[0][j]);
+      const __m128 e_fft_im = _mm_loadu_ps(&e_fft[1][j]);
+      // Calculate the product of conjugate(x_fft_buf) by e_fft.
       //   re(conjugate(a) * b) = aRe * bRe + aIm * bIm
       //   im(conjugate(a) * b)=  aRe * bIm - aIm * bRe
-      const __m128 a = _mm_mul_ps(xfBuf_re, ef_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, ef_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, ef_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, ef_re);
+      const __m128 a = _mm_mul_ps(x_fft_buf_re, e_fft_re);
+      const __m128 b = _mm_mul_ps(x_fft_buf_im, e_fft_im);
+      const __m128 c = _mm_mul_ps(x_fft_buf_re, e_fft_im);
+      const __m128 d = _mm_mul_ps(x_fft_buf_im, e_fft_re);
       const __m128 e = _mm_add_ps(a, b);
       const __m128 f = _mm_sub_ps(c, d);
       // Interleave real and imaginary parts.
@@ -176,10 +188,10 @@ static void FilterAdaptationSSE2(AecCore* aec,
       _mm_storeu_ps(&fft[2 * j + 4], h);
     }
     // ... and fixup the first imaginary entry.
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN],
-                   ef[1][PART_LEN]);
+    fft[1] = MulRe(x_fft_buf[0][xPos + PART_LEN],
+                   -x_fft_buf[1][xPos + PART_LEN],
+                   e_fft[0][PART_LEN],
+                   e_fft[1][PART_LEN]);
 
     aec_rdft_inverse_128(fft);
     memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
@@ -197,11 +209,11 @@ static void FilterAdaptationSSE2(AecCore* aec,
     aec_rdft_forward_128(fft);
 
     {
-      float wt1 = aec->wfBuf[1][pos];
-      aec->wfBuf[0][pos + PART_LEN] += fft[1];
+      float wt1 = h_fft_buf[1][pos];
+      h_fft_buf[0][pos + PART_LEN] += fft[1];
       for (j = 0; j < PART_LEN; j += 4) {
-        __m128 wtBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-        __m128 wtBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
+        __m128 wtBuf_re = _mm_loadu_ps(&h_fft_buf[0][pos + j]);
+        __m128 wtBuf_im = _mm_loadu_ps(&h_fft_buf[1][pos + j]);
         const __m128 fft0 = _mm_loadu_ps(&fft[2 * j + 0]);
         const __m128 fft4 = _mm_loadu_ps(&fft[2 * j + 4]);
         const __m128 fft_re =
@@ -210,10 +222,10 @@ static void FilterAdaptationSSE2(AecCore* aec,
             _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(3, 1, 3, 1));
         wtBuf_re = _mm_add_ps(wtBuf_re, fft_re);
         wtBuf_im = _mm_add_ps(wtBuf_im, fft_im);
-        _mm_storeu_ps(&aec->wfBuf[0][pos + j], wtBuf_re);
-        _mm_storeu_ps(&aec->wfBuf[1][pos + j], wtBuf_im);
+        _mm_storeu_ps(&h_fft_buf[0][pos + j], wtBuf_re);
+        _mm_storeu_ps(&h_fft_buf[1][pos + j], wtBuf_im);
       }
-      aec->wfBuf[1][pos] = wt1;
+      h_fft_buf[1][pos] = wt1;
     }
   }
 }
@@ -427,7 +439,8 @@ __inline static void _mm_add_ps_4x1(__m128 sum, float *dst) {
   sum = _mm_add_ps(sum, _mm_shuffle_ps(sum, sum, _MM_SHUFFLE(1, 1, 1, 1)));
   _mm_store_ss(dst, sum);
 }
-static int PartitionDelay(const AecCore* aec) {
+
+static int PartitionDelaySSE2(const AecCore* aec) {
   // Measures the energy in each filter partition and returns the partition with
   // highest energy.
   // TODO(bjornv): Spread computational cost by computing one partition per
@@ -476,7 +489,8 @@ static int PartitionDelay(const AecCore* aec) {
 static void SmoothedPSD(AecCore* aec,
                         float efw[2][PART_LEN1],
                         float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1]) {
+                        float xfw[2][PART_LEN1],
+                        int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh = aec->extended_filter_enabled
       ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
@@ -595,19 +609,16 @@ static void SmoothedPSD(AecCore* aec,
     seSum += aec->se[i];
   }
 
-  // Divergent filter safeguard.
+  // Divergent filter safeguard update.
   aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
 
-  if (aec->divergeState)
-    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
-
-  // Reset if error is significantly larger than nearend (13 dB).
-  if (!aec->extended_filter_enabled && seSum > (19.95f * sdSum))
-    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+  // Signal extreme filter divergence if the error is significantly larger
+  // than the nearend (13 dB).
+  *extreme_filter_divergence = (seSum > (19.95f * sdSum));
 }
 
 // Window time domain data to be used by the fft.
-__inline static void WindowData(float* x_windowed, const float* x) {
+static void WindowDataSSE2(float* x_windowed, const float* x) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const __m128 vec_Buf1 = _mm_loadu_ps(&x[i]);
@@ -627,8 +638,8 @@ __inline static void WindowData(float* x_windowed, const float* x) {
 }
 
 // Puts fft output data into a complex valued array.
-__inline static void StoreAsComplex(const float* data,
-                                    float data_complex[2][PART_LEN1]) {
+static void StoreAsComplexSSE2(const float* data,
+                               float data_complex[2][PART_LEN1]) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const __m128 vec_fft0 = _mm_loadu_ps(&data[2 * i]);
@@ -649,32 +660,15 @@ __inline static void StoreAsComplex(const float* data,
 
 static void SubbandCoherenceSSE2(AecCore* aec,
                                  float efw[2][PART_LEN1],
+                                 float dfw[2][PART_LEN1],
                                  float xfw[2][PART_LEN1],
                                  float* fft,
                                  float* cohde,
-                                 float* cohxd) {
-  float dfw[2][PART_LEN1];
+                                 float* cohxd,
+                                 int* extreme_filter_divergence) {
   int i;
 
-  if (aec->delayEstCtr == 0)
-    aec->delayIdx = PartitionDelay(aec);
-
-  // Use delayed far.
-  memcpy(xfw,
-         aec->xfwBuf + aec->delayIdx * PART_LEN1,
-         sizeof(xfw[0][0]) * 2 * PART_LEN1);
-
-  // Windowed near fft
-  WindowData(fft, aec->dBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, dfw);
-
-  // Windowed error fft
-  WindowData(fft, aec->eBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, efw);
-
-  SmoothedPSD(aec, efw, dfw, xfw);
+  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
 
   {
     const __m128 vec_1eminus10 =  _mm_set1_ps(1e-10f);
@@ -728,4 +722,7 @@ void WebRtcAec_InitAec_SSE2(void) {
   WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
   WebRtcAec_SubbandCoherence = SubbandCoherenceSSE2;
+  WebRtcAec_StoreAsComplex = StoreAsComplexSSE2;
+  WebRtcAec_PartitionDelay = PartitionDelaySSE2;
+  WebRtcAec_WindowData = WindowDataSSE2;
 }
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.c b/webrtc/modules/audio_processing/aec/echo_cancellation.c
index 0f5cd31ddb..aab1718b24 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.c
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.c
@@ -11,7 +11,7 @@
 /*
  * Contains the API functions for the AEC.
  */
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 
 #include <math.h>
 #ifdef WEBRTC_AEC_DEBUG_DUMP
@@ -146,7 +146,6 @@ void* WebRtcAec_Create() {
   }
 
   aecpc->initFlag = 0;
-  aecpc->lastError = 0;
 
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   {
@@ -192,26 +191,22 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) {
       sampFreq != 16000 &&
       sampFreq != 32000 &&
       sampFreq != 48000) {
-    aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
+    return AEC_BAD_PARAMETER_ERROR;
   }
   aecpc->sampFreq = sampFreq;
 
   if (scSampFreq < 1 || scSampFreq > 96000) {
-    aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
+    return AEC_BAD_PARAMETER_ERROR;
   }
   aecpc->scSampFreq = scSampFreq;
 
   // Initialize echo canceller core
   if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
-    aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-    return -1;
+    return AEC_UNSPECIFIED_ERROR;
   }
 
   if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
-    aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-    return -1;
+    return AEC_UNSPECIFIED_ERROR;
   }
 
   WebRtc_InitBuffer(aecpc->far_pre_buf);
@@ -261,13 +256,32 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) {
   aecConfig.delay_logging = kAecFalse;
 
   if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
-    aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-    return -1;
+    return AEC_UNSPECIFIED_ERROR;
   }
 
   return 0;
 }
 
+// Returns any error that is caused when buffering the
+// far-end signal.
+int32_t WebRtcAec_GetBufferFarendError(void* aecInst,
+                                       const float* farend,
+                                       size_t nrOfSamples) {
+  Aec* aecpc = aecInst;
+
+  if (!farend)
+    return AEC_NULL_POINTER_ERROR;
+
+  if (aecpc->initFlag != initCheck)
+    return AEC_UNINITIALIZED_ERROR;
+
+  // number of samples == 160 for SWB input
+  if (nrOfSamples != 80 && nrOfSamples != 160)
+    return AEC_BAD_PARAMETER_ERROR;
+
+  return 0;
+}
+
 // only buffer L band for farend
 int32_t WebRtcAec_BufferFarend(void* aecInst,
                                const float* farend,
@@ -277,21 +291,13 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
   float new_farend[MAX_RESAMP_LEN];
   const float* farend_ptr = farend;
 
-  if (farend == NULL) {
-    aecpc->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
-  }
+  // Get any error caused by buffering the farend signal.
+  int32_t error_code = WebRtcAec_GetBufferFarendError(aecInst, farend,
+                                                      nrOfSamples);
 
-  if (aecpc->initFlag != initCheck) {
-    aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-    return -1;
-  }
+  if (error_code != 0)
+    return error_code;
 
-  // number of samples == 160 for SWB input
-  if (nrOfSamples != 80 && nrOfSamples != 160) {
-    aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
-  }
 
   if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
     // Resample and get a new number of samples
@@ -311,7 +317,8 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
   // Write the time-domain data to |far_pre_buf|.
   WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, newNrOfSamples);
 
-  // Transform to frequency domain if we have enough data.
+  // TODO(minyue): reduce to |PART_LEN| samples for each buffering, when
+  // WebRtcAec_BufferFarendPartition() is changed to take |PART_LEN| samples.
   while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) {
     // We have enough data to pass to the FFT, hence read PART_LEN2 samples.
     {
@@ -319,10 +326,6 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
       float tmp[PART_LEN2];
       WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**)&ptmp, tmp, PART_LEN2);
       WebRtcAec_BufferFarendPartition(aecpc->aec, ptmp);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-      WebRtc_WriteBuffer(
-          WebRtcAec_far_time_buf(aecpc->aec), &ptmp[PART_LEN], 1);
-#endif
     }
 
     // Rewind |far_pre_buf| PART_LEN samples for overlap before continuing.
@@ -343,29 +346,24 @@ int32_t WebRtcAec_Process(void* aecInst,
   int32_t retVal = 0;
 
   if (out == NULL) {
-    aecpc->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
+    return AEC_NULL_POINTER_ERROR;
   }
 
   if (aecpc->initFlag != initCheck) {
-    aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-    return -1;
+    return AEC_UNINITIALIZED_ERROR;
   }
 
   // number of samples == 160 for SWB input
   if (nrOfSamples != 80 && nrOfSamples != 160) {
-    aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
+    return AEC_BAD_PARAMETER_ERROR;
   }
 
   if (msInSndCardBuf < 0) {
     msInSndCardBuf = 0;
-    aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-    retVal = -1;
+    retVal = AEC_BAD_PARAMETER_WARNING;
   } else if (msInSndCardBuf > kMaxTrustedDelayMs) {
     // The clamping is now done in ProcessExtended/Normal().
-    aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-    retVal = -1;
+    retVal = AEC_BAD_PARAMETER_WARNING;
   }
 
   // This returns the value of aec->extended_filter_enabled.
@@ -378,15 +376,13 @@ int32_t WebRtcAec_Process(void* aecInst,
                     msInSndCardBuf,
                     skew);
   } else {
-    if (ProcessNormal(aecpc,
-                      nearend,
-                      num_bands,
-                      out,
-                      nrOfSamples,
-                      msInSndCardBuf,
-                      skew) != 0) {
-      retVal = -1;
-    }
+    retVal = ProcessNormal(aecpc,
+                           nearend,
+                           num_bands,
+                           out,
+                           nrOfSamples,
+                           msInSndCardBuf,
+                           skew);
   }
 
 #ifdef WEBRTC_AEC_DEBUG_DUMP
@@ -405,31 +401,26 @@ int32_t WebRtcAec_Process(void* aecInst,
 int WebRtcAec_set_config(void* handle, AecConfig config) {
   Aec* self = (Aec*)handle;
   if (self->initFlag != initCheck) {
-    self->lastError = AEC_UNINITIALIZED_ERROR;
-    return -1;
+    return AEC_UNINITIALIZED_ERROR;
   }
 
   if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
-    self->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
+    return AEC_BAD_PARAMETER_ERROR;
   }
   self->skewMode = config.skewMode;
 
   if (config.nlpMode != kAecNlpConservative &&
       config.nlpMode != kAecNlpModerate &&
       config.nlpMode != kAecNlpAggressive) {
-    self->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
+    return AEC_BAD_PARAMETER_ERROR;
   }
 
   if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
-    self->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
+    return AEC_BAD_PARAMETER_ERROR;
   }
 
   if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) {
-    self->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
+    return AEC_BAD_PARAMETER_ERROR;
   }
 
   WebRtcAec_SetConfigCore(
@@ -440,12 +431,10 @@ int WebRtcAec_set_config(void* handle, AecConfig config) {
 int WebRtcAec_get_echo_status(void* handle, int* status) {
   Aec* self = (Aec*)handle;
   if (status == NULL) {
-    self->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
+    return AEC_NULL_POINTER_ERROR;
   }
   if (self->initFlag != initCheck) {
-    self->lastError = AEC_UNINITIALIZED_ERROR;
-    return -1;
+    return AEC_UNINITIALIZED_ERROR;
   }
 
   *status = WebRtcAec_echo_state(self->aec);
@@ -466,12 +455,10 @@ int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics) {
     return -1;
   }
   if (metrics == NULL) {
-    self->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
+    return AEC_NULL_POINTER_ERROR;
   }
   if (self->initFlag != initCheck) {
-    self->lastError = AEC_UNINITIALIZED_ERROR;
-    return -1;
+    return AEC_UNINITIALIZED_ERROR;
   }
 
   WebRtcAec_GetEchoStats(self->aec, &erl, &erle, &a_nlp);
@@ -556,32 +543,24 @@ int WebRtcAec_GetDelayMetrics(void* handle,
                               float* fraction_poor_delays) {
   Aec* self = handle;
   if (median == NULL) {
-    self->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
+    return AEC_NULL_POINTER_ERROR;
   }
   if (std == NULL) {
-    self->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
+    return AEC_NULL_POINTER_ERROR;
   }
   if (self->initFlag != initCheck) {
-    self->lastError = AEC_UNINITIALIZED_ERROR;
-    return -1;
+    return AEC_UNINITIALIZED_ERROR;
   }
   if (WebRtcAec_GetDelayMetricsCore(self->aec, median, std,
                                     fraction_poor_delays) ==
       -1) {
     // Logging disabled.
-    self->lastError = AEC_UNSUPPORTED_FUNCTION_ERROR;
-    return -1;
+    return AEC_UNSUPPORTED_FUNCTION_ERROR;
   }
 
   return 0;
 }
 
-int32_t WebRtcAec_get_error_code(void* aecInst) {
-  Aec* aecpc = aecInst;
-  return aecpc->lastError;
-}
 
 AecCore* WebRtcAec_aec_core(void* handle) {
   if (!handle) {
@@ -617,7 +596,7 @@ static int ProcessNormal(Aec* aecpc,
       retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
       if (retVal == -1) {
         aecpc->skew = 0;
-        aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+        retVal = AEC_BAD_PARAMETER_WARNING;
       }
 
       aecpc->skew /= aecpc->sampFactor * nrOfSamples;
diff --git a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h b/webrtc/modules/audio_processing/aec/echo_cancellation.h
index a340cf84d0..de84b2e6d1 100644
--- a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
 
 #include <stddef.h>
 
@@ -109,13 +109,32 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq);
  * Outputs                      Description
  * -------------------------------------------------------------------
  * int32_t        return        0: OK
- *                             -1: error
+ *                              12000-12050: error code
  */
 int32_t WebRtcAec_BufferFarend(void* aecInst,
                                const float* farend,
                                size_t nrOfSamples);
 
 /*
+ * Reports any errors that would arise if buffering a farend buffer
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*          aecInst       Pointer to the AEC instance
+ * const float*   farend        In buffer containing one frame of
+ *                              farend signal for L band
+ * int16_t        nrOfSamples   Number of samples in farend buffer
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t        return        0: OK
+ *                              12000-12050: error code
+ */
+int32_t WebRtcAec_GetBufferFarendError(void* aecInst,
+                                       const float* farend,
+                                       size_t nrOfSamples);
+
+/*
  * Runs the echo canceller on an 80 or 160 sample blocks of data.
  *
  * Inputs                       Description
@@ -136,7 +155,7 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
  * float* const* out            Out buffer, one frame of processed nearend
  *                              for each band
  * int32_t       return         0: OK
- *                             -1: error
+ *                              12000-12050: error code
  */
 int32_t WebRtcAec_Process(void* aecInst,
                           const float* const* nearend,
@@ -157,8 +176,8 @@ int32_t WebRtcAec_Process(void* aecInst,
  *
  * Outputs                      Description
  * -------------------------------------------------------------------
- * int            return         0: OK
- *                              -1: error
+ * int            return        0: OK
+ *                              12000-12050: error code
  */
 int WebRtcAec_set_config(void* handle, AecConfig config);
 
@@ -173,8 +192,8 @@ int WebRtcAec_set_config(void* handle, AecConfig config);
  * -------------------------------------------------------------------
  * int*           status        0: Almost certainly nearend single-talk
  *                              1: Might not be neared single-talk
- * int            return         0: OK
- *                              -1: error
+ * int            return        0: OK
+ *                              12000-12050: error code
  */
 int WebRtcAec_get_echo_status(void* handle, int* status);
 
@@ -189,8 +208,8 @@ int WebRtcAec_get_echo_status(void* handle, int* status);
  * -------------------------------------------------------------------
  * AecMetrics*    metrics       Struct which will be filled out with the
  *                              current echo metrics.
- * int            return         0: OK
- *                              -1: error
+ * int            return        0: OK
+ *                              12000-12050: error code
  */
 int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics);
 
@@ -208,27 +227,14 @@ int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics);
  * float*  fraction_poor_delays Fraction of the delay estimates that may
  *                              cause the AEC to perform poorly.
  *
- * int     return                0: OK
- *                              -1: error
+ * int            return        0: OK
+ *                              12000-12050: error code
  */
 int WebRtcAec_GetDelayMetrics(void* handle,
                               int* median,
                               int* std,
                               float* fraction_poor_delays);
 
-/*
- * Gets the last error code.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void*          aecInst       Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * int32_t        return        11000-11100: error code
- */
-int32_t WebRtcAec_get_error_code(void* aecInst);
-
 // Returns a pointer to the low level AEC handle.
 //
 // Input:
@@ -242,4 +248,4 @@ struct AecCore* WebRtcAec_aec_core(void* handle);
 #ifdef __cplusplus
 }
 #endif
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
index 95a6cf3324..e87219f33d 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
@@ -57,8 +57,6 @@ typedef struct {
 
   RingBuffer* far_pre_buf;  // Time domain far-end pre-buffer.
 
-  int lastError;
-
   int farend_started;
 
   AecCore* aec;
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc b/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
index 315ac3e9f9..42db082ff9 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
@@ -10,7 +10,7 @@
 
 // TODO(bjornv): Make this a comprehensive test.
 
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 
 #include <stdlib.h>
 #include <time.h>
diff --git a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index 07e3cf8add..567118d828 100644
--- a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -13,8 +13,7 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
 #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/typedefs.h"
 
 namespace {
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index f0d85d5328..6bf1cf7f3e 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -16,7 +16,7 @@
 
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
 #include "webrtc/system_wrappers/include/compile_assert_c.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
index df95e8bedf..3a8fafa4ec 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_c.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
@@ -16,7 +16,7 @@
 
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
 #include "webrtc/system_wrappers/include/compile_assert_c.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_mips.c b/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
index 3c2343a892..3ca9982ebf 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
@@ -12,7 +12,7 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
 
 static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
diff --git a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
index 83781e97fe..91e6f0e80c 100644
--- a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
+++ b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 
 #ifdef AEC_DEBUG
 #include <stdio.h>
@@ -68,8 +68,6 @@ typedef struct
     // Structures
     RingBuffer *farendBuf;
 
-    int lastError;
-
     AecmCore* aecmCore;
 } AecMobile;
 
@@ -100,7 +98,6 @@ void* WebRtcAecm_Create() {
     }
 
     aecm->initFlag = 0;
-    aecm->lastError = 0;
 
 #ifdef AEC_DEBUG
     aecm->aecmCore->farFile = fopen("aecFar.pcm","wb");
@@ -151,16 +148,14 @@ int32_t WebRtcAecm_Init(void *aecmInst, int32_t sampFreq)
 
     if (sampFreq != 8000 && sampFreq != 16000)
     {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
+        return AECM_BAD_PARAMETER_ERROR;
     }
     aecm->sampFreq = sampFreq;
 
     // Initialize AECM core
     if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1)
     {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
+        return AECM_UNSPECIFIED_ERROR;
     }
 
     // Initialize farend buffer
@@ -191,51 +186,53 @@ int32_t WebRtcAecm_Init(void *aecmInst, int32_t sampFreq)
 
     if (WebRtcAecm_set_config(aecm, aecConfig) == -1)
     {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
+        return AECM_UNSPECIFIED_ERROR;
     }
 
     return 0;
 }
 
-int32_t WebRtcAecm_BufferFarend(void *aecmInst, const int16_t *farend,
-                                size_t nrOfSamples)
-{
+// Returns any error that is caused when buffering the
+// farend signal.
+int32_t WebRtcAecm_GetBufferFarendError(void *aecmInst, const int16_t *farend,
+                                size_t nrOfSamples) {
   AecMobile* aecm = aecmInst;
-    int32_t retVal = 0;
 
-    if (aecm == NULL)
-    {
-        return -1;
-    }
+  if (aecm == NULL)
+    return -1;
 
-    if (farend == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
+  if (farend == NULL)
+    return AECM_NULL_POINTER_ERROR;
 
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
+  if (aecm->initFlag != kInitCheck)
+    return AECM_UNINITIALIZED_ERROR;
 
-    if (nrOfSamples != 80 && nrOfSamples != 160)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
+  if (nrOfSamples != 80 && nrOfSamples != 160)
+    return AECM_BAD_PARAMETER_ERROR;
 
-    // TODO: Is this really a good idea?
-    if (!aecm->ECstartup)
-    {
-        WebRtcAecm_DelayComp(aecm);
-    }
+  return 0;
+}
 
-    WebRtc_WriteBuffer(aecm->farendBuf, farend, nrOfSamples);
 
-    return retVal;
+int32_t WebRtcAecm_BufferFarend(void *aecmInst, const int16_t *farend,
+                                size_t nrOfSamples) {
+  AecMobile* aecm = aecmInst;
+
+  const int32_t err =
+      WebRtcAecm_GetBufferFarendError(aecmInst, farend, nrOfSamples);
+
+  if (err != 0)
+    return err;
+
+  // TODO(unknown): Is this really a good idea?
+  if (!aecm->ECstartup)
+  {
+    WebRtcAecm_DelayComp(aecm);
+  }
+
+  WebRtc_WriteBuffer(aecm->farendBuf, farend, nrOfSamples);
+
+  return 0;
 }
 
 int32_t WebRtcAecm_Process(void *aecmInst, const int16_t *nearendNoisy,
@@ -259,38 +256,32 @@ int32_t WebRtcAecm_Process(void *aecmInst, const int16_t *nearendNoisy,
 
     if (nearendNoisy == NULL)
     {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
+        return AECM_NULL_POINTER_ERROR;
     }
 
     if (out == NULL)
     {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
+        return AECM_NULL_POINTER_ERROR;
     }
 
     if (aecm->initFlag != kInitCheck)
     {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
+        return AECM_UNINITIALIZED_ERROR;
     }
 
     if (nrOfSamples != 80 && nrOfSamples != 160)
     {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
+        return AECM_BAD_PARAMETER_ERROR;
     }
 
     if (msInSndCardBuf < 0)
     {
         msInSndCardBuf = 0;
-        aecm->lastError = AECM_BAD_PARAMETER_WARNING;
-        retVal = -1;
+        retVal = AECM_BAD_PARAMETER_WARNING;
     } else if (msInSndCardBuf > 500)
     {
         msInSndCardBuf = 500;
-        aecm->lastError = AECM_BAD_PARAMETER_WARNING;
-        retVal = -1;
+        retVal = AECM_BAD_PARAMETER_WARNING;
     }
     msInSndCardBuf += 10;
     aecm->msInSndCardBuf = msInSndCardBuf;
@@ -453,21 +444,18 @@ int32_t WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
 
     if (aecm->initFlag != kInitCheck)
     {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
+        return AECM_UNINITIALIZED_ERROR;
     }
 
     if (config.cngMode != AecmFalse && config.cngMode != AecmTrue)
     {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
+        return AECM_BAD_PARAMETER_ERROR;
     }
     aecm->aecmCore->cngMode = config.cngMode;
 
     if (config.echoMode < 0 || config.echoMode > 4)
     {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
+        return AECM_BAD_PARAMETER_ERROR;
     }
     aecm->echoMode = config.echoMode;
 
@@ -524,33 +512,6 @@ int32_t WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
     return 0;
 }
 
-int32_t WebRtcAecm_get_config(void *aecmInst, AecmConfig *config)
-{
-  AecMobile* aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (config == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    config->cngMode = aecm->aecmCore->cngMode;
-    config->echoMode = aecm->echoMode;
-
-    return 0;
-}
-
 int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
                                 const void* echo_path,
                                 size_t size_bytes)
@@ -562,19 +523,16 @@ int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
       return -1;
     }
     if (echo_path == NULL) {
-      aecm->lastError = AECM_NULL_POINTER_ERROR;
-      return -1;
+      return AECM_NULL_POINTER_ERROR;
     }
     if (size_bytes != WebRtcAecm_echo_path_size_bytes())
     {
         // Input channel size does not match the size of AECM
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
+        return AECM_BAD_PARAMETER_ERROR;
     }
     if (aecm->initFlag != kInitCheck)
     {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
+        return AECM_UNINITIALIZED_ERROR;
     }
 
     WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
@@ -593,19 +551,16 @@ int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
       return -1;
     }
     if (echo_path == NULL) {
-      aecm->lastError = AECM_NULL_POINTER_ERROR;
-      return -1;
+      return AECM_NULL_POINTER_ERROR;
     }
     if (size_bytes != WebRtcAecm_echo_path_size_bytes())
     {
         // Input channel size does not match the size of AECM
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
+        return AECM_BAD_PARAMETER_ERROR;
     }
     if (aecm->initFlag != kInitCheck)
     {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
+        return AECM_UNINITIALIZED_ERROR;
     }
 
     memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
@@ -617,17 +572,6 @@ size_t WebRtcAecm_echo_path_size_bytes()
     return (PART_LEN1 * sizeof(int16_t));
 }
 
-int32_t WebRtcAecm_get_error_code(void *aecmInst)
-{
-  AecMobile* aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    return aecm->lastError;
-}
 
 static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf) {
     short delayNew, nSampSndCard;
diff --git a/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h b/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
index 7ae15c2a3d..b45ff59907 100644
--- a/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h
+++ b/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
 
 #include <stdlib.h>
 
@@ -66,7 +66,7 @@ void WebRtcAecm_Free(void* aecmInst);
  * Outputs                      Description
  * -------------------------------------------------------------------
  * int32_t        return        0: OK
- *                             -1: error
+ *                              1200-12004,12100: error/warning
  */
 int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq);
 
@@ -83,13 +83,32 @@ int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq);
  * Outputs                      Description
  * -------------------------------------------------------------------
  * int32_t        return        0: OK
- *                             -1: error
+ *                              1200-12004,12100: error/warning
  */
 int32_t WebRtcAecm_BufferFarend(void* aecmInst,
                                 const int16_t* farend,
                                 size_t nrOfSamples);
 
 /*
+ * Reports any errors that would arise when buffering a farend buffer.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*          aecmInst      Pointer to the AECM instance
+ * int16_t*       farend        In buffer containing one frame of
+ *                              farend signal
+ * int16_t        nrOfSamples   Number of samples in farend buffer
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t        return        0: OK
+ *                              1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_GetBufferFarendError(void* aecmInst,
+                                        const int16_t* farend,
+                                        size_t nrOfSamples);
+
+/*
  * Runs the AECM on an 80 or 160 sample blocks of data.
  *
  * Inputs                        Description
@@ -112,7 +131,7 @@ int32_t WebRtcAecm_BufferFarend(void* aecmInst,
  * -------------------------------------------------------------------
  * int16_t*       out            Out buffer, one frame of processed nearend
  * int32_t        return         0: OK
- *                              -1: error
+ *                               1200-12004,12100: error/warning
  */
 int32_t WebRtcAecm_Process(void* aecmInst,
                            const int16_t* nearendNoisy,
@@ -133,27 +152,11 @@ int32_t WebRtcAecm_Process(void* aecmInst,
  * Outputs                      Description
  * -------------------------------------------------------------------
  * int32_t        return        0: OK
- *                             -1: error
+ *                              1200-12004,12100: error/warning
  */
 int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config);
 
 /*
- * This function enables the user to set certain parameters on-the-fly
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void*          aecmInst      Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * AecmConfig*    config        Pointer to the config instance that
- *                              all properties will be written to
- * int32_t        return        0: OK
- *                             -1: error
- */
-int32_t WebRtcAecm_get_config(void *aecmInst, AecmConfig *config);
-
-/*
  * This function enables the user to set the echo path on-the-fly.
  *
  * Inputs                       Description
@@ -165,7 +168,7 @@ int32_t WebRtcAecm_get_config(void *aecmInst, AecmConfig *config);
  * Outputs                      Description
  * -------------------------------------------------------------------
  * int32_t      return          0: OK
- *                             -1: error
+ *                              1200-12004,12100: error/warning
  */
 int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
                                 const void* echo_path,
@@ -184,7 +187,7 @@ int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
  * Outputs                      Description
  * -------------------------------------------------------------------
  * int32_t      return          0: OK
- *                             -1: error
+ *                              1200-12004,12100: error/warning
  */
 int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
                                void* echo_path,
@@ -199,20 +202,8 @@ int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
  */
 size_t WebRtcAecm_echo_path_size_bytes();
 
-/*
- * Gets the last error code.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void*          aecmInst      Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * int32_t        return        11000-11100: error code
- */
-int32_t WebRtcAecm_get_error_code(void *aecmInst);
 
 #ifdef __cplusplus
 }
 #endif
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
diff --git a/webrtc/modules/audio_processing/agc/agc.cc b/webrtc/modules/audio_processing/agc/agc.cc
index 706b963aa1..fc78f07ebb 100644
--- a/webrtc/modules/audio_processing/agc/agc.cc
+++ b/webrtc/modules/audio_processing/agc/agc.cc
@@ -19,7 +19,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/agc/histogram.h"
 #include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
index 867022dcbf..e56984a1b1 100644
--- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
@@ -19,7 +19,7 @@
 
 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
@@ -168,19 +168,19 @@ int AgcManagerDirect::Initialize() {
   // example, what happens when we change devices.
 
   if (gctrl_->set_mode(GainControl::kFixedDigital) != 0) {
-    LOG_FERR1(LS_ERROR, set_mode, GainControl::kFixedDigital);
+    LOG(LS_ERROR) << "set_mode(GainControl::kFixedDigital) failed.";
     return -1;
   }
   if (gctrl_->set_target_level_dbfs(2) != 0) {
-    LOG_FERR1(LS_ERROR, set_target_level_dbfs, 2);
+    LOG(LS_ERROR) << "set_target_level_dbfs(2) failed.";
     return -1;
   }
   if (gctrl_->set_compression_gain_db(kDefaultCompressionGain) != 0) {
-    LOG_FERR1(LS_ERROR, set_compression_gain_db, kDefaultCompressionGain);
+    LOG(LS_ERROR) << "set_compression_gain_db(kDefaultCompressionGain) failed.";
     return -1;
   }
   if (gctrl_->enable_limiter(true) != 0) {
-    LOG_FERR1(LS_ERROR, enable_limiter, true);
+    LOG(LS_ERROR) << "enable_limiter(true) failed.";
     return -1;
   }
   return 0;
@@ -244,7 +244,7 @@ void AgcManagerDirect::Process(const int16_t* audio,
   }
 
   if (agc_->Process(audio, length, sample_rate_hz) != 0) {
-    LOG_FERR0(LS_ERROR, Agc::Process);
+    LOG(LS_ERROR) << "Agc::Process failed";
     assert(false);
   }
 
@@ -434,7 +434,8 @@ void AgcManagerDirect::UpdateCompressor() {
     compression_ = new_compression;
     compression_accumulator_ = new_compression;
     if (gctrl_->set_compression_gain_db(compression_) != 0) {
-      LOG_FERR1(LS_ERROR, set_compression_gain_db, compression_);
+      LOG(LS_ERROR) << "set_compression_gain_db(" << compression_
+                    << ") failed.";
     }
   }
 }
diff --git a/webrtc/modules/audio_processing/agc/agc_unittest.cc b/webrtc/modules/audio_processing/agc/agc_unittest.cc
index 66a8a2b1b3..25b99d8773 100644
--- a/webrtc/modules/audio_processing/agc/agc_unittest.cc
+++ b/webrtc/modules/audio_processing/agc/agc_unittest.cc
@@ -13,7 +13,7 @@
 #include "gmock/gmock.h"
 #include "gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/tools/agc/test_utils.h"
 
diff --git a/webrtc/modules/audio_processing/agc/histogram.cc b/webrtc/modules/audio_processing/agc/histogram.cc
index 1d3035fe12..5c66727a9f 100644
--- a/webrtc/modules/audio_processing/agc/histogram.cc
+++ b/webrtc/modules/audio_processing/agc/histogram.cc
@@ -13,7 +13,7 @@
 #include <cmath>
 #include <cstring>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
index be644d9701..3a1dc9d5ce 100644
--- a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
@@ -250,34 +250,35 @@ int WebRtcAgc_AddMic(void *state, int16_t* const* in_mic, size_t num_bands,
     return 0;
 }
 
-int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, size_t samples)
-{
+int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, size_t samples) {
+  LegacyAgc* stt = (LegacyAgc*)state;
+
+    int err =  WebRtcAgc_GetAddFarendError(state, samples);
+
+    if (err != 0)
+      return err;
+
+    return WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, in_far, samples);
+}
+
+int WebRtcAgc_GetAddFarendError(void *state, size_t samples) {
   LegacyAgc* stt;
   stt = (LegacyAgc*)state;
 
-    if (stt == NULL)
-    {
-        return -1;
-    }
+  if (stt == NULL)
+    return -1;
 
-    if (stt->fs == 8000)
-    {
-        if (samples != 80)
-        {
-            return -1;
-        }
-    } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000)
-    {
-        if (samples != 160)
-        {
-            return -1;
-        }
-    } else
-    {
-        return -1;
-    }
+  if (stt->fs == 8000) {
+    if (samples != 80)
+      return -1;
+  } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000) {
+    if (samples != 160)
+      return -1;
+  } else {
+    return -1;
+  }
 
-    return WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, in_far, samples);
+  return 0;
 }
 
 int WebRtcAgc_VirtualMic(void *agcInst, int16_t* const* in_near,
diff --git a/webrtc/modules/audio_processing/agc/legacy/gain_control.h b/webrtc/modules/audio_processing/agc/legacy/gain_control.h
index 08c1988f01..db942fe5ec 100644
--- a/webrtc/modules/audio_processing/agc/legacy/gain_control.h
+++ b/webrtc/modules/audio_processing/agc/legacy/gain_control.h
@@ -50,6 +50,20 @@ extern "C"
 #endif
 
 /*
+ * This function analyses the number of samples passed to
+ * farend and produces any error code that could arise.
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - samples           : Number of samples in input vector.
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error.
+ */
+int WebRtcAgc_GetAddFarendError(void* state, size_t samples);
+
+/*
  * This function processes a 10 ms frame of far-end speech to determine
  * if there is active speech. The length of the input speech vector must be
  * given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
diff --git a/webrtc/modules/audio_processing/agc/mock_agc.h b/webrtc/modules/audio_processing/agc/mock_agc.h
index 13dbd2edd5..e362200d86 100644
--- a/webrtc/modules/audio_processing/agc/mock_agc.h
+++ b/webrtc/modules/audio_processing/agc/mock_agc.h
@@ -14,7 +14,7 @@
 #include "webrtc/modules/audio_processing/agc/agc.h"
 
 #include "gmock/gmock.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index 81790a159b..ff64267e8c 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -26,7 +26,7 @@ const size_t kSamplesPer48kHzChannel = 480;
 int KeyboardChannelIndex(const StreamConfig& stream_config) {
   if (!stream_config.has_keyboard()) {
     assert(false);
-    return -1;
+    return 0;
   }
 
   return stream_config.num_channels();
@@ -44,9 +44,9 @@ size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
 }  // namespace
 
 AudioBuffer::AudioBuffer(size_t input_num_frames,
-                         int num_input_channels,
+                         size_t num_input_channels,
                          size_t process_num_frames,
-                         int num_process_channels,
+                         size_t num_process_channels,
                          size_t output_num_frames)
   : input_num_frames_(input_num_frames),
     num_input_channels_(num_input_channels),
@@ -74,7 +74,7 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
                                                    num_proc_channels_));
 
     if (input_num_frames_ != proc_num_frames_) {
-      for (int i = 0; i < num_proc_channels_; ++i) {
+      for (size_t i = 0; i < num_proc_channels_; ++i) {
         input_resamplers_.push_back(
             new PushSincResampler(input_num_frames_,
                                   proc_num_frames_));
@@ -82,7 +82,7 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
     }
 
     if (output_num_frames_ != proc_num_frames_) {
-      for (int i = 0; i < num_proc_channels_; ++i) {
+      for (size_t i = 0; i < num_proc_channels_; ++i) {
         output_resamplers_.push_back(
             new PushSincResampler(proc_num_frames_,
                                   output_num_frames_));
@@ -130,7 +130,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
 
   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_proc_channels_; ++i) {
+    for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(data_ptr[i],
                                      input_num_frames_,
                                      process_buffer_->channels()[i],
@@ -140,7 +140,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
   }
 
   // Convert to the S16 range.
-  for (int i = 0; i < num_proc_channels_; ++i) {
+  for (size_t i = 0; i < num_proc_channels_; ++i) {
     FloatToFloatS16(data_ptr[i],
                     proc_num_frames_,
                     data_->fbuf()->channels()[i]);
@@ -150,7 +150,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
 void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                          float* const* data) {
   assert(stream_config.num_frames() == output_num_frames_);
-  assert(stream_config.num_channels() == num_channels_);
+  assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1);
 
   // Convert to the float range.
   float* const* data_ptr = data;
@@ -158,7 +158,7 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
     // Convert to an intermediate buffer for subsequent resampling.
     data_ptr = process_buffer_->channels();
   }
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     FloatS16ToFloat(data_->fbuf()->channels()[i],
                     proc_num_frames_,
                     data_ptr[i]);
@@ -166,13 +166,18 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
 
   // Resample.
   if (output_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_channels_; ++i) {
+    for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(data_ptr[i],
                                       proc_num_frames_,
                                       data[i],
                                       output_num_frames_);
     }
   }
+
+  // Upmix.
+  for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
+    memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
+  }
 }
 
 void AudioBuffer::InitForNewData() {
@@ -192,13 +197,13 @@ int16_t* const* AudioBuffer::channels() {
   return data_->ibuf()->channels();
 }
 
-const int16_t* const* AudioBuffer::split_bands_const(int channel) const {
+const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
   return split_data_.get() ?
          split_data_->ibuf_const()->bands(channel) :
          data_->ibuf_const()->bands(channel);
 }
 
-int16_t* const* AudioBuffer::split_bands(int channel) {
+int16_t* const* AudioBuffer::split_bands(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->ibuf()->bands(channel) :
@@ -249,13 +254,13 @@ float* const* AudioBuffer::channels_f() {
   return data_->fbuf()->channels();
 }
 
-const float* const* AudioBuffer::split_bands_const_f(int channel) const {
+const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
   return split_data_.get() ?
          split_data_->fbuf_const()->bands(channel) :
          data_->fbuf_const()->bands(channel);
 }
 
-float* const* AudioBuffer::split_bands_f(int channel) {
+float* const* AudioBuffer::split_bands_f(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->fbuf()->bands(channel) :
@@ -336,11 +341,11 @@ AudioFrame::VADActivity AudioBuffer::activity() const {
   return activity_;
 }
 
-int AudioBuffer::num_channels() const {
+size_t AudioBuffer::num_channels() const {
   return num_channels_;
 }
 
-void AudioBuffer::set_num_channels(int num_channels) {
+void AudioBuffer::set_num_channels(size_t num_channels) {
   num_channels_ = num_channels;
 }
 
@@ -393,7 +398,7 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
 
   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_proc_channels_; ++i) {
+    for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
                                      input_num_frames_,
                                      data_->fbuf()->channels()[i],
@@ -418,7 +423,7 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
       output_buffer_.reset(
           new IFChannelBuffer(output_num_frames_, num_channels_));
     }
-    for (int i = 0; i < num_channels_; ++i) {
+    for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(
           data_->fbuf()->channels()[i], proc_num_frames_,
           output_buffer_->fbuf()->channels()[i], output_num_frames_);
@@ -443,7 +448,7 @@ void AudioBuffer::CopyLowPassToReference() {
         new ChannelBuffer<int16_t>(num_split_frames_,
                                    num_proc_channels_));
   }
-  for (int i = 0; i < num_proc_channels_; i++) {
+  for (size_t i = 0; i < num_proc_channels_; i++) {
     memcpy(low_pass_reference_channels_->channels()[i],
            split_bands_const(i)[kBand0To8kHz],
            low_pass_reference_channels_->num_frames_per_band() *
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 864633f267..ff12ca2d95 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -15,7 +15,7 @@
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/splitting_filter.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/scoped_vector.h"
 #include "webrtc/typedefs.h"
 
@@ -34,14 +34,14 @@ class AudioBuffer {
  public:
   // TODO(ajm): Switch to take ChannelLayouts.
   AudioBuffer(size_t input_num_frames,
-              int num_input_channels,
+              size_t num_input_channels,
               size_t process_num_frames,
-              int num_process_channels,
+              size_t num_process_channels,
               size_t output_num_frames);
   virtual ~AudioBuffer();
 
-  int num_channels() const;
-  void set_num_channels(int num_channels);
+  size_t num_channels() const;
+  void set_num_channels(size_t num_channels);
   size_t num_frames() const;
   size_t num_frames_per_band() const;
   size_t num_keyboard_frames() const;
@@ -65,10 +65,10 @@ class AudioBuffer {
   // 0 <= channel < |num_proc_channels_|
   // 0 <= band < |num_bands_|
   // 0 <= sample < |num_split_frames_|
-  int16_t* const* split_bands(int channel);
-  const int16_t* const* split_bands_const(int channel) const;
-  float* const* split_bands_f(int channel);
-  const float* const* split_bands_const_f(int channel) const;
+  int16_t* const* split_bands(size_t channel);
+  const int16_t* const* split_bands_const(size_t channel) const;
+  float* const* split_bands_f(size_t channel);
+  const float* const* split_bands_const_f(size_t channel) const;
 
   // Returns a pointer array to the channels for a specific band.
   // Usage:
@@ -128,16 +128,16 @@ class AudioBuffer {
   // The audio is passed into DeinterleaveFrom() or CopyFrom() with input
   // format (samples per channel and number of channels).
   const size_t input_num_frames_;
-  const int num_input_channels_;
+  const size_t num_input_channels_;
   // The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
   // format.
   const size_t proc_num_frames_;
-  const int num_proc_channels_;
+  const size_t num_proc_channels_;
   // The audio is returned by InterleaveTo() and CopyTo() with output samples
   // per channels and the current number of channels. This last one can be
   // changed at any time using set_num_channels().
   const size_t output_num_frames_;
-  int num_channels_;
+  size_t num_channels_;
 
   size_t num_bands_;
   size_t num_split_frames_;
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 8f1fbdf0be..7ddd4f5a15 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -41,11 +41,11 @@
         'aec/aec_resampler.h',
         'aec/echo_cancellation.c',
         'aec/echo_cancellation_internal.h',
-        'aec/include/echo_cancellation.h',
+        'aec/echo_cancellation.h',
         'aecm/aecm_core.c',
         'aecm/aecm_core.h',
         'aecm/echo_control_mobile.c',
-        'aecm/include/echo_control_mobile.h',
+        'aecm/echo_control_mobile.h',
         'agc/agc.cc',
         'agc/agc.h',
         'agc/agc_manager_direct.cc',
@@ -162,7 +162,7 @@
         ['prefer_fixed_point==1', {
           'defines': ['WEBRTC_NS_FIXED'],
           'sources': [
-            'ns/include/noise_suppression_x.h',
+            'ns/noise_suppression_x.h',
             'ns/noise_suppression_x.c',
             'ns/nsx_core.c',
             'ns/nsx_core.h',
@@ -183,7 +183,7 @@
           'defines': ['WEBRTC_NS_FLOAT'],
           'sources': [
             'ns/defines.h',
-            'ns/include/noise_suppression.h',
+            'ns/noise_suppression.h',
             'ns/noise_suppression.c',
             'ns/ns_core.c',
             'ns/ns_core.h',
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index c6574151d0..744309c774 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -15,6 +15,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_file.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/include/audio_util.h"
@@ -36,8 +37,7 @@ extern "C" {
 #include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
@@ -75,7 +75,6 @@ static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) {
   assert(false);
   return false;
 }
-
 }  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
@@ -147,6 +146,35 @@ class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
   int volume_;
 };
 
+struct AudioProcessingImpl::ApmPublicSubmodules {
+  ApmPublicSubmodules()
+      : echo_cancellation(nullptr),
+        echo_control_mobile(nullptr),
+        gain_control(nullptr) {}
+  // Accessed externally of APM without any lock acquired.
+  EchoCancellationImpl* echo_cancellation;
+  EchoControlMobileImpl* echo_control_mobile;
+  GainControlImpl* gain_control;
+  rtc::scoped_ptr<HighPassFilterImpl> high_pass_filter;
+  rtc::scoped_ptr<LevelEstimatorImpl> level_estimator;
+  rtc::scoped_ptr<NoiseSuppressionImpl> noise_suppression;
+  rtc::scoped_ptr<VoiceDetectionImpl> voice_detection;
+  rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
+
+  // Accessed internally from both render and capture.
+  rtc::scoped_ptr<TransientSuppressor> transient_suppressor;
+  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
+};
+
+struct AudioProcessingImpl::ApmPrivateSubmodules {
+  explicit ApmPrivateSubmodules(Beamformer<float>* beamformer)
+      : beamformer(beamformer) {}
+  // Accessed internally from capture or during initialization
+  std::list<ProcessingComponent*> component_list;
+  rtc::scoped_ptr<Beamformer<float>> beamformer;
+  rtc::scoped_ptr<AgcManagerDirect> agc_manager;
+};
+
 const int AudioProcessing::kNativeSampleRatesHz[] = {
     AudioProcessing::kSampleRate8kHz,
     AudioProcessing::kSampleRate16kHz,
@@ -172,7 +200,7 @@ AudioProcessing* AudioProcessing::Create(const Config& config,
   AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
   if (apm->Initialize() != kNoError) {
     delete apm;
-    apm = NULL;
+    apm = nullptr;
   }
 
   return apm;
@@ -183,102 +211,82 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config)
 
 AudioProcessingImpl::AudioProcessingImpl(const Config& config,
                                          Beamformer<float>* beamformer)
-    : echo_cancellation_(NULL),
-      echo_control_mobile_(NULL),
-      gain_control_(NULL),
-      high_pass_filter_(NULL),
-      level_estimator_(NULL),
-      noise_suppression_(NULL),
-      voice_detection_(NULL),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-      debug_file_(FileWrapper::Create()),
-      event_msg_(new audioproc::Event()),
-#endif
-      api_format_({{{kSampleRate16kHz, 1, false},
-                    {kSampleRate16kHz, 1, false},
-                    {kSampleRate16kHz, 1, false},
-                    {kSampleRate16kHz, 1, false}}}),
-      fwd_proc_format_(kSampleRate16kHz),
-      rev_proc_format_(kSampleRate16kHz, 1),
-      split_rate_(kSampleRate16kHz),
-      stream_delay_ms_(0),
-      delay_offset_ms_(0),
-      was_stream_delay_set_(false),
-      last_stream_delay_ms_(0),
-      last_aec_system_delay_ms_(0),
-      stream_delay_jumps_(-1),
-      aec_system_delay_jumps_(-1),
-      output_will_be_muted_(false),
-      key_pressed_(false),
+    : public_submodules_(new ApmPublicSubmodules()),
+      private_submodules_(new ApmPrivateSubmodules(beamformer)),
+      constants_(config.Get<ExperimentalAgc>().startup_min_volume,
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-      use_new_agc_(false),
+                 false,
 #else
-      use_new_agc_(config.Get<ExperimentalAgc>().enabled),
+                 config.Get<ExperimentalAgc>().enabled,
 #endif
-      agc_startup_min_volume_(config.Get<ExperimentalAgc>().startup_min_volume),
+                 config.Get<Intelligibility>().enabled),
+
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-      transient_suppressor_enabled_(false),
+      capture_(false,
 #else
-      transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
+      capture_(config.Get<ExperimentalNs>().enabled,
 #endif
-      beamformer_enabled_(config.Get<Beamforming>().enabled),
-      beamformer_(beamformer),
-      array_geometry_(config.Get<Beamforming>().array_geometry),
-      target_direction_(config.Get<Beamforming>().target_direction),
-      intelligibility_enabled_(config.Get<Intelligibility>().enabled) {
-  echo_cancellation_ = new EchoCancellationImpl(this, crit_);
-  component_list_.push_back(echo_cancellation_);
-
-  echo_control_mobile_ = new EchoControlMobileImpl(this, crit_);
-  component_list_.push_back(echo_control_mobile_);
-
-  gain_control_ = new GainControlImpl(this, crit_);
-  component_list_.push_back(gain_control_);
-
-  high_pass_filter_ = new HighPassFilterImpl(this, crit_);
-  component_list_.push_back(high_pass_filter_);
-
-  level_estimator_ = new LevelEstimatorImpl(this, crit_);
-  component_list_.push_back(level_estimator_);
-
-  noise_suppression_ = new NoiseSuppressionImpl(this, crit_);
-  component_list_.push_back(noise_suppression_);
-
-  voice_detection_ = new VoiceDetectionImpl(this, crit_);
-  component_list_.push_back(voice_detection_);
-
-  gain_control_for_new_agc_.reset(new GainControlForNewAgc(gain_control_));
+               config.Get<Beamforming>().array_geometry,
+               config.Get<Beamforming>().target_direction),
+      capture_nonlocked_(config.Get<Beamforming>().enabled)
+{
+  {
+    rtc::CritScope cs_render(&crit_render_);
+    rtc::CritScope cs_capture(&crit_capture_);
+
+    public_submodules_->echo_cancellation =
+        new EchoCancellationImpl(this, &crit_render_, &crit_capture_);
+    public_submodules_->echo_control_mobile =
+        new EchoControlMobileImpl(this, &crit_render_, &crit_capture_);
+    public_submodules_->gain_control =
+        new GainControlImpl(this, &crit_capture_, &crit_capture_);
+    public_submodules_->high_pass_filter.reset(
+        new HighPassFilterImpl(&crit_capture_));
+    public_submodules_->level_estimator.reset(
+        new LevelEstimatorImpl(&crit_capture_));
+    public_submodules_->noise_suppression.reset(
+        new NoiseSuppressionImpl(&crit_capture_));
+    public_submodules_->voice_detection.reset(
+        new VoiceDetectionImpl(&crit_capture_));
+    public_submodules_->gain_control_for_new_agc.reset(
+        new GainControlForNewAgc(public_submodules_->gain_control));
+
+    private_submodules_->component_list.push_back(
+        public_submodules_->echo_cancellation);
+    private_submodules_->component_list.push_back(
+        public_submodules_->echo_control_mobile);
+    private_submodules_->component_list.push_back(
+        public_submodules_->gain_control);
+  }
 
   SetExtraOptions(config);
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
-  {
-    CriticalSectionScoped crit_scoped(crit_);
-    // Depends on gain_control_ and gain_control_for_new_agc_.
-    agc_manager_.reset();
-    // Depends on gain_control_.
-    gain_control_for_new_agc_.reset();
-    while (!component_list_.empty()) {
-      ProcessingComponent* component = component_list_.front();
-      component->Destroy();
-      delete component;
-      component_list_.pop_front();
-    }
+  // Depends on gain_control_ and
+  // public_submodules_->gain_control_for_new_agc.
+  private_submodules_->agc_manager.reset();
+  // Depends on gain_control_.
+  public_submodules_->gain_control_for_new_agc.reset();
+  while (!private_submodules_->component_list.empty()) {
+    ProcessingComponent* component =
+        private_submodules_->component_list.front();
+    component->Destroy();
+    delete component;
+    private_submodules_->component_list.pop_front();
+  }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-    if (debug_file_->Open()) {
-      debug_file_->CloseFile();
-    }
-#endif
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.debug_file->CloseFile();
   }
-  delete crit_;
-  crit_ = NULL;
+#endif
 }
 
 int AudioProcessingImpl::Initialize() {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner during initialization.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   return InitializeLocked();
 }
 
@@ -306,44 +314,73 @@ int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
 }
 
 int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner during initialization.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  return InitializeLocked(processing_config);
+}
+
+int AudioProcessingImpl::MaybeInitializeRender(
+    const ProcessingConfig& processing_config) {
+  return MaybeInitialize(processing_config);
+}
+
+int AudioProcessingImpl::MaybeInitializeCapture(
+    const ProcessingConfig& processing_config) {
+  return MaybeInitialize(processing_config);
+}
+
+// Calls InitializeLocked() if any of the audio parameters have changed from
+// their current values (needs to be called while holding the crit_render_lock).
+int AudioProcessingImpl::MaybeInitialize(
+    const ProcessingConfig& processing_config) {
+  // Called from both threads. Thread check is therefore not possible.
+  if (processing_config == formats_.api_format) {
+    return kNoError;
+  }
+
+  rtc::CritScope cs_capture(&crit_capture_);
   return InitializeLocked(processing_config);
 }
 
 int AudioProcessingImpl::InitializeLocked() {
   const int fwd_audio_buffer_channels =
-      beamformer_enabled_ ? api_format_.input_stream().num_channels()
-                          : api_format_.output_stream().num_channels();
+      capture_nonlocked_.beamformer_enabled
+          ? formats_.api_format.input_stream().num_channels()
+          : formats_.api_format.output_stream().num_channels();
   const int rev_audio_buffer_out_num_frames =
-      api_format_.reverse_output_stream().num_frames() == 0
-          ? rev_proc_format_.num_frames()
-          : api_format_.reverse_output_stream().num_frames();
-  if (api_format_.reverse_input_stream().num_channels() > 0) {
-    render_audio_.reset(new AudioBuffer(
-        api_format_.reverse_input_stream().num_frames(),
-        api_format_.reverse_input_stream().num_channels(),
-        rev_proc_format_.num_frames(), rev_proc_format_.num_channels(),
+      formats_.api_format.reverse_output_stream().num_frames() == 0
+          ? formats_.rev_proc_format.num_frames()
+          : formats_.api_format.reverse_output_stream().num_frames();
+  if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
+    render_.render_audio.reset(new AudioBuffer(
+        formats_.api_format.reverse_input_stream().num_frames(),
+        formats_.api_format.reverse_input_stream().num_channels(),
+        formats_.rev_proc_format.num_frames(),
+        formats_.rev_proc_format.num_channels(),
         rev_audio_buffer_out_num_frames));
     if (rev_conversion_needed()) {
-      render_converter_ = AudioConverter::Create(
-          api_format_.reverse_input_stream().num_channels(),
-          api_format_.reverse_input_stream().num_frames(),
-          api_format_.reverse_output_stream().num_channels(),
-          api_format_.reverse_output_stream().num_frames());
+      render_.render_converter = AudioConverter::Create(
+          formats_.api_format.reverse_input_stream().num_channels(),
+          formats_.api_format.reverse_input_stream().num_frames(),
+          formats_.api_format.reverse_output_stream().num_channels(),
+          formats_.api_format.reverse_output_stream().num_frames());
     } else {
-      render_converter_.reset(nullptr);
+      render_.render_converter.reset(nullptr);
     }
   } else {
-    render_audio_.reset(nullptr);
-    render_converter_.reset(nullptr);
+    render_.render_audio.reset(nullptr);
+    render_.render_converter.reset(nullptr);
   }
-  capture_audio_.reset(new AudioBuffer(
-      api_format_.input_stream().num_frames(),
-      api_format_.input_stream().num_channels(), fwd_proc_format_.num_frames(),
-      fwd_audio_buffer_channels, api_format_.output_stream().num_frames()));
+  capture_.capture_audio.reset(
+      new AudioBuffer(formats_.api_format.input_stream().num_frames(),
+                      formats_.api_format.input_stream().num_channels(),
+                      capture_nonlocked_.fwd_proc_format.num_frames(),
+                      fwd_audio_buffer_channels,
+                      formats_.api_format.output_stream().num_frames()));
 
   // Initialize all components.
-  for (auto item : component_list_) {
+  for (auto item : private_submodules_->component_list) {
     int err = item->Initialize();
     if (err != kNoError) {
       return err;
@@ -351,15 +388,16 @@ int AudioProcessingImpl::InitializeLocked() {
   }
 
   InitializeExperimentalAgc();
-
   InitializeTransient();
-
   InitializeBeamformer();
-
   InitializeIntelligibility();
+  InitializeHighPassFilter();
+  InitializeNoiseSuppression();
+  InitializeLevelEstimator();
+  InitializeVoiceDetection();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
+  if (debug_dump_.debug_file->Open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
       return err;
@@ -372,16 +410,13 @@ int AudioProcessingImpl::InitializeLocked() {
 
 int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
   for (const auto& stream : config.streams) {
-    if (stream.num_channels() < 0) {
-      return kBadNumberChannelsError;
-    }
     if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
       return kBadSampleRateError;
     }
   }
 
-  const int num_in_channels = config.input_stream().num_channels();
-  const int num_out_channels = config.output_stream().num_channels();
+  const size_t num_in_channels = config.input_stream().num_channels();
+  const size_t num_out_channels = config.output_stream().num_channels();
 
   // Need at least one input channel.
   // Need either one output channel or as many outputs as there are inputs.
@@ -390,18 +425,17 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
     return kBadNumberChannelsError;
   }
 
-  if (beamformer_enabled_ &&
-      (static_cast<size_t>(num_in_channels) != array_geometry_.size() ||
-       num_out_channels > 1)) {
+  if (capture_nonlocked_.beamformer_enabled &&
+      num_in_channels != capture_.array_geometry.size()) {
     return kBadNumberChannelsError;
   }
 
-  api_format_ = config;
+  formats_.api_format = config;
 
   // We process at the closest native rate >= min(input rate, output rate)...
   const int min_proc_rate =
-      std::min(api_format_.input_stream().sample_rate_hz(),
-               api_format_.output_stream().sample_rate_hz());
+      std::min(formats_.api_format.input_stream().sample_rate_hz(),
+               formats_.api_format.output_stream().sample_rate_hz());
   int fwd_proc_rate;
   for (size_t i = 0; i < kNumNativeSampleRates; ++i) {
     fwd_proc_rate = kNativeSampleRatesHz[i];
@@ -410,20 +444,20 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
     }
   }
   // ...with one exception.
-  if (echo_control_mobile_->is_enabled() &&
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
       min_proc_rate > kMaxAECMSampleRateHz) {
     fwd_proc_rate = kMaxAECMSampleRateHz;
   }
 
-  fwd_proc_format_ = StreamConfig(fwd_proc_rate);
+  capture_nonlocked_.fwd_proc_format = StreamConfig(fwd_proc_rate);
 
   // We normally process the reverse stream at 16 kHz. Unless...
   int rev_proc_rate = kSampleRate16kHz;
-  if (fwd_proc_format_.sample_rate_hz() == kSampleRate8kHz) {
+  if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
     // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    if (api_format_.reverse_input_stream().sample_rate_hz() ==
+    if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
         kSampleRate32kHz) {
       // ...or the input is at 32 kHz, in which case we use the splitting
       // filter rather than the resampler.
@@ -433,66 +467,89 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
 
   // Always downmix the reverse stream to mono for analysis. This has been
   // demonstrated to work well for AEC in most practical scenarios.
-  rev_proc_format_ = StreamConfig(rev_proc_rate, 1);
+  formats_.rev_proc_format = StreamConfig(rev_proc_rate, 1);
 
-  if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
-      fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
-    split_rate_ = kSampleRate16kHz;
+  if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate32kHz ||
+      capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate48kHz) {
+    capture_nonlocked_.split_rate = kSampleRate16kHz;
   } else {
-    split_rate_ = fwd_proc_format_.sample_rate_hz();
+    capture_nonlocked_.split_rate =
+        capture_nonlocked_.fwd_proc_format.sample_rate_hz();
   }
 
   return InitializeLocked();
 }
 
-// Calls InitializeLocked() if any of the audio parameters have changed from
-// their current values.
-int AudioProcessingImpl::MaybeInitializeLocked(
-    const ProcessingConfig& processing_config) {
-  if (processing_config == api_format_) {
-    return kNoError;
-  }
-  return InitializeLocked(processing_config);
-}
-
 void AudioProcessingImpl::SetExtraOptions(const Config& config) {
-  CriticalSectionScoped crit_scoped(crit_);
-  for (auto item : component_list_) {
+  // Run in a single-threaded manner when setting the extra options.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  for (auto item : private_submodules_->component_list) {
     item->SetExtraOptions(config);
   }
 
-  if (transient_suppressor_enabled_ != config.Get<ExperimentalNs>().enabled) {
-    transient_suppressor_enabled_ = config.Get<ExperimentalNs>().enabled;
+  if (capture_.transient_suppressor_enabled !=
+      config.Get<ExperimentalNs>().enabled) {
+    capture_.transient_suppressor_enabled =
+        config.Get<ExperimentalNs>().enabled;
     InitializeTransient();
   }
+
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+  if (capture_nonlocked_.beamformer_enabled !=
+          config.Get<Beamforming>().enabled) {
+    capture_nonlocked_.beamformer_enabled = config.Get<Beamforming>().enabled;
+    if (config.Get<Beamforming>().array_geometry.size() > 1) {
+      capture_.array_geometry = config.Get<Beamforming>().array_geometry;
+    }
+    capture_.target_direction = config.Get<Beamforming>().target_direction;
+    InitializeBeamformer();
+  }
+#endif  // WEBRTC_ANDROID_PLATFORM_BUILD
 }
 
+int AudioProcessingImpl::input_sample_rate_hz() const {
+  // Accessed from outside APM, hence a lock is needed.
+  rtc::CritScope cs(&crit_capture_);
+  return formats_.api_format.input_stream().sample_rate_hz();
+}
 
 int AudioProcessingImpl::proc_sample_rate_hz() const {
-  return fwd_proc_format_.sample_rate_hz();
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
-  return split_rate_;
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.split_rate;
+}
+
+size_t AudioProcessingImpl::num_reverse_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.rev_proc_format.num_channels();
 }
 
-int AudioProcessingImpl::num_reverse_channels() const {
-  return rev_proc_format_.num_channels();
+size_t AudioProcessingImpl::num_input_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.api_format.input_stream().num_channels();
 }
 
-int AudioProcessingImpl::num_input_channels() const {
-  return api_format_.input_stream().num_channels();
+size_t AudioProcessingImpl::num_proc_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.beamformer_enabled ? 1 : num_output_channels();
 }
 
-int AudioProcessingImpl::num_output_channels() const {
-  return api_format_.output_stream().num_channels();
+size_t AudioProcessingImpl::num_output_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.api_format.output_stream().num_channels();
 }
 
 void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
-  CriticalSectionScoped lock(crit_);
-  output_will_be_muted_ = muted;
-  if (agc_manager_.get()) {
-    agc_manager_->SetCaptureMuted(output_will_be_muted_);
+  rtc::CritScope cs(&crit_capture_);
+  capture_.output_will_be_muted = muted;
+  if (private_submodules_->agc_manager.get()) {
+    private_submodules_->agc_manager->SetCaptureMuted(
+        capture_.output_will_be_muted);
   }
 }
 
@@ -504,13 +561,21 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
                                        int output_sample_rate_hz,
                                        ChannelLayout output_layout,
                                        float* const* dest) {
-  CriticalSectionScoped crit_scoped(crit_);
-  StreamConfig input_stream = api_format_.input_stream();
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_ChannelLayout");
+  StreamConfig input_stream;
+  StreamConfig output_stream;
+  {
+    // Access the formats_.api_format.input_stream beneath the capture lock.
+    // The lock must be released as it is later required in the call
+    // to ProcessStream(,,,);
+    rtc::CritScope cs(&crit_capture_);
+    input_stream = formats_.api_format.input_stream();
+    output_stream = formats_.api_format.output_stream();
+  }
+
   input_stream.set_sample_rate_hz(input_sample_rate_hz);
   input_stream.set_num_channels(ChannelsFromLayout(input_layout));
   input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
-
-  StreamConfig output_stream = api_format_.output_stream();
   output_stream.set_sample_rate_hz(output_sample_rate_hz);
   output_stream.set_num_channels(ChannelsFromLayout(output_layout));
   output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout));
@@ -525,44 +590,64 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
                                        const StreamConfig& input_config,
                                        const StreamConfig& output_config,
                                        float* const* dest) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (!src || !dest) {
-    return kNullPointerError;
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
+  ProcessingConfig processing_config;
+  {
+    // Acquire the capture lock in order to safely call the function
+    // that retrieves the render side data. This function accesses apm
+    // getters that need the capture lock held when being called.
+    rtc::CritScope cs_capture(&crit_capture_);
+    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    public_submodules_->echo_control_mobile->ReadQueuedRenderData();
+    public_submodules_->gain_control->ReadQueuedRenderData();
+
+    if (!src || !dest) {
+      return kNullPointerError;
+    }
+
+    processing_config = formats_.api_format;
   }
 
-  ProcessingConfig processing_config = api_format_;
   processing_config.input_stream() = input_config;
   processing_config.output_stream() = output_config;
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  {
+    // Do conditional reinitialization.
+    rtc::CritScope cs_render(&crit_render_);
+    RETURN_ON_ERR(MaybeInitializeCapture(processing_config));
+  }
+  rtc::CritScope cs_capture(&crit_capture_);
   assert(processing_config.input_stream().num_frames() ==
-         api_format_.input_stream().num_frames());
+         formats_.api_format.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
+  if (debug_dump_.debug_file->Open()) {
     RETURN_ON_ERR(WriteConfigMessage(false));
 
-    event_msg_->set_type(audioproc::Event::STREAM);
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+    debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
-        sizeof(float) * api_format_.input_stream().num_frames();
-    for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
+        sizeof(float) * formats_.api_format.input_stream().num_frames();
+    for (size_t i = 0; i < formats_.api_format.input_stream().num_channels();
+         ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
-  capture_audio_->CopyFrom(src, api_format_.input_stream());
+  capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->CopyTo(api_format_.output_stream(), dest);
+  capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
-        sizeof(float) * api_format_.output_stream().num_frames();
-    for (int i = 0; i < api_format_.output_stream().num_channels(); ++i)
+        sizeof(float) * formats_.api_format.output_stream().num_frames();
+    for (size_t i = 0; i < formats_.api_format.output_stream().num_channels();
+         ++i)
       msg->add_output_channel(dest[i], channel_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
@@ -570,7 +655,20 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
 }
 
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(crit_);
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
+  {
+    // Acquire the capture lock in order to safely call the function
+    // that retrieves the render side data. This function accesses apm
+    // getters that need the capture lock held when being called.
+    // The lock needs to be released as
+    // public_submodules_->echo_control_mobile->is_enabled() aquires this lock
+    // as well.
+    rtc::CritScope cs_capture(&crit_capture_);
+    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    public_submodules_->echo_control_mobile->ReadQueuedRenderData();
+    public_submodules_->gain_control->ReadQueuedRenderData();
+  }
+
   if (!frame) {
     return kNullPointerError;
   }
@@ -581,46 +679,62 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
       frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
-  if (echo_control_mobile_->is_enabled() &&
+
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
       frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
     LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
     return kUnsupportedComponentError;
   }
 
-  // TODO(ajm): The input and output rates and channels are currently
-  // constrained to be identical in the int16 interface.
-  ProcessingConfig processing_config = api_format_;
+  ProcessingConfig processing_config;
+  {
+    // Aquire lock for the access of api_format.
+    // The lock is released immediately due to the conditional
+    // reinitialization.
+    rtc::CritScope cs_capture(&crit_capture_);
+    // TODO(ajm): The input and output rates and channels are currently
+    // constrained to be identical in the int16 interface.
+    processing_config = formats_.api_format;
+  }
   processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
   processing_config.input_stream().set_num_channels(frame->num_channels_);
   processing_config.output_stream().set_sample_rate_hz(frame->sample_rate_hz_);
   processing_config.output_stream().set_num_channels(frame->num_channels_);
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
-  if (frame->samples_per_channel_ != api_format_.input_stream().num_frames()) {
+  {
+    // Do conditional reinitialization.
+    rtc::CritScope cs_render(&crit_render_);
+    RETURN_ON_ERR(MaybeInitializeCapture(processing_config));
+  }
+  rtc::CritScope cs_capture(&crit_capture_);
+  if (frame->samples_per_channel_ !=
+      formats_.api_format.input_stream().num_frames()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::STREAM);
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_input_data(frame->data_, data_size);
   }
 #endif
 
-  capture_audio_->DeinterleaveFrom(frame);
+  capture_.capture_audio->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->InterleaveTo(frame, output_copy_needed(is_data_processed()));
+  capture_.capture_audio->InterleaveTo(frame,
+                                       output_copy_needed(is_data_processed()));
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_output_data(frame->data_, data_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
@@ -629,22 +743,25 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
 
 int AudioProcessingImpl::ProcessStreamLocked() {
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
-    msg->set_delay(stream_delay_ms_);
-    msg->set_drift(echo_cancellation_->stream_drift_samples());
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
+    msg->set_delay(capture_nonlocked_.stream_delay_ms);
+    msg->set_drift(
+        public_submodules_->echo_cancellation->stream_drift_samples());
     msg->set_level(gain_control()->stream_analog_level());
-    msg->set_keypress(key_pressed_);
+    msg->set_keypress(capture_.key_pressed);
   }
 #endif
 
   MaybeUpdateHistograms();
 
-  AudioBuffer* ca = capture_audio_.get();  // For brevity.
+  AudioBuffer* ca = capture_.capture_audio.get();  // For brevity.
 
-  if (use_new_agc_ && gain_control_->is_enabled()) {
-    agc_manager_->AnalyzePreProcess(ca->channels()[0], ca->num_channels(),
-                                    fwd_proc_format_.num_frames());
+  if (constants_.use_new_agc &&
+      public_submodules_->gain_control->is_enabled()) {
+    private_submodules_->agc_manager->AnalyzePreProcess(
+        ca->channels()[0], ca->num_channels(),
+        capture_nonlocked_.fwd_proc_format.num_frames());
   }
 
   bool data_processed = is_data_processed();
@@ -652,34 +769,41 @@ int AudioProcessingImpl::ProcessStreamLocked() {
     ca->SplitIntoFrequencyBands();
   }
 
-  if (intelligibility_enabled_) {
-    intelligibility_enhancer_->AnalyzeCaptureAudio(
-        ca->split_channels_f(kBand0To8kHz), split_rate_, ca->num_channels());
+  if (constants_.intelligibility_enabled) {
+    public_submodules_->intelligibility_enhancer->AnalyzeCaptureAudio(
+        ca->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
+        ca->num_channels());
   }
 
-  if (beamformer_enabled_) {
-    beamformer_->ProcessChunk(*ca->split_data_f(), ca->split_data_f());
+  if (capture_nonlocked_.beamformer_enabled) {
+    private_submodules_->beamformer->ProcessChunk(*ca->split_data_f(),
+                                                  ca->split_data_f());
     ca->set_num_channels(1);
   }
 
-  RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
-  RETURN_ON_ERR(noise_suppression_->AnalyzeCaptureAudio(ca));
-  RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(ca));
+  public_submodules_->high_pass_filter->ProcessCaptureAudio(ca);
+  RETURN_ON_ERR(public_submodules_->gain_control->AnalyzeCaptureAudio(ca));
+  public_submodules_->noise_suppression->AnalyzeCaptureAudio(ca);
+  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(ca));
 
-  if (echo_control_mobile_->is_enabled() && noise_suppression_->is_enabled()) {
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
+      public_submodules_->noise_suppression->is_enabled()) {
     ca->CopyLowPassToReference();
   }
-  RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
+  public_submodules_->noise_suppression->ProcessCaptureAudio(ca);
+  RETURN_ON_ERR(
+      public_submodules_->echo_control_mobile->ProcessCaptureAudio(ca));
+  public_submodules_->voice_detection->ProcessCaptureAudio(ca);
 
-  if (use_new_agc_ && gain_control_->is_enabled() &&
-      (!beamformer_enabled_ || beamformer_->is_target_present())) {
-    agc_manager_->Process(ca->split_bands_const(0)[kBand0To8kHz],
-                          ca->num_frames_per_band(), split_rate_);
+  if (constants_.use_new_agc &&
+      public_submodules_->gain_control->is_enabled() &&
+      (!capture_nonlocked_.beamformer_enabled ||
+       private_submodules_->beamformer->is_target_present())) {
+    private_submodules_->agc_manager->Process(
+        ca->split_bands_const(0)[kBand0To8kHz], ca->num_frames_per_band(),
+        capture_nonlocked_.split_rate);
   }
-  RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(ca));
 
   if (synthesis_needed(data_processed)) {
     ca->MergeFrequencyBands();
@@ -687,21 +811,23 @@ int AudioProcessingImpl::ProcessStreamLocked() {
 
   // TODO(aluebs): Investigate if the transient suppression placement should be
   // before or after the AGC.
-  if (transient_suppressor_enabled_) {
+  if (capture_.transient_suppressor_enabled) {
     float voice_probability =
-        agc_manager_.get() ? agc_manager_->voice_probability() : 1.f;
+        private_submodules_->agc_manager.get()
+            ? private_submodules_->agc_manager->voice_probability()
+            : 1.f;
 
-    transient_suppressor_->Suppress(
+    public_submodules_->transient_suppressor->Suppress(
         ca->channels_f()[0], ca->num_frames(), ca->num_channels(),
         ca->split_bands_const_f(0)[kBand0To8kHz], ca->num_frames_per_band(),
         ca->keyboard_data(), ca->num_keyboard_frames(), voice_probability,
-        key_pressed_);
+        capture_.key_pressed);
   }
 
   // The level estimator operates on the recombined data.
-  RETURN_ON_ERR(level_estimator_->ProcessStream(ca));
+  public_submodules_->level_estimator->ProcessStream(ca);
 
-  was_stream_delay_set_ = false;
+  capture_.was_stream_delay_set = false;
   return kNoError;
 }
 
@@ -709,13 +835,15 @@ int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
                                               size_t samples_per_channel,
                                               int rev_sample_rate_hz,
                                               ChannelLayout layout) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_ChannelLayout");
+  rtc::CritScope cs(&crit_render_);
   const StreamConfig reverse_config = {
       rev_sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
   };
   if (samples_per_channel != reverse_config.num_frames()) {
     return kBadDataLengthError;
   }
-  return AnalyzeReverseStream(data, reverse_config, reverse_config);
+  return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config);
 }
 
 int AudioProcessingImpl::ProcessReverseStream(
@@ -723,13 +851,17 @@ int AudioProcessingImpl::ProcessReverseStream(
     const StreamConfig& reverse_input_config,
     const StreamConfig& reverse_output_config,
     float* const* dest) {
-  RETURN_ON_ERR(
-      AnalyzeReverseStream(src, reverse_input_config, reverse_output_config));
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig");
+  rtc::CritScope cs(&crit_render_);
+  RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, reverse_input_config,
+                                           reverse_output_config));
   if (is_rev_processed()) {
-    render_audio_->CopyTo(api_format_.reverse_output_stream(), dest);
-  } else if (rev_conversion_needed()) {
-    render_converter_->Convert(src, reverse_input_config.num_samples(), dest,
-                               reverse_output_config.num_samples());
+    render_.render_audio->CopyTo(formats_.api_format.reverse_output_stream(),
+                                 dest);
+  } else if (render_check_rev_conversion_needed()) {
+    render_.render_converter->Convert(src, reverse_input_config.num_samples(),
+                                      dest,
+                                      reverse_output_config.num_samples());
   } else {
     CopyAudioIfNeeded(src, reverse_input_config.num_frames(),
                       reverse_input_config.num_channels(), dest);
@@ -738,55 +870,61 @@ int AudioProcessingImpl::ProcessReverseStream(
   return kNoError;
 }
 
-int AudioProcessingImpl::AnalyzeReverseStream(
+int AudioProcessingImpl::AnalyzeReverseStreamLocked(
     const float* const* src,
     const StreamConfig& reverse_input_config,
     const StreamConfig& reverse_output_config) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (src == NULL) {
+  if (src == nullptr) {
     return kNullPointerError;
   }
 
-  if (reverse_input_config.num_channels() <= 0) {
+  if (reverse_input_config.num_channels() == 0) {
     return kBadNumberChannelsError;
   }
 
-  ProcessingConfig processing_config = api_format_;
+  ProcessingConfig processing_config = formats_.api_format;
   processing_config.reverse_input_stream() = reverse_input_config;
   processing_config.reverse_output_stream() = reverse_output_config;
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  RETURN_ON_ERR(MaybeInitializeRender(processing_config));
   assert(reverse_input_config.num_frames() ==
-         api_format_.reverse_input_stream().num_frames());
+         formats_.api_format.reverse_input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
-    audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
+    audioproc::ReverseStream* msg =
+        debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t channel_size =
-        sizeof(float) * api_format_.reverse_input_stream().num_frames();
-    for (int i = 0; i < api_format_.reverse_input_stream().num_channels(); ++i)
+        sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
+    for (size_t i = 0;
+         i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
       msg->add_channel(src[i], channel_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.render));
   }
 #endif
 
-  render_audio_->CopyFrom(src, api_format_.reverse_input_stream());
+  render_.render_audio->CopyFrom(src,
+                                 formats_.api_format.reverse_input_stream());
   return ProcessReverseStreamLocked();
 }
 
 int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
   RETURN_ON_ERR(AnalyzeReverseStream(frame));
+  rtc::CritScope cs(&crit_render_);
   if (is_rev_processed()) {
-    render_audio_->InterleaveTo(frame, true);
+    render_.render_audio->InterleaveTo(frame, true);
   }
 
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (frame == NULL) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_AudioFrame");
+  rtc::CritScope cs(&crit_render_);
+  if (frame == nullptr) {
     return kNullPointerError;
   }
   // Must be a native rate.
@@ -797,7 +935,8 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
     return kBadSampleRateError;
   }
   // This interface does not tolerate different forward and reverse rates.
-  if (frame->sample_rate_hz_ != api_format_.input_stream().sample_rate_hz()) {
+  if (frame->sample_rate_hz_ !=
+      formats_.api_format.input_stream().sample_rate_hz()) {
     return kBadSampleRateError;
   }
 
@@ -805,7 +944,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
     return kBadNumberChannelsError;
   }
 
-  ProcessingConfig processing_config = api_format_;
+  ProcessingConfig processing_config = formats_.api_format;
   processing_config.reverse_input_stream().set_sample_rate_hz(
       frame->sample_rate_hz_);
   processing_config.reverse_input_stream().set_num_channels(
@@ -815,44 +954,52 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
   processing_config.reverse_output_stream().set_num_channels(
       frame->num_channels_);
 
-  RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
+  RETURN_ON_ERR(MaybeInitializeRender(processing_config));
   if (frame->samples_per_channel_ !=
-      api_format_.reverse_input_stream().num_frames()) {
+      formats_.api_format.reverse_input_stream().num_frames()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
-    audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
+    audioproc::ReverseStream* msg =
+        debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_data(frame->data_, data_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.render));
   }
 #endif
-  render_audio_->DeinterleaveFrom(frame);
+  render_.render_audio->DeinterleaveFrom(frame);
   return ProcessReverseStreamLocked();
 }
 
 int AudioProcessingImpl::ProcessReverseStreamLocked() {
-  AudioBuffer* ra = render_audio_.get();  // For brevity.
-  if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz) {
+  AudioBuffer* ra = render_.render_audio.get();  // For brevity.
+  if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz) {
     ra->SplitIntoFrequencyBands();
   }
 
-  if (intelligibility_enabled_) {
-    intelligibility_enhancer_->ProcessRenderAudio(
-        ra->split_channels_f(kBand0To8kHz), split_rate_, ra->num_channels());
+  if (constants_.intelligibility_enabled) {
+    // Currently run in single-threaded mode when the intelligibility
+    // enhancer is activated.
+    // TODO(peah): Fix to be properly multi-threaded.
+    rtc::CritScope cs(&crit_capture_);
+    public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
+        ra->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
+        ra->num_channels());
   }
 
-  RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(ra));
-  RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(ra));
-  if (!use_new_agc_) {
-    RETURN_ON_ERR(gain_control_->ProcessRenderAudio(ra));
+  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessRenderAudio(ra));
+  RETURN_ON_ERR(
+      public_submodules_->echo_control_mobile->ProcessRenderAudio(ra));
+  if (!constants_.use_new_agc) {
+    RETURN_ON_ERR(public_submodules_->gain_control->ProcessRenderAudio(ra));
   }
 
-  if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz &&
+  if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
       is_rev_processed()) {
     ra->MergeFrequencyBands();
   }
@@ -861,9 +1008,10 @@ int AudioProcessingImpl::ProcessReverseStreamLocked() {
 }
 
 int AudioProcessingImpl::set_stream_delay_ms(int delay) {
+  rtc::CritScope cs(&crit_capture_);
   Error retval = kNoError;
-  was_stream_delay_set_ = true;
-  delay += delay_offset_ms_;
+  capture_.was_stream_delay_set = true;
+  delay += capture_.delay_offset_ms;
 
   if (delay < 0) {
     delay = 0;
@@ -876,50 +1024,56 @@ int AudioProcessingImpl::set_stream_delay_ms(int delay) {
     retval = kBadStreamParameterWarning;
   }
 
-  stream_delay_ms_ = delay;
+  capture_nonlocked_.stream_delay_ms = delay;
   return retval;
 }
 
 int AudioProcessingImpl::stream_delay_ms() const {
-  return stream_delay_ms_;
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.stream_delay_ms;
 }
 
 bool AudioProcessingImpl::was_stream_delay_set() const {
-  return was_stream_delay_set_;
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_.was_stream_delay_set;
 }
 
 void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
-  key_pressed_ = key_pressed;
+  rtc::CritScope cs(&crit_capture_);
+  capture_.key_pressed = key_pressed;
 }
 
 void AudioProcessingImpl::set_delay_offset_ms(int offset) {
-  CriticalSectionScoped crit_scoped(crit_);
-  delay_offset_ms_ = offset;
+  rtc::CritScope cs(&crit_capture_);
+  capture_.delay_offset_ms = offset;
 }
 
 int AudioProcessingImpl::delay_offset_ms() const {
-  return delay_offset_ms_;
+  rtc::CritScope cs(&crit_capture_);
+  return capture_.delay_offset_ms;
 }
 
 int AudioProcessingImpl::StartDebugRecording(
     const char filename[AudioProcessing::kMaxFilenameSize]) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
 
-  if (filename == NULL) {
+  if (filename == nullptr) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
-  if (debug_file_->OpenFile(filename, false) == -1) {
-    debug_file_->CloseFile();
+  if (debug_dump_.debug_file->OpenFile(filename, false) == -1) {
+    debug_dump_.debug_file->CloseFile();
     return kFileError;
   }
 
@@ -932,21 +1086,23 @@ int AudioProcessingImpl::StartDebugRecording(
 }
 
 int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
 
-  if (handle == NULL) {
+  if (handle == nullptr) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
-  if (debug_file_->OpenFromFileHandle(handle, true, false) == -1) {
+  if (debug_dump_.debug_file->OpenFromFileHandle(handle, true, false) == -1) {
     return kFileError;
   }
 
@@ -960,17 +1116,22 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
 
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
   return StartDebugRecording(stream);
 }
 
 int AudioProcessingImpl::StopDebugRecording() {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
@@ -981,58 +1142,87 @@ int AudioProcessingImpl::StopDebugRecording() {
 }
 
 EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
-  return echo_cancellation_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->echo_cancellation;
 }
 
 EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
-  return echo_control_mobile_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->echo_control_mobile;
 }
 
 GainControl* AudioProcessingImpl::gain_control() const {
-  if (use_new_agc_) {
-    return gain_control_for_new_agc_.get();
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  if (constants_.use_new_agc) {
+    return public_submodules_->gain_control_for_new_agc.get();
   }
-  return gain_control_;
+  return public_submodules_->gain_control;
 }
 
 HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
-  return high_pass_filter_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->high_pass_filter.get();
 }
 
 LevelEstimator* AudioProcessingImpl::level_estimator() const {
-  return level_estimator_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->level_estimator.get();
 }
 
 NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
-  return noise_suppression_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->noise_suppression.get();
 }
 
 VoiceDetection* AudioProcessingImpl::voice_detection() const {
-  return voice_detection_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->voice_detection.get();
 }
 
 bool AudioProcessingImpl::is_data_processed() const {
-  if (beamformer_enabled_) {
+  if (capture_nonlocked_.beamformer_enabled) {
     return true;
   }
 
   int enabled_count = 0;
-  for (auto item : component_list_) {
+  for (auto item : private_submodules_->component_list) {
     if (item->is_component_enabled()) {
       enabled_count++;
     }
   }
+  if (public_submodules_->high_pass_filter->is_enabled()) {
+    enabled_count++;
+  }
+  if (public_submodules_->noise_suppression->is_enabled()) {
+    enabled_count++;
+  }
+  if (public_submodules_->level_estimator->is_enabled()) {
+    enabled_count++;
+  }
+  if (public_submodules_->voice_detection->is_enabled()) {
+    enabled_count++;
+  }
 
-  // Data is unchanged if no components are enabled, or if only level_estimator_
-  // or voice_detection_ is enabled.
+  // Data is unchanged if no components are enabled, or if only
+  // public_submodules_->level_estimator
+  // or public_submodules_->voice_detection is enabled.
   if (enabled_count == 0) {
     return false;
   } else if (enabled_count == 1) {
-    if (level_estimator_->is_enabled() || voice_detection_->is_enabled()) {
+    if (public_submodules_->level_estimator->is_enabled() ||
+        public_submodules_->voice_detection->is_enabled()) {
       return false;
     }
   } else if (enabled_count == 2) {
-    if (level_estimator_->is_enabled() && voice_detection_->is_enabled()) {
+    if (public_submodules_->level_estimator->is_enabled() &&
+        public_submodules_->voice_detection->is_enabled()) {
       return false;
     }
   }
@@ -1041,149 +1231,194 @@ bool AudioProcessingImpl::is_data_processed() const {
 
 bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
   // Check if we've upmixed or downmixed the audio.
-  return ((api_format_.output_stream().num_channels() !=
-           api_format_.input_stream().num_channels()) ||
-          is_data_processed || transient_suppressor_enabled_);
+  return ((formats_.api_format.output_stream().num_channels() !=
+           formats_.api_format.input_stream().num_channels()) ||
+          is_data_processed || capture_.transient_suppressor_enabled);
 }
 
 bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
   return (is_data_processed &&
-          (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
-           fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz));
+          (capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+               kSampleRate32kHz ||
+           capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+               kSampleRate48kHz));
 }
 
 bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
-  if (!is_data_processed && !voice_detection_->is_enabled() &&
-      !transient_suppressor_enabled_) {
-    // Only level_estimator_ is enabled.
+  if (!is_data_processed &&
+      !public_submodules_->voice_detection->is_enabled() &&
+      !capture_.transient_suppressor_enabled) {
+    // Only public_submodules_->level_estimator is enabled.
     return false;
-  } else if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
-             fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
-    // Something besides level_estimator_ is enabled, and we have super-wb.
+  } else if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+                 kSampleRate32kHz ||
+             capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+                 kSampleRate48kHz) {
+    // Something besides public_submodules_->level_estimator is enabled, and we
+    // have super-wb.
     return true;
   }
   return false;
 }
 
 bool AudioProcessingImpl::is_rev_processed() const {
-  return intelligibility_enabled_ && intelligibility_enhancer_->active();
+  return constants_.intelligibility_enabled &&
+         public_submodules_->intelligibility_enhancer->active();
+}
+
+bool AudioProcessingImpl::render_check_rev_conversion_needed() const {
+  return rev_conversion_needed();
 }
 
 bool AudioProcessingImpl::rev_conversion_needed() const {
-  return (api_format_.reverse_input_stream() !=
-          api_format_.reverse_output_stream());
+  return (formats_.api_format.reverse_input_stream() !=
+          formats_.api_format.reverse_output_stream());
 }
 
 void AudioProcessingImpl::InitializeExperimentalAgc() {
-  if (use_new_agc_) {
-    if (!agc_manager_.get()) {
-      agc_manager_.reset(new AgcManagerDirect(gain_control_,
-                                              gain_control_for_new_agc_.get(),
-                                              agc_startup_min_volume_));
+  if (constants_.use_new_agc) {
+    if (!private_submodules_->agc_manager.get()) {
+      private_submodules_->agc_manager.reset(new AgcManagerDirect(
+          public_submodules_->gain_control,
+          public_submodules_->gain_control_for_new_agc.get(),
+          constants_.agc_startup_min_volume));
     }
-    agc_manager_->Initialize();
-    agc_manager_->SetCaptureMuted(output_will_be_muted_);
+    private_submodules_->agc_manager->Initialize();
+    private_submodules_->agc_manager->SetCaptureMuted(
+        capture_.output_will_be_muted);
   }
 }
 
 void AudioProcessingImpl::InitializeTransient() {
-  if (transient_suppressor_enabled_) {
-    if (!transient_suppressor_.get()) {
-      transient_suppressor_.reset(new TransientSuppressor());
+  if (capture_.transient_suppressor_enabled) {
+    if (!public_submodules_->transient_suppressor.get()) {
+      public_submodules_->transient_suppressor.reset(new TransientSuppressor());
     }
-    transient_suppressor_->Initialize(
-        fwd_proc_format_.sample_rate_hz(), split_rate_,
-        api_format_.output_stream().num_channels());
+    public_submodules_->transient_suppressor->Initialize(
+        capture_nonlocked_.fwd_proc_format.sample_rate_hz(),
+        capture_nonlocked_.split_rate,
+        num_proc_channels());
   }
 }
 
 void AudioProcessingImpl::InitializeBeamformer() {
-  if (beamformer_enabled_) {
-    if (!beamformer_) {
-      beamformer_.reset(
-          new NonlinearBeamformer(array_geometry_, target_direction_));
+  if (capture_nonlocked_.beamformer_enabled) {
+    if (!private_submodules_->beamformer) {
+      private_submodules_->beamformer.reset(new NonlinearBeamformer(
+          capture_.array_geometry, capture_.target_direction));
     }
-    beamformer_->Initialize(kChunkSizeMs, split_rate_);
+    private_submodules_->beamformer->Initialize(kChunkSizeMs,
+                                                capture_nonlocked_.split_rate);
   }
 }
 
 void AudioProcessingImpl::InitializeIntelligibility() {
-  if (intelligibility_enabled_) {
+  if (constants_.intelligibility_enabled) {
     IntelligibilityEnhancer::Config config;
-    config.sample_rate_hz = split_rate_;
-    config.num_capture_channels = capture_audio_->num_channels();
-    config.num_render_channels = render_audio_->num_channels();
-    intelligibility_enhancer_.reset(new IntelligibilityEnhancer(config));
+    config.sample_rate_hz = capture_nonlocked_.split_rate;
+    config.num_capture_channels = capture_.capture_audio->num_channels();
+    config.num_render_channels = render_.render_audio->num_channels();
+    public_submodules_->intelligibility_enhancer.reset(
+        new IntelligibilityEnhancer(config));
   }
 }
 
+void AudioProcessingImpl::InitializeHighPassFilter() {
+  public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
+                                                   proc_sample_rate_hz());
+}
+
+void AudioProcessingImpl::InitializeNoiseSuppression() {
+  public_submodules_->noise_suppression->Initialize(num_proc_channels(),
+                                                    proc_sample_rate_hz());
+}
+
+void AudioProcessingImpl::InitializeLevelEstimator() {
+  public_submodules_->level_estimator->Initialize();
+}
+
+void AudioProcessingImpl::InitializeVoiceDetection() {
+  public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz());
+}
+
 void AudioProcessingImpl::MaybeUpdateHistograms() {
   static const int kMinDiffDelayMs = 60;
 
   if (echo_cancellation()->is_enabled()) {
     // Activate delay_jumps_ counters if we know echo_cancellation is runnning.
     // If a stream has echo we know that the echo_cancellation is in process.
-    if (stream_delay_jumps_ == -1 && echo_cancellation()->stream_has_echo()) {
-      stream_delay_jumps_ = 0;
+    if (capture_.stream_delay_jumps == -1 &&
+        echo_cancellation()->stream_has_echo()) {
+      capture_.stream_delay_jumps = 0;
     }
-    if (aec_system_delay_jumps_ == -1 &&
+    if (capture_.aec_system_delay_jumps == -1 &&
         echo_cancellation()->stream_has_echo()) {
-      aec_system_delay_jumps_ = 0;
+      capture_.aec_system_delay_jumps = 0;
     }
 
     // Detect a jump in platform reported system delay and log the difference.
-    const int diff_stream_delay_ms = stream_delay_ms_ - last_stream_delay_ms_;
-    if (diff_stream_delay_ms > kMinDiffDelayMs && last_stream_delay_ms_ != 0) {
-      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
-                           diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
-      if (stream_delay_jumps_ == -1) {
-        stream_delay_jumps_ = 0;  // Activate counter if needed.
+    const int diff_stream_delay_ms =
+        capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
+    if (diff_stream_delay_ms > kMinDiffDelayMs &&
+        capture_.last_stream_delay_ms != 0) {
+      RTC_HISTOGRAM_COUNTS_SPARSE(
+          "WebRTC.Audio.PlatformReportedStreamDelayJump", diff_stream_delay_ms,
+          kMinDiffDelayMs, 1000, 100);
+      if (capture_.stream_delay_jumps == -1) {
+        capture_.stream_delay_jumps = 0;  // Activate counter if needed.
       }
-      stream_delay_jumps_++;
+      capture_.stream_delay_jumps++;
     }
-    last_stream_delay_ms_ = stream_delay_ms_;
+    capture_.last_stream_delay_ms = capture_nonlocked_.stream_delay_ms;
 
     // Detect a jump in AEC system delay and log the difference.
-    const int frames_per_ms = rtc::CheckedDivExact(split_rate_, 1000);
+    const int frames_per_ms =
+        rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
     const int aec_system_delay_ms =
         WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
     const int diff_aec_system_delay_ms =
-        aec_system_delay_ms - last_aec_system_delay_ms_;
+        aec_system_delay_ms - capture_.last_aec_system_delay_ms;
     if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
-        last_aec_system_delay_ms_ != 0) {
-      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
-                           diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
-                           100);
-      if (aec_system_delay_jumps_ == -1) {
-        aec_system_delay_jumps_ = 0;  // Activate counter if needed.
+        capture_.last_aec_system_delay_ms != 0) {
+      RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.AecSystemDelayJump",
+                                  diff_aec_system_delay_ms, kMinDiffDelayMs,
+                                  1000, 100);
+      if (capture_.aec_system_delay_jumps == -1) {
+        capture_.aec_system_delay_jumps = 0;  // Activate counter if needed.
       }
-      aec_system_delay_jumps_++;
+      capture_.aec_system_delay_jumps++;
     }
-    last_aec_system_delay_ms_ = aec_system_delay_ms;
+    capture_.last_aec_system_delay_ms = aec_system_delay_ms;
   }
 }
 
 void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (stream_delay_jumps_ > -1) {
-    RTC_HISTOGRAM_ENUMERATION(
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+
+  if (capture_.stream_delay_jumps > -1) {
+    RTC_HISTOGRAM_ENUMERATION_SPARSE(
         "WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
-        stream_delay_jumps_, 51);
+        capture_.stream_delay_jumps, 51);
   }
-  stream_delay_jumps_ = -1;
-  last_stream_delay_ms_ = 0;
+  capture_.stream_delay_jumps = -1;
+  capture_.last_stream_delay_ms = 0;
 
-  if (aec_system_delay_jumps_ > -1) {
-    RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
-                              aec_system_delay_jumps_, 51);
+  if (capture_.aec_system_delay_jumps > -1) {
+    RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Audio.NumOfAecSystemDelayJumps",
+                                     capture_.aec_system_delay_jumps, 51);
   }
-  aec_system_delay_jumps_ = -1;
-  last_aec_system_delay_ms_ = 0;
+  capture_.aec_system_delay_jumps = -1;
+  capture_.last_aec_system_delay_ms = 0;
 }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-int AudioProcessingImpl::WriteMessageToDebugFile() {
-  int32_t size = event_msg_->ByteSize();
+int AudioProcessingImpl::WriteMessageToDebugFile(
+    FileWrapper* debug_file,
+    rtc::CriticalSection* crit_debug,
+    ApmDebugDumpThreadState* debug_state) {
+  int32_t size = debug_state->event_msg->ByteSize();
   if (size <= 0) {
     return kUnspecifiedError;
   }
@@ -1192,82 +1427,100 @@ int AudioProcessingImpl::WriteMessageToDebugFile() {
 //            pretty safe in assuming little-endian.
 #endif
 
-  if (!event_msg_->SerializeToString(&event_str_)) {
+  if (!debug_state->event_msg->SerializeToString(&debug_state->event_str)) {
     return kUnspecifiedError;
   }
 
-  // Write message preceded by its size.
-  if (!debug_file_->Write(&size, sizeof(int32_t))) {
-    return kFileError;
-  }
-  if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
-    return kFileError;
+  {
+    // Ensure atomic writes of the message.
+    rtc::CritScope cs_capture(crit_debug);
+    // Write message preceded by its size.
+    if (!debug_file->Write(&size, sizeof(int32_t))) {
+      return kFileError;
+    }
+    if (!debug_file->Write(debug_state->event_str.data(),
+                           debug_state->event_str.length())) {
+      return kFileError;
+    }
   }
 
-  event_msg_->Clear();
+  debug_state->event_msg->Clear();
 
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteInitMessage() {
-  event_msg_->set_type(audioproc::Event::INIT);
-  audioproc::Init* msg = event_msg_->mutable_init();
-  msg->set_sample_rate(api_format_.input_stream().sample_rate_hz());
-  msg->set_num_input_channels(api_format_.input_stream().num_channels());
-  msg->set_num_output_channels(api_format_.output_stream().num_channels());
-  msg->set_num_reverse_channels(
-      api_format_.reverse_input_stream().num_channels());
+  debug_dump_.capture.event_msg->set_type(audioproc::Event::INIT);
+  audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
+  msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
+
+  msg->set_num_input_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.input_stream().num_channels()));
+  msg->set_num_output_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.output_stream().num_channels()));
+  msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.reverse_input_stream().num_channels()));
   msg->set_reverse_sample_rate(
-      api_format_.reverse_input_stream().sample_rate_hz());
-  msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
-  // TODO(ekmeyerson): Add reverse output fields to event_msg_.
-
-  RETURN_ON_ERR(WriteMessageToDebugFile());
+      formats_.api_format.reverse_input_stream().sample_rate_hz());
+  msg->set_output_sample_rate(
+      formats_.api_format.output_stream().sample_rate_hz());
+  // TODO(ekmeyerson): Add reverse output fields to
+  // debug_dump_.capture.event_msg.
+
+  RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   audioproc::Config config;
 
-  config.set_aec_enabled(echo_cancellation_->is_enabled());
+  config.set_aec_enabled(public_submodules_->echo_cancellation->is_enabled());
   config.set_aec_delay_agnostic_enabled(
-      echo_cancellation_->is_delay_agnostic_enabled());
+      public_submodules_->echo_cancellation->is_delay_agnostic_enabled());
   config.set_aec_drift_compensation_enabled(
-      echo_cancellation_->is_drift_compensation_enabled());
+      public_submodules_->echo_cancellation->is_drift_compensation_enabled());
   config.set_aec_extended_filter_enabled(
-      echo_cancellation_->is_extended_filter_enabled());
-  config.set_aec_suppression_level(
-      static_cast<int>(echo_cancellation_->suppression_level()));
+      public_submodules_->echo_cancellation->is_extended_filter_enabled());
+  config.set_aec_suppression_level(static_cast<int>(
+      public_submodules_->echo_cancellation->suppression_level()));
 
-  config.set_aecm_enabled(echo_control_mobile_->is_enabled());
+  config.set_aecm_enabled(
+      public_submodules_->echo_control_mobile->is_enabled());
   config.set_aecm_comfort_noise_enabled(
-      echo_control_mobile_->is_comfort_noise_enabled());
-  config.set_aecm_routing_mode(
-      static_cast<int>(echo_control_mobile_->routing_mode()));
+      public_submodules_->echo_control_mobile->is_comfort_noise_enabled());
+  config.set_aecm_routing_mode(static_cast<int>(
+      public_submodules_->echo_control_mobile->routing_mode()));
 
-  config.set_agc_enabled(gain_control_->is_enabled());
-  config.set_agc_mode(static_cast<int>(gain_control_->mode()));
-  config.set_agc_limiter_enabled(gain_control_->is_limiter_enabled());
-  config.set_noise_robust_agc_enabled(use_new_agc_);
+  config.set_agc_enabled(public_submodules_->gain_control->is_enabled());
+  config.set_agc_mode(
+      static_cast<int>(public_submodules_->gain_control->mode()));
+  config.set_agc_limiter_enabled(
+      public_submodules_->gain_control->is_limiter_enabled());
+  config.set_noise_robust_agc_enabled(constants_.use_new_agc);
 
-  config.set_hpf_enabled(high_pass_filter_->is_enabled());
+  config.set_hpf_enabled(public_submodules_->high_pass_filter->is_enabled());
 
-  config.set_ns_enabled(noise_suppression_->is_enabled());
-  config.set_ns_level(static_cast<int>(noise_suppression_->level()));
+  config.set_ns_enabled(public_submodules_->noise_suppression->is_enabled());
+  config.set_ns_level(
+      static_cast<int>(public_submodules_->noise_suppression->level()));
 
-  config.set_transient_suppression_enabled(transient_suppressor_enabled_);
+  config.set_transient_suppression_enabled(
+      capture_.transient_suppressor_enabled);
 
   std::string serialized_config = config.SerializeAsString();
-  if (!forced && last_serialized_config_ == serialized_config) {
+  if (!forced &&
+      debug_dump_.capture.last_serialized_config == serialized_config) {
     return kNoError;
   }
 
-  last_serialized_config_ = serialized_config;
+  debug_dump_.capture.last_serialized_config = serialized_config;
 
-  event_msg_->set_type(audioproc::Event::CONFIG);
-  event_msg_->mutable_config()->CopyFrom(config);
+  debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
+  debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
-  RETURN_ON_ERR(WriteMessageToDebugFile());
+  RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 542886ee10..b310896903 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -15,50 +15,38 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 namespace webrtc {
 
 class AgcManagerDirect;
-class AudioBuffer;
 class AudioConverter;
 
 template<typename T>
 class Beamformer;
 
-class CriticalSectionWrapper;
-class EchoCancellationImpl;
-class EchoControlMobileImpl;
-class FileWrapper;
-class GainControlImpl;
-class GainControlForNewAgc;
-class HighPassFilterImpl;
-class LevelEstimatorImpl;
-class NoiseSuppressionImpl;
-class ProcessingComponent;
-class TransientSuppressor;
-class VoiceDetectionImpl;
-class IntelligibilityEnhancer;
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-namespace audioproc {
-
-class Event;
-
-}  // namespace audioproc
-#endif
-
 class AudioProcessingImpl : public AudioProcessing {
  public:
+  // Methods forcing APM to run in a single-threaded manner.
+  // Acquires both the render and capture locks.
   explicit AudioProcessingImpl(const Config& config);
-
   // AudioProcessingImpl takes ownership of beamformer.
   AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
   virtual ~AudioProcessingImpl();
-
-  // AudioProcessing methods.
   int Initialize() override;
   int Initialize(int input_sample_rate_hz,
                  int output_sample_rate_hz,
@@ -68,12 +56,14 @@ class AudioProcessingImpl : public AudioProcessing {
                  ChannelLayout reverse_layout) override;
   int Initialize(const ProcessingConfig& processing_config) override;
   void SetExtraOptions(const Config& config) override;
-  int proc_sample_rate_hz() const override;
-  int proc_split_sample_rate_hz() const override;
-  int num_input_channels() const override;
-  int num_output_channels() const override;
-  int num_reverse_channels() const override;
-  void set_output_will_be_muted(bool muted) override;
+  void UpdateHistogramsOnCallEnd() override;
+  int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
+  int StartDebugRecording(FILE* handle) override;
+  int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
+  int StopDebugRecording() override;
+
+  // Capture-side exclusive methods possibly running APM in a
+  // multi-threaded manner. Acquire the capture lock.
   int ProcessStream(AudioFrame* frame) override;
   int ProcessStream(const float* const* src,
                     size_t samples_per_channel,
@@ -86,6 +76,15 @@ class AudioProcessingImpl : public AudioProcessing {
                     const StreamConfig& input_config,
                     const StreamConfig& output_config,
                     float* const* dest) override;
+  void set_output_will_be_muted(bool muted) override;
+  int set_stream_delay_ms(int delay) override;
+  void set_delay_offset_ms(int offset) override;
+  int delay_offset_ms() const override;
+  void set_stream_key_pressed(bool key_pressed) override;
+  int input_sample_rate_hz() const override;
+
+  // Render-side exclusive methods possibly running APM in a
+  // multi-threaded manner. Acquire the render lock.
   int AnalyzeReverseStream(AudioFrame* frame) override;
   int ProcessReverseStream(AudioFrame* frame) override;
   int AnalyzeReverseStream(const float* const* data,
@@ -96,17 +95,25 @@ class AudioProcessingImpl : public AudioProcessing {
                            const StreamConfig& reverse_input_config,
                            const StreamConfig& reverse_output_config,
                            float* const* dest) override;
-  int set_stream_delay_ms(int delay) override;
+
+  // Methods only accessed from APM submodules or
+  // from AudioProcessing tests in a single-threaded manner.
+  // Hence there is no need for locks in these.
+  int proc_sample_rate_hz() const override;
+  int proc_split_sample_rate_hz() const override;
+  size_t num_input_channels() const override;
+  size_t num_proc_channels() const override;
+  size_t num_output_channels() const override;
+  size_t num_reverse_channels() const override;
   int stream_delay_ms() const override;
-  bool was_stream_delay_set() const override;
-  void set_delay_offset_ms(int offset) override;
-  int delay_offset_ms() const override;
-  void set_stream_key_pressed(bool key_pressed) override;
-  int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
-  int StartDebugRecording(FILE* handle) override;
-  int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
-  int StopDebugRecording() override;
-  void UpdateHistogramsOnCallEnd() override;
+  bool was_stream_delay_set() const override
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Methods returning pointers to APM submodules.
+  // No locks are aquired in those, as those locks
+  // would offer no protection (the submodules are
+  // created only once in a single-treaded manner
+  // during APM creation).
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
@@ -117,101 +124,216 @@ class AudioProcessingImpl : public AudioProcessing {
 
  protected:
   // Overridden in a mock.
-  virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  virtual int InitializeLocked()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
  private:
+  struct ApmPublicSubmodules;
+  struct ApmPrivateSubmodules;
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  // State for the debug dump.
+  struct ApmDebugDumpThreadState {
+    ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {}
+    rtc::scoped_ptr<audioproc::Event> event_msg;  // Protobuf message.
+    std::string event_str;  // Memory for protobuf serialization.
+
+    // Serialized string of last saved APM configuration.
+    std::string last_serialized_config;
+  };
+
+  struct ApmDebugDumpState {
+    ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
+    rtc::scoped_ptr<FileWrapper> debug_file;
+    ApmDebugDumpThreadState render;
+    ApmDebugDumpThreadState capture;
+  };
+#endif
+
+  // Method for modifying the formats struct that are called from both
+  // the render and capture threads. The check for whether modifications
+  // are needed is done while holding the render lock only, thereby avoiding
+  // that the capture thread blocks the render thread.
+  // The struct is modified in a single-threaded manner by holding both the
+  // render and capture locks.
+  int MaybeInitialize(const ProcessingConfig& config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  int MaybeInitializeRender(const ProcessingConfig& processing_config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  int MaybeInitializeCapture(const ProcessingConfig& processing_config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  // Method for checking for the need of conversion. Accesses the formats
+  // structs in a read manner but the requirement for the render lock to be held
+  // was added as it currently anyway is always called in that manner.
+  bool rev_conversion_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  bool render_check_rev_conversion_needed() const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  // Methods requiring APM running in a single-threaded manner.
+  // Are called with both the render and capture locks already
+  // acquired.
+  void InitializeExperimentalAgc()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeTransient()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeBeamformer()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeIntelligibility()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeHighPassFilter()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void InitializeNoiseSuppression()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void InitializeLevelEstimator()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void InitializeVoiceDetection()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   int InitializeLocked(const ProcessingConfig& config)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int MaybeInitializeLocked(const ProcessingConfig& config)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+
+  // Capture-side exclusive methods possibly running APM in a multi-threaded
+  // manner that are called with the render lock already acquired.
+  int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool output_copy_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool is_data_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool synthesis_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool analysis_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Render-side exclusive methods possibly running APM in a multi-threaded
+  // manner that are called with the render lock already acquired.
   // TODO(ekm): Remove once all clients updated to new interface.
-  int AnalyzeReverseStream(const float* const* src,
-                           const StreamConfig& input_config,
-                           const StreamConfig& output_config);
-  int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-
-  bool is_data_processed() const;
-  bool output_copy_needed(bool is_data_processed) const;
-  bool synthesis_needed(bool is_data_processed) const;
-  bool analysis_needed(bool is_data_processed) const;
-  bool is_rev_processed() const;
-  bool rev_conversion_needed() const;
-  void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-
-  EchoCancellationImpl* echo_cancellation_;
-  EchoControlMobileImpl* echo_control_mobile_;
-  GainControlImpl* gain_control_;
-  HighPassFilterImpl* high_pass_filter_;
-  LevelEstimatorImpl* level_estimator_;
-  NoiseSuppressionImpl* noise_suppression_;
-  VoiceDetectionImpl* voice_detection_;
-  rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
-
-  std::list<ProcessingComponent*> component_list_;
-  CriticalSectionWrapper* crit_;
-  rtc::scoped_ptr<AudioBuffer> render_audio_;
-  rtc::scoped_ptr<AudioBuffer> capture_audio_;
-  rtc::scoped_ptr<AudioConverter> render_converter_;
+  int AnalyzeReverseStreamLocked(const float* const* src,
+                                 const StreamConfig& input_config,
+                                 const StreamConfig& output_config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  bool is_rev_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+// Debug dump methods that are internal and called without locks.
+// TODO(peah): Make thread safe.
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // TODO(andrew): make this more graceful. Ideally we would split this stuff
   // out into a separate class with an "enabled" and "disabled" implementation.
-  int WriteMessageToDebugFile();
-  int WriteInitMessage();
+  static int WriteMessageToDebugFile(FileWrapper* debug_file,
+                                     rtc::CriticalSection* crit_debug,
+                                     ApmDebugDumpThreadState* debug_state);
+  int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
   // Writes Config message. If not |forced|, only writes the current config if
   // it is different from the last saved one; if |forced|, writes the config
   // regardless of the last saved.
-  int WriteConfigMessage(bool forced);
+  int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
-  rtc::scoped_ptr<FileWrapper> debug_file_;
-  rtc::scoped_ptr<audioproc::Event> event_msg_;  // Protobuf message.
-  std::string event_str_;  // Memory for protobuf serialization.
+  // Critical section.
+  mutable rtc::CriticalSection crit_debug_;
 
-  // Serialized string of last saved APM configuration.
-  std::string last_serialized_config_;
+  // Debug dump state.
+  ApmDebugDumpState debug_dump_;
 #endif
 
-  // Format of processing streams at input/output call sites.
-  ProcessingConfig api_format_;
-
-  // Only the rate and samples fields of fwd_proc_format_ are used because the
-  // forward processing number of channels is mutable and is tracked by the
-  // capture_audio_.
-  StreamConfig fwd_proc_format_;
-  StreamConfig rev_proc_format_;
-  int split_rate_;
-
-  int stream_delay_ms_;
-  int delay_offset_ms_;
-  bool was_stream_delay_set_;
-  int last_stream_delay_ms_;
-  int last_aec_system_delay_ms_;
-  int stream_delay_jumps_;
-  int aec_system_delay_jumps_;
-
-  bool output_will_be_muted_ GUARDED_BY(crit_);
-
-  bool key_pressed_;
-
-  // Only set through the constructor's Config parameter.
-  const bool use_new_agc_;
-  rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
-  int agc_startup_min_volume_;
-
-  bool transient_suppressor_enabled_;
-  rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
-  const bool beamformer_enabled_;
-  rtc::scoped_ptr<Beamformer<float>> beamformer_;
-  const std::vector<Point> array_geometry_;
-  const SphericalPointf target_direction_;
-
-  bool intelligibility_enabled_;
-  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
+  // Critical sections.
+  mutable rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  mutable rtc::CriticalSection crit_capture_;
+
+  // Structs containing the pointers to the submodules.
+  rtc::scoped_ptr<ApmPublicSubmodules> public_submodules_;
+  rtc::scoped_ptr<ApmPrivateSubmodules> private_submodules_
+      GUARDED_BY(crit_capture_);
+
+  // State that is written to while holding both the render and capture locks
+  // but can be read without any lock being held.
+  // As this is only accessed internally of APM, and all internal methods in APM
+  // either are holding the render or capture locks, this construct is safe as
+  // it is not possible to read the variables while writing them.
+  struct ApmFormatState {
+    ApmFormatState()
+        :  // Format of processing streams at input/output call sites.
+          api_format({{{kSampleRate16kHz, 1, false},
+                       {kSampleRate16kHz, 1, false},
+                       {kSampleRate16kHz, 1, false},
+                       {kSampleRate16kHz, 1, false}}}),
+          rev_proc_format(kSampleRate16kHz, 1) {}
+    ProcessingConfig api_format;
+    StreamConfig rev_proc_format;
+  } formats_;
+
+  // APM constants.
+  const struct ApmConstants {
+    ApmConstants(int agc_startup_min_volume,
+                 bool use_new_agc,
+                 bool intelligibility_enabled)
+        :  // Format of processing streams at input/output call sites.
+          agc_startup_min_volume(agc_startup_min_volume),
+          use_new_agc(use_new_agc),
+          intelligibility_enabled(intelligibility_enabled) {}
+    int agc_startup_min_volume;
+    bool use_new_agc;
+    bool intelligibility_enabled;
+  } constants_;
+
+  struct ApmCaptureState {
+    ApmCaptureState(bool transient_suppressor_enabled,
+                    const std::vector<Point>& array_geometry,
+                    SphericalPointf target_direction)
+        : aec_system_delay_jumps(-1),
+          delay_offset_ms(0),
+          was_stream_delay_set(false),
+          last_stream_delay_ms(0),
+          last_aec_system_delay_ms(0),
+          stream_delay_jumps(-1),
+          output_will_be_muted(false),
+          key_pressed(false),
+          transient_suppressor_enabled(transient_suppressor_enabled),
+          array_geometry(array_geometry),
+          target_direction(target_direction),
+          fwd_proc_format(kSampleRate16kHz),
+          split_rate(kSampleRate16kHz) {}
+    int aec_system_delay_jumps;
+    int delay_offset_ms;
+    bool was_stream_delay_set;
+    int last_stream_delay_ms;
+    int last_aec_system_delay_ms;
+    int stream_delay_jumps;
+    bool output_will_be_muted;
+    bool key_pressed;
+    bool transient_suppressor_enabled;
+    std::vector<Point> array_geometry;
+    SphericalPointf target_direction;
+    rtc::scoped_ptr<AudioBuffer> capture_audio;
+    // Only the rate and samples fields of fwd_proc_format_ are used because the
+    // forward processing number of channels is mutable and is tracked by the
+    // capture_audio_.
+    StreamConfig fwd_proc_format;
+    int split_rate;
+  } capture_ GUARDED_BY(crit_capture_);
+
+  struct ApmCaptureNonLockedState {
+    ApmCaptureNonLockedState(bool beamformer_enabled)
+        : fwd_proc_format(kSampleRate16kHz),
+          split_rate(kSampleRate16kHz),
+          stream_delay_ms(0),
+          beamformer_enabled(beamformer_enabled) {}
+    // Only the rate and samples fields of fwd_proc_format_ are used because the
+    // forward processing number of channels is mutable and is tracked by the
+    // capture_audio_.
+    StreamConfig fwd_proc_format;
+    int split_rate;
+    int stream_delay_ms;
+    bool beamformer_enabled;
+  } capture_nonlocked_;
+
+  struct ApmRenderState {
+    rtc::scoped_ptr<AudioConverter> render_converter;
+    rtc::scoped_ptr<AudioBuffer> render_audio;
+  } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
new file mode 100644
index 0000000000..e1e6a310a5
--- /dev/null
+++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -0,0 +1,1133 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/random.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+namespace {
+
+class AudioProcessingImplLockTest;
+
+// Type of the render thread APM API call to use in the test.
+enum class RenderApiImpl {
+  ProcessReverseStreamImpl1,
+  ProcessReverseStreamImpl2,
+  AnalyzeReverseStreamImpl1,
+  AnalyzeReverseStreamImpl2
+};
+
+// Type of the capture thread APM API call to use in the test.
+enum class CaptureApiImpl {
+  ProcessStreamImpl1,
+  ProcessStreamImpl2,
+  ProcessStreamImpl3
+};
+
+// The runtime parameter setting scheme to use in the test.
+enum class RuntimeParameterSettingScheme {
+  SparseStreamMetadataChangeScheme,
+  ExtremeStreamMetadataChangeScheme,
+  FixedMonoStreamMetadataScheme,
+  FixedStereoStreamMetadataScheme
+};
+
+// Variant of echo canceller settings to use in the test.
+enum class AecType {
+  BasicWebRtcAecSettings,
+  AecTurnedOff,
+  BasicWebRtcAecSettingsWithExtentedFilter,
+  BasicWebRtcAecSettingsWithDelayAgnosticAec,
+  BasicWebRtcAecSettingsWithAecMobile
+};
+
+// Thread-safe random number generator wrapper.
+class RandomGenerator {
+ public:
+  RandomGenerator() : rand_gen_(42U) {}
+
+  int RandInt(int min, int max) {
+    rtc::CritScope cs(&crit_);
+    return rand_gen_.Rand(min, max);
+  }
+
+  int RandInt(int max) {
+    rtc::CritScope cs(&crit_);
+    return rand_gen_.Rand(max);
+  }
+
+  float RandFloat() {
+    rtc::CritScope cs(&crit_);
+    return rand_gen_.Rand<float>();
+  }
+
+ private:
+  rtc::CriticalSection crit_;
+  Random rand_gen_ GUARDED_BY(crit_);
+};
+
+// Variables related to the audio data and formats.
+struct AudioFrameData {
+  explicit AudioFrameData(int max_frame_size) {
+    // Set up the two-dimensional arrays needed for the APM API calls.
+    input_framechannels.resize(2 * max_frame_size);
+    input_frame.resize(2);
+    input_frame[0] = &input_framechannels[0];
+    input_frame[1] = &input_framechannels[max_frame_size];
+
+    output_frame_channels.resize(2 * max_frame_size);
+    output_frame.resize(2);
+    output_frame[0] = &output_frame_channels[0];
+    output_frame[1] = &output_frame_channels[max_frame_size];
+  }
+
+  AudioFrame frame;
+  std::vector<float*> output_frame;
+  std::vector<float> output_frame_channels;
+  AudioProcessing::ChannelLayout output_channel_layout =
+      AudioProcessing::ChannelLayout::kMono;
+  int input_sample_rate_hz = 16000;
+  int input_number_of_channels = -1;
+  std::vector<float*> input_frame;
+  std::vector<float> input_framechannels;
+  AudioProcessing::ChannelLayout input_channel_layout =
+      AudioProcessing::ChannelLayout::kMono;
+  int output_sample_rate_hz = 16000;
+  int output_number_of_channels = -1;
+  StreamConfig input_stream_config;
+  StreamConfig output_stream_config;
+  int input_samples_per_channel = -1;
+  int output_samples_per_channel = -1;
+};
+
+// The configuration for the test.
+struct TestConfig {
+  // Test case generator for the test configurations to use in the brief tests.
+  static std::vector<TestConfig> GenerateBriefTestConfigs() {
+    std::vector<TestConfig> test_configs;
+    AecType aec_types[] = {AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec,
+                           AecType::BasicWebRtcAecSettingsWithAecMobile};
+    for (auto aec_type : aec_types) {
+      TestConfig test_config;
+      test_config.aec_type = aec_type;
+
+      test_config.min_number_of_calls = 300;
+
+      // Perform tests only with the extreme runtime parameter setting scheme.
+      test_config.runtime_parameter_setting_scheme =
+          RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme;
+
+      // Only test 16 kHz for this test suite.
+      test_config.initial_sample_rate_hz = 16000;
+
+      // Create test config for the second processing API function set.
+      test_config.render_api_function =
+          RenderApiImpl::ProcessReverseStreamImpl2;
+      test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl2;
+
+      // Create test config for the first processing API function set.
+      test_configs.push_back(test_config);
+      test_config.render_api_function =
+          RenderApiImpl::AnalyzeReverseStreamImpl2;
+      test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl3;
+      test_configs.push_back(test_config);
+    }
+
+    // Return the created test configurations.
+    return test_configs;
+  }
+
+  // Test case generator for the test configurations to use in the extensive
+  // tests.
+  static std::vector<TestConfig> GenerateExtensiveTestConfigs() {
+    // Lambda functions for the test config generation.
+    auto add_processing_apis = [](TestConfig test_config) {
+      struct AllowedApiCallCombinations {
+        RenderApiImpl render_api;
+        CaptureApiImpl capture_api;
+      };
+
+      const AllowedApiCallCombinations api_calls[] = {
+          {RenderApiImpl::ProcessReverseStreamImpl1,
+           CaptureApiImpl::ProcessStreamImpl1},
+          {RenderApiImpl::AnalyzeReverseStreamImpl1,
+           CaptureApiImpl::ProcessStreamImpl1},
+          {RenderApiImpl::ProcessReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl2},
+          {RenderApiImpl::ProcessReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl3},
+          {RenderApiImpl::AnalyzeReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl2},
+          {RenderApiImpl::AnalyzeReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl3}};
+      std::vector<TestConfig> out;
+      for (auto api_call : api_calls) {
+        test_config.render_api_function = api_call.render_api;
+        test_config.capture_api_function = api_call.capture_api;
+        out.push_back(test_config);
+      }
+      return out;
+    };
+
+    auto add_aec_settings = [](const std::vector<TestConfig>& in) {
+      std::vector<TestConfig> out;
+      AecType aec_types[] = {
+          AecType::BasicWebRtcAecSettings, AecType::AecTurnedOff,
+          AecType::BasicWebRtcAecSettingsWithExtentedFilter,
+          AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec,
+          AecType::BasicWebRtcAecSettingsWithAecMobile};
+      for (auto test_config : in) {
+        for (auto aec_type : aec_types) {
+          test_config.aec_type = aec_type;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    auto add_settings_scheme = [](const std::vector<TestConfig>& in) {
+      std::vector<TestConfig> out;
+      RuntimeParameterSettingScheme schemes[] = {
+          RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme,
+          RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme,
+          RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme,
+          RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme};
+
+      for (auto test_config : in) {
+        for (auto scheme : schemes) {
+          test_config.runtime_parameter_setting_scheme = scheme;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    auto add_sample_rates = [](const std::vector<TestConfig>& in) {
+      const int sample_rates[] = {8000, 16000, 32000, 48000};
+
+      std::vector<TestConfig> out;
+      for (auto test_config : in) {
+        auto available_rates =
+            (test_config.aec_type ==
+                     AecType::BasicWebRtcAecSettingsWithAecMobile
+                 ? rtc::ArrayView<const int>(sample_rates, 2)
+                 : rtc::ArrayView<const int>(sample_rates));
+
+        for (auto rate : available_rates) {
+          test_config.initial_sample_rate_hz = rate;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    // Generate test configurations of the relevant combinations of the
+    // parameters to
+    // test.
+    TestConfig test_config;
+    test_config.min_number_of_calls = 10000;
+    return add_sample_rates(add_settings_scheme(
+        add_aec_settings(add_processing_apis(test_config))));
+  }
+
+  RenderApiImpl render_api_function = RenderApiImpl::ProcessReverseStreamImpl2;
+  CaptureApiImpl capture_api_function = CaptureApiImpl::ProcessStreamImpl2;
+  RuntimeParameterSettingScheme runtime_parameter_setting_scheme =
+      RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme;
+  int initial_sample_rate_hz = 16000;
+  AecType aec_type = AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec;
+  int min_number_of_calls = 300;
+};
+
+// Handler for the frame counters.
+class FrameCounters {
+ public:
+  void IncreaseRenderCounter() {
+    rtc::CritScope cs(&crit_);
+    render_count++;
+  }
+
+  void IncreaseCaptureCounter() {
+    rtc::CritScope cs(&crit_);
+    capture_count++;
+  }
+
+  int GetCaptureCounter() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count;
+  }
+
+  int GetRenderCounter() const {
+    rtc::CritScope cs(&crit_);
+    return render_count;
+  }
+
+  int CaptureMinusRenderCounters() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count - render_count;
+  }
+
+  int RenderMinusCaptureCounters() const {
+    return -CaptureMinusRenderCounters();
+  }
+
+  bool BothCountersExceedeThreshold(int threshold) {
+    rtc::CritScope cs(&crit_);
+    return (render_count > threshold && capture_count > threshold);
+  }
+
+ private:
+  mutable rtc::CriticalSection crit_;
+  int render_count GUARDED_BY(crit_) = 0;
+  int capture_count GUARDED_BY(crit_) = 0;
+};
+
+// Class for handling the capture side processing.
+class CaptureProcessor {
+ public:
+  CaptureProcessor(int max_frame_size,
+                   RandomGenerator* rand_gen,
+                   rtc::Event* render_call_event,
+                   rtc::Event* capture_call_event,
+                   FrameCounters* shared_counters_state,
+                   AudioProcessingImplLockTest* test_framework,
+                   TestConfig* test_config,
+                   AudioProcessing* apm);
+  bool Process();
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const float kCaptureInputFloatLevel;
+  static const int kCaptureInputFixLevel = 1024;
+
+  void PrepareFrame();
+  void CallApmCaptureSide();
+  void ApplyRuntimeSettingScheme();
+
+  RandomGenerator* const rand_gen_ = nullptr;
+  rtc::Event* const render_call_event_ = nullptr;
+  rtc::Event* const capture_call_event_ = nullptr;
+  FrameCounters* const frame_counters_ = nullptr;
+  AudioProcessingImplLockTest* const test_ = nullptr;
+  const TestConfig* const test_config_ = nullptr;
+  AudioProcessing* const apm_ = nullptr;
+  AudioFrameData frame_data_;
+};
+
+// Class for handling the stats processing.
+class StatsProcessor {
+ public:
+  StatsProcessor(RandomGenerator* rand_gen,
+                 TestConfig* test_config,
+                 AudioProcessing* apm);
+  bool Process();
+
+ private:
+  RandomGenerator* rand_gen_ = nullptr;
+  TestConfig* test_config_ = nullptr;
+  AudioProcessing* apm_ = nullptr;
+};
+
+// Class for handling the render side processing.
+class RenderProcessor {
+ public:
+  RenderProcessor(int max_frame_size,
+                  RandomGenerator* rand_gen,
+                  rtc::Event* render_call_event,
+                  rtc::Event* capture_call_event,
+                  FrameCounters* shared_counters_state,
+                  AudioProcessingImplLockTest* test_framework,
+                  TestConfig* test_config,
+                  AudioProcessing* apm);
+  bool Process();
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const int kRenderInputFixLevel = 16384;
+  static const float kRenderInputFloatLevel;
+
+  void PrepareFrame();
+  void CallApmRenderSide();
+  void ApplyRuntimeSettingScheme();
+
+  RandomGenerator* const rand_gen_ = nullptr;
+  rtc::Event* const render_call_event_ = nullptr;
+  rtc::Event* const capture_call_event_ = nullptr;
+  FrameCounters* const frame_counters_ = nullptr;
+  AudioProcessingImplLockTest* const test_ = nullptr;
+  const TestConfig* const test_config_ = nullptr;
+  AudioProcessing* const apm_ = nullptr;
+  AudioFrameData frame_data_;
+  bool first_render_call_ = true;
+};
+
+class AudioProcessingImplLockTest
+    : public ::testing::TestWithParam<TestConfig> {
+ public:
+  AudioProcessingImplLockTest();
+  bool RunTest();
+  bool MaybeEndTest();
+
+ private:
+  static const int kTestTimeOutLimit = 10 * 60 * 1000;
+  static const int kMaxFrameSize = 480;
+
+  // ::testing::TestWithParam<> implementation
+  void SetUp() override;
+  void TearDown() override;
+
+  // Thread callback for the render thread
+  static bool RenderProcessorThreadFunc(void* context) {
+    return reinterpret_cast<AudioProcessingImplLockTest*>(context)
+        ->render_thread_state_.Process();
+  }
+
+  // Thread callback for the capture thread
+  static bool CaptureProcessorThreadFunc(void* context) {
+    return reinterpret_cast<AudioProcessingImplLockTest*>(context)
+        ->capture_thread_state_.Process();
+  }
+
+  // Thread callback for the stats thread
+  static bool StatsProcessorThreadFunc(void* context) {
+    return reinterpret_cast<AudioProcessingImplLockTest*>(context)
+        ->stats_thread_state_.Process();
+  }
+
+  // Tests whether all the required render and capture side calls have been
+  // done.
+  bool TestDone() {
+    return frame_counters_.BothCountersExceedeThreshold(
+        test_config_.min_number_of_calls);
+  }
+
+  // Start the threads used in the test.
+  void StartThreads() {
+    render_thread_.Start();
+    render_thread_.SetPriority(rtc::kRealtimePriority);
+    capture_thread_.Start();
+    capture_thread_.SetPriority(rtc::kRealtimePriority);
+    stats_thread_.Start();
+    stats_thread_.SetPriority(rtc::kNormalPriority);
+  }
+
+  // Event handlers for the test.
+  rtc::Event test_complete_;
+  rtc::Event render_call_event_;
+  rtc::Event capture_call_event_;
+
+  // Thread related variables.
+  rtc::PlatformThread render_thread_;
+  rtc::PlatformThread capture_thread_;
+  rtc::PlatformThread stats_thread_;
+  mutable RandomGenerator rand_gen_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+  TestConfig test_config_;
+  FrameCounters frame_counters_;
+  RenderProcessor render_thread_state_;
+  CaptureProcessor capture_thread_state_;
+  StatsProcessor stats_thread_state_;
+};
+
+// Sleeps a random time between 0 and max_sleep milliseconds.
+void SleepRandomMs(int max_sleep, RandomGenerator* rand_gen) {
+  int sleeptime = rand_gen->RandInt(0, max_sleep);
+  SleepMs(sleeptime);
+}
+
+// Populates a float audio frame with random data.
+void PopulateAudioFrame(float** frame,
+                        float amplitude,
+                        size_t num_channels,
+                        size_t samples_per_channel,
+                        RandomGenerator* rand_gen) {
+  for (size_t ch = 0; ch < num_channels; ch++) {
+    for (size_t k = 0; k < samples_per_channel; k++) {
+      // Store random 16 bit quantized float number between +-amplitude.
+      frame[ch][k] = amplitude * (2 * rand_gen->RandFloat() - 1);
+    }
+  }
+}
+
+// Populates an audioframe frame of AudioFrame type with random data.
+void PopulateAudioFrame(AudioFrame* frame,
+                        int16_t amplitude,
+                        RandomGenerator* rand_gen) {
+  ASSERT_GT(amplitude, 0);
+  ASSERT_LE(amplitude, 32767);
+  for (size_t ch = 0; ch < frame->num_channels_; ch++) {
+    for (size_t k = 0; k < frame->samples_per_channel_; k++) {
+      // Store random 16 bit number between -(amplitude+1) and
+      // amplitude.
+      frame->data_[k * ch] =
+          rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
+    }
+  }
+}
+
+AudioProcessingImplLockTest::AudioProcessingImplLockTest()
+    : test_complete_(false, false),
+      render_call_event_(false, false),
+      capture_call_event_(false, false),
+      render_thread_(RenderProcessorThreadFunc, this, "render"),
+      capture_thread_(CaptureProcessorThreadFunc, this, "capture"),
+      stats_thread_(StatsProcessorThreadFunc, this, "stats"),
+      apm_(AudioProcessingImpl::Create()),
+      render_thread_state_(kMaxFrameSize,
+                           &rand_gen_,
+                           &render_call_event_,
+                           &capture_call_event_,
+                           &frame_counters_,
+                           this,
+                           &test_config_,
+                           apm_.get()),
+      capture_thread_state_(kMaxFrameSize,
+                            &rand_gen_,
+                            &render_call_event_,
+                            &capture_call_event_,
+                            &frame_counters_,
+                            this,
+                            &test_config_,
+                            apm_.get()),
+      stats_thread_state_(&rand_gen_, &test_config_, apm_.get()) {}
+
+// Run the test with a timeout.
+bool AudioProcessingImplLockTest::RunTest() {
+  StartThreads();
+  return test_complete_.Wait(kTestTimeOutLimit);
+}
+
+bool AudioProcessingImplLockTest::MaybeEndTest() {
+  if (HasFatalFailure() || TestDone()) {
+    test_complete_.Set();
+    return true;
+  }
+  return false;
+}
+
+// Setup of test and APM.
+void AudioProcessingImplLockTest::SetUp() {
+  test_config_ = static_cast<TestConfig>(GetParam());
+
+  ASSERT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+  ASSERT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+
+  ASSERT_EQ(apm_->kNoError,
+            apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+  ASSERT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+
+  ASSERT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
+  ASSERT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+
+  Config config;
+  if (test_config_.aec_type == AecType::AecTurnedOff) {
+    ASSERT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
+  } else if (test_config_.aec_type ==
+             AecType::BasicWebRtcAecSettingsWithAecMobile) {
+    ASSERT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
+  } else {
+    ASSERT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->enable_metrics(true));
+    ASSERT_EQ(apm_->kNoError,
+              apm_->echo_cancellation()->enable_delay_logging(true));
+
+    config.Set<ExtendedFilter>(
+        new ExtendedFilter(test_config_.aec_type ==
+                           AecType::BasicWebRtcAecSettingsWithExtentedFilter));
+
+    config.Set<DelayAgnostic>(
+        new DelayAgnostic(test_config_.aec_type ==
+                          AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec));
+
+    apm_->SetExtraOptions(config);
+  }
+}
+
+void AudioProcessingImplLockTest::TearDown() {
+  render_call_event_.Set();
+  capture_call_event_.Set();
+  render_thread_.Stop();
+  capture_thread_.Stop();
+  stats_thread_.Stop();
+}
+
+StatsProcessor::StatsProcessor(RandomGenerator* rand_gen,
+                               TestConfig* test_config,
+                               AudioProcessing* apm)
+    : rand_gen_(rand_gen), test_config_(test_config), apm_(apm) {}
+
+// Implements the callback functionality for the statistics
+// collection thread.
+bool StatsProcessor::Process() {
+  SleepRandomMs(100, rand_gen_);
+
+  EXPECT_EQ(apm_->echo_cancellation()->is_enabled(),
+            ((test_config_->aec_type != AecType::AecTurnedOff) &&
+             (test_config_->aec_type !=
+              AecType::BasicWebRtcAecSettingsWithAecMobile)));
+  apm_->echo_cancellation()->stream_drift_samples();
+  EXPECT_EQ(apm_->echo_control_mobile()->is_enabled(),
+            (test_config_->aec_type != AecType::AecTurnedOff) &&
+                (test_config_->aec_type ==
+                 AecType::BasicWebRtcAecSettingsWithAecMobile));
+  EXPECT_TRUE(apm_->gain_control()->is_enabled());
+  apm_->gain_control()->stream_analog_level();
+  EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
+
+  // The below return values are not testable.
+  apm_->noise_suppression()->speech_probability();
+  apm_->voice_detection()->is_enabled();
+
+  return true;
+}
+
+const float CaptureProcessor::kCaptureInputFloatLevel = 0.03125f;
+
+CaptureProcessor::CaptureProcessor(int max_frame_size,
+                                   RandomGenerator* rand_gen,
+                                   rtc::Event* render_call_event,
+                                   rtc::Event* capture_call_event,
+                                   FrameCounters* shared_counters_state,
+                                   AudioProcessingImplLockTest* test_framework,
+                                   TestConfig* test_config,
+                                   AudioProcessing* apm)
+    : rand_gen_(rand_gen),
+      render_call_event_(render_call_event),
+      capture_call_event_(capture_call_event),
+      frame_counters_(shared_counters_state),
+      test_(test_framework),
+      test_config_(test_config),
+      apm_(apm),
+      frame_data_(max_frame_size) {}
+
+// Implements the callback functionality for the capture thread.
+bool CaptureProcessor::Process() {
+  // Sleep a random time to simulate thread jitter.
+  SleepRandomMs(3, rand_gen_);
+
+  // Check whether the test is done.
+  if (test_->MaybeEndTest()) {
+    return false;
+  }
+
+  // Ensure that the number of render and capture calls do not
+  // differ too much.
+  if (frame_counters_->CaptureMinusRenderCounters() > kMaxCallDifference) {
+    render_call_event_->Wait(rtc::Event::kForever);
+  }
+
+  // Apply any specified capture side APM non-processing runtime calls.
+  ApplyRuntimeSettingScheme();
+
+  // Apply the capture side processing call.
+  CallApmCaptureSide();
+
+  // Increase the number of capture-side calls.
+  frame_counters_->IncreaseCaptureCounter();
+
+  // Flag to the render thread that another capture API call has occurred
+  // by triggering this threads call event.
+  capture_call_event_->Set();
+
+  return true;
+}
+
+// Prepares a frame with relevant audio data and metadata.
+void CaptureProcessor::PrepareFrame() {
+  // Restrict to a common fixed sample rate if the AudioFrame
+  // interface is used.
+  if (test_config_->capture_api_function ==
+      CaptureApiImpl::ProcessStreamImpl1) {
+    frame_data_.input_sample_rate_hz = test_config_->initial_sample_rate_hz;
+    frame_data_.output_sample_rate_hz = test_config_->initial_sample_rate_hz;
+  }
+
+  // Prepare the audioframe data and metadata.
+  frame_data_.input_samples_per_channel =
+      frame_data_.input_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.frame.sample_rate_hz_ = frame_data_.input_sample_rate_hz;
+  frame_data_.frame.num_channels_ = frame_data_.input_number_of_channels;
+  frame_data_.frame.samples_per_channel_ =
+      frame_data_.input_samples_per_channel;
+  PopulateAudioFrame(&frame_data_.frame, kCaptureInputFixLevel, rand_gen_);
+
+  // Prepare the float audio input data and metadata.
+  frame_data_.input_stream_config.set_sample_rate_hz(
+      frame_data_.input_sample_rate_hz);
+  frame_data_.input_stream_config.set_num_channels(
+      frame_data_.input_number_of_channels);
+  frame_data_.input_stream_config.set_has_keyboard(false);
+  PopulateAudioFrame(&frame_data_.input_frame[0], kCaptureInputFloatLevel,
+                     frame_data_.input_number_of_channels,
+                     frame_data_.input_samples_per_channel, rand_gen_);
+  frame_data_.input_channel_layout =
+      (frame_data_.input_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+
+  // Prepare the float audio output data and metadata.
+  frame_data_.output_samples_per_channel =
+      frame_data_.output_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.output_stream_config.set_sample_rate_hz(
+      frame_data_.output_sample_rate_hz);
+  frame_data_.output_stream_config.set_num_channels(
+      frame_data_.output_number_of_channels);
+  frame_data_.output_stream_config.set_has_keyboard(false);
+  frame_data_.output_channel_layout =
+      (frame_data_.output_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+}
+
+// Applies the capture side processing API call.
+void CaptureProcessor::CallApmCaptureSide() {
+  // Prepare a proper capture side processing API call input.
+  PrepareFrame();
+
+  // Set the stream delay
+  apm_->set_stream_delay_ms(30);
+
+  // Call the specified capture side API processing method.
+  int result = AudioProcessing::kNoError;
+  switch (test_config_->capture_api_function) {
+    case CaptureApiImpl::ProcessStreamImpl1:
+      result = apm_->ProcessStream(&frame_data_.frame);
+      break;
+    case CaptureApiImpl::ProcessStreamImpl2:
+      result = apm_->ProcessStream(
+          &frame_data_.input_frame[0], frame_data_.input_samples_per_channel,
+          frame_data_.input_sample_rate_hz, frame_data_.input_channel_layout,
+          frame_data_.output_sample_rate_hz, frame_data_.output_channel_layout,
+          &frame_data_.output_frame[0]);
+      break;
+    case CaptureApiImpl::ProcessStreamImpl3:
+      result = apm_->ProcessStream(
+          &frame_data_.input_frame[0], frame_data_.input_stream_config,
+          frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+      break;
+    default:
+      FAIL();
+  }
+
+  // Check the return code for error.
+  ASSERT_EQ(AudioProcessing::kNoError, result);
+}
+
+// Applies any runtime capture APM API calls and audio stream characteristics
+// specified by the scheme for the test.
+void CaptureProcessor::ApplyRuntimeSettingScheme() {
+  const int capture_count_local = frame_counters_->GetCaptureCounter();
+
+  // Update the number of channels and sample rates for the input and output.
+  // Note that the counts frequencies for when to set parameters
+  // are set using prime numbers in order to ensure that the
+  // permutation scheme in the parameter setting changes.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+      if (capture_count_local == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (capture_count_local % 11 == 0)
+        frame_data_.input_sample_rate_hz = 32000;
+      else if (capture_count_local % 73 == 0)
+        frame_data_.input_sample_rate_hz = 48000;
+      else if (capture_count_local % 89 == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (capture_count_local % 97 == 0)
+        frame_data_.input_sample_rate_hz = 8000;
+
+      if (capture_count_local == 0)
+        frame_data_.input_number_of_channels = 1;
+      else if (capture_count_local % 4 == 0)
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+
+      if (capture_count_local == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (capture_count_local % 5 == 0)
+        frame_data_.output_sample_rate_hz = 32000;
+      else if (capture_count_local % 47 == 0)
+        frame_data_.output_sample_rate_hz = 48000;
+      else if (capture_count_local % 53 == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (capture_count_local % 71 == 0)
+        frame_data_.output_sample_rate_hz = 8000;
+
+      if (capture_count_local == 0)
+        frame_data_.output_number_of_channels = 1;
+      else if (capture_count_local % 8 == 0)
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+      if (capture_count_local % 2 == 0) {
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+      } else {
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.input_sample_rate_hz == 8000)
+          frame_data_.input_sample_rate_hz = 16000;
+        else if (frame_data_.input_sample_rate_hz == 16000)
+          frame_data_.input_sample_rate_hz = 32000;
+        else if (frame_data_.input_sample_rate_hz == 32000)
+          frame_data_.input_sample_rate_hz = 48000;
+        else if (frame_data_.input_sample_rate_hz == 48000)
+          frame_data_.input_sample_rate_hz = 8000;
+
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.output_sample_rate_hz == 8000)
+          frame_data_.output_sample_rate_hz = 16000;
+        else if (frame_data_.output_sample_rate_hz == 16000)
+          frame_data_.output_sample_rate_hz = 32000;
+        else if (frame_data_.output_sample_rate_hz == 32000)
+          frame_data_.output_sample_rate_hz = 48000;
+        else if (frame_data_.output_sample_rate_hz == 48000)
+          frame_data_.output_sample_rate_hz = 8000;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      if (capture_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (capture_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 2;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 2;
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Call any specified runtime APM setter and
+  // getter calls.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (capture_count_local % 2 == 0) {
+        ASSERT_EQ(AudioProcessing::Error::kNoError,
+                  apm_->set_stream_delay_ms(30));
+        apm_->set_stream_key_pressed(true);
+        apm_->set_delay_offset_ms(15);
+        EXPECT_EQ(apm_->delay_offset_ms(), 15);
+      } else {
+        ASSERT_EQ(AudioProcessing::Error::kNoError,
+                  apm_->set_stream_delay_ms(50));
+        apm_->set_stream_key_pressed(false);
+        apm_->set_delay_offset_ms(20);
+        EXPECT_EQ(apm_->delay_offset_ms(), 20);
+        apm_->delay_offset_ms();
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Restric the number of output channels not to exceed
+  // the number of input channels.
+  frame_data_.output_number_of_channels =
+      std::min(frame_data_.output_number_of_channels,
+               frame_data_.input_number_of_channels);
+}
+
+const float RenderProcessor::kRenderInputFloatLevel = 0.5f;
+
+RenderProcessor::RenderProcessor(int max_frame_size,
+                                 RandomGenerator* rand_gen,
+                                 rtc::Event* render_call_event,
+                                 rtc::Event* capture_call_event,
+                                 FrameCounters* shared_counters_state,
+                                 AudioProcessingImplLockTest* test_framework,
+                                 TestConfig* test_config,
+                                 AudioProcessing* apm)
+    : rand_gen_(rand_gen),
+      render_call_event_(render_call_event),
+      capture_call_event_(capture_call_event),
+      frame_counters_(shared_counters_state),
+      test_(test_framework),
+      test_config_(test_config),
+      apm_(apm),
+      frame_data_(max_frame_size) {}
+
+// Implements the callback functionality for the render thread.
+bool RenderProcessor::Process() {
+  // Conditional wait to ensure that a capture call has been done
+  // before the first render call is performed (implicitly
+  // required by the APM API).
+  if (first_render_call_) {
+    capture_call_event_->Wait(rtc::Event::kForever);
+    first_render_call_ = false;
+  }
+
+  // Sleep a random time to simulate thread jitter.
+  SleepRandomMs(3, rand_gen_);
+
+  // Check whether the test is done.
+  if (test_->MaybeEndTest()) {
+    return false;
+  }
+
+  // Ensure that the number of render and capture calls do not
+  // differ too much.
+  if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
+    capture_call_event_->Wait(rtc::Event::kForever);
+  }
+
+  // Apply any specified render side APM non-processing runtime calls.
+  ApplyRuntimeSettingScheme();
+
+  // Apply the render side processing call.
+  CallApmRenderSide();
+
+  // Increase the number of render-side calls.
+  frame_counters_->IncreaseRenderCounter();
+
+  // Flag to the capture thread that another render API call has occurred
+  // by triggering this threads call event.
+  render_call_event_->Set();
+  return true;
+}
+
+// Prepares the render side frame and the accompanying metadata
+// with the appropriate information.
+void RenderProcessor::PrepareFrame() {
+  // Restrict to a common fixed sample rate if the AudioFrame interface is
+  // used.
+  if ((test_config_->render_api_function ==
+       RenderApiImpl::AnalyzeReverseStreamImpl1) ||
+      (test_config_->render_api_function ==
+       RenderApiImpl::ProcessReverseStreamImpl1) ||
+      (test_config_->aec_type !=
+       AecType::BasicWebRtcAecSettingsWithAecMobile)) {
+    frame_data_.input_sample_rate_hz = test_config_->initial_sample_rate_hz;
+    frame_data_.output_sample_rate_hz = test_config_->initial_sample_rate_hz;
+  }
+
+  // Prepare the audioframe data and metadata
+  frame_data_.input_samples_per_channel =
+      frame_data_.input_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.frame.sample_rate_hz_ = frame_data_.input_sample_rate_hz;
+  frame_data_.frame.num_channels_ = frame_data_.input_number_of_channels;
+  frame_data_.frame.samples_per_channel_ =
+      frame_data_.input_samples_per_channel;
+  PopulateAudioFrame(&frame_data_.frame, kRenderInputFixLevel, rand_gen_);
+
+  // Prepare the float audio input data and metadata.
+  frame_data_.input_stream_config.set_sample_rate_hz(
+      frame_data_.input_sample_rate_hz);
+  frame_data_.input_stream_config.set_num_channels(
+      frame_data_.input_number_of_channels);
+  frame_data_.input_stream_config.set_has_keyboard(false);
+  PopulateAudioFrame(&frame_data_.input_frame[0], kRenderInputFloatLevel,
+                     frame_data_.input_number_of_channels,
+                     frame_data_.input_samples_per_channel, rand_gen_);
+  frame_data_.input_channel_layout =
+      (frame_data_.input_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+
+  // Prepare the float audio output data and metadata.
+  frame_data_.output_samples_per_channel =
+      frame_data_.output_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.output_stream_config.set_sample_rate_hz(
+      frame_data_.output_sample_rate_hz);
+  frame_data_.output_stream_config.set_num_channels(
+      frame_data_.output_number_of_channels);
+  frame_data_.output_stream_config.set_has_keyboard(false);
+  frame_data_.output_channel_layout =
+      (frame_data_.output_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+}
+
+// Makes the render side processing API call.
+void RenderProcessor::CallApmRenderSide() {
+  // Prepare a proper render side processing API call input.
+  PrepareFrame();
+
+  // Call the specified render side API processing method.
+  int result = AudioProcessing::kNoError;
+  switch (test_config_->render_api_function) {
+    case RenderApiImpl::ProcessReverseStreamImpl1:
+      result = apm_->ProcessReverseStream(&frame_data_.frame);
+      break;
+    case RenderApiImpl::ProcessReverseStreamImpl2:
+      result = apm_->ProcessReverseStream(
+          &frame_data_.input_frame[0], frame_data_.input_stream_config,
+          frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+      break;
+    case RenderApiImpl::AnalyzeReverseStreamImpl1:
+      result = apm_->AnalyzeReverseStream(&frame_data_.frame);
+      break;
+    case RenderApiImpl::AnalyzeReverseStreamImpl2:
+      result = apm_->AnalyzeReverseStream(
+          &frame_data_.input_frame[0], frame_data_.input_samples_per_channel,
+          frame_data_.input_sample_rate_hz, frame_data_.input_channel_layout);
+      break;
+    default:
+      FAIL();
+  }
+
+  // Check the return code for error.
+  ASSERT_EQ(AudioProcessing::kNoError, result);
+}
+
+// Applies any render capture side APM API calls and audio stream
+// characteristics
+// specified by the scheme for the test.
+void RenderProcessor::ApplyRuntimeSettingScheme() {
+  const int render_count_local = frame_counters_->GetRenderCounter();
+
+  // Update the number of channels and sample rates for the input and output.
+  // Note that the counts frequencies for when to set parameters
+  // are set using prime numbers in order to ensure that the
+  // permutation scheme in the parameter setting changes.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+      if (render_count_local == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (render_count_local % 47 == 0)
+        frame_data_.input_sample_rate_hz = 32000;
+      else if (render_count_local % 71 == 0)
+        frame_data_.input_sample_rate_hz = 48000;
+      else if (render_count_local % 79 == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (render_count_local % 83 == 0)
+        frame_data_.input_sample_rate_hz = 8000;
+
+      if (render_count_local == 0)
+        frame_data_.input_number_of_channels = 1;
+      else if (render_count_local % 4 == 0)
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+
+      if (render_count_local == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (render_count_local % 17 == 0)
+        frame_data_.output_sample_rate_hz = 32000;
+      else if (render_count_local % 19 == 0)
+        frame_data_.output_sample_rate_hz = 48000;
+      else if (render_count_local % 29 == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (render_count_local % 61 == 0)
+        frame_data_.output_sample_rate_hz = 8000;
+
+      if (render_count_local == 0)
+        frame_data_.output_number_of_channels = 1;
+      else if (render_count_local % 8 == 0)
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+      } else {
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.input_sample_rate_hz == 8000)
+          frame_data_.input_sample_rate_hz = 16000;
+        else if (frame_data_.input_sample_rate_hz == 16000)
+          frame_data_.input_sample_rate_hz = 32000;
+        else if (frame_data_.input_sample_rate_hz == 32000)
+          frame_data_.input_sample_rate_hz = 48000;
+        else if (frame_data_.input_sample_rate_hz == 48000)
+          frame_data_.input_sample_rate_hz = 8000;
+
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.output_sample_rate_hz == 8000)
+          frame_data_.output_sample_rate_hz = 16000;
+        else if (frame_data_.output_sample_rate_hz == 16000)
+          frame_data_.output_sample_rate_hz = 32000;
+        else if (frame_data_.output_sample_rate_hz == 32000)
+          frame_data_.output_sample_rate_hz = 48000;
+        else if (frame_data_.output_sample_rate_hz == 48000)
+          frame_data_.output_sample_rate_hz = 8000;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 2;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 2;
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Restric the number of output channels not to exceed
+  // the number of input channels.
+  frame_data_.output_number_of_channels =
+      std::min(frame_data_.output_number_of_channels,
+               frame_data_.input_number_of_channels);
+}
+
+}  // anonymous namespace
+
+TEST_P(AudioProcessingImplLockTest, LockTest) {
+  // Run test and verify that it did not time out.
+  ASSERT_TRUE(RunTest());
+}
+
+// Instantiate tests from the extreme test configuration set.
+INSTANTIATE_TEST_CASE_P(
+    DISABLED_AudioProcessingImplLockExtensive,
+    AudioProcessingImplLockTest,
+    ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs()));
+
+INSTANTIATE_TEST_CASE_P(
+    AudioProcessingImplLockBrief,
+    AudioProcessingImplLockTest,
+    ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs()));
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
index f4c36d0009..ed20daaa61 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -14,7 +14,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 using ::testing::Invoke;
 using ::testing::Return;
diff --git a/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
new file mode 100644
index 0000000000..0c8c060ea3
--- /dev/null
+++ b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -0,0 +1,724 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/random.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/testsupport/perf_test.h"
+
+namespace webrtc {
+
+namespace {
+
+static const bool kPrintAllDurations = false;
+
+class CallSimulator;
+
+// Type of the render thread APM API call to use in the test.
+enum class ProcessorType { kRender, kCapture };
+
+// Variant of APM processing settings to use in the test.
+enum class SettingsType {
+  kDefaultApmDesktop,
+  kDefaultApmMobile,
+  kDefaultApmDesktopAndBeamformer,
+  kDefaultApmDesktopAndIntelligibilityEnhancer,
+  kAllSubmodulesTurnedOff,
+  kDefaultDesktopApmWithoutDelayAgnostic,
+  kDefaultDesktopApmWithoutExtendedFilter
+};
+
+// Variables related to the audio data and formats.
+struct AudioFrameData {
+  explicit AudioFrameData(size_t max_frame_size) {
+    // Set up the two-dimensional arrays needed for the APM API calls.
+    input_framechannels.resize(2 * max_frame_size);
+    input_frame.resize(2);
+    input_frame[0] = &input_framechannels[0];
+    input_frame[1] = &input_framechannels[max_frame_size];
+
+    output_frame_channels.resize(2 * max_frame_size);
+    output_frame.resize(2);
+    output_frame[0] = &output_frame_channels[0];
+    output_frame[1] = &output_frame_channels[max_frame_size];
+  }
+
+  std::vector<float> output_frame_channels;
+  std::vector<float*> output_frame;
+  std::vector<float> input_framechannels;
+  std::vector<float*> input_frame;
+  StreamConfig input_stream_config;
+  StreamConfig output_stream_config;
+};
+
+// The configuration for the test.
+struct SimulationConfig {
+  SimulationConfig(int sample_rate_hz, SettingsType simulation_settings)
+      : sample_rate_hz(sample_rate_hz),
+        simulation_settings(simulation_settings) {}
+
+  static std::vector<SimulationConfig> GenerateSimulationConfigs() {
+    std::vector<SimulationConfig> simulation_configs;
+#ifndef WEBRTC_ANDROID
+    const SettingsType desktop_settings[] = {
+        SettingsType::kDefaultApmDesktop, SettingsType::kAllSubmodulesTurnedOff,
+        SettingsType::kDefaultDesktopApmWithoutDelayAgnostic,
+        SettingsType::kDefaultDesktopApmWithoutExtendedFilter};
+
+    const int desktop_sample_rates[] = {8000, 16000, 32000, 48000};
+
+    for (auto sample_rate : desktop_sample_rates) {
+      for (auto settings : desktop_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    const SettingsType intelligibility_enhancer_settings[] = {
+        SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer};
+
+    const int intelligibility_enhancer_sample_rates[] = {8000, 16000, 32000,
+                                                         48000};
+
+    for (auto sample_rate : intelligibility_enhancer_sample_rates) {
+      for (auto settings : intelligibility_enhancer_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    const SettingsType beamformer_settings[] = {
+        SettingsType::kDefaultApmDesktopAndBeamformer};
+
+    const int beamformer_sample_rates[] = {8000, 16000, 32000, 48000};
+
+    for (auto sample_rate : beamformer_sample_rates) {
+      for (auto settings : beamformer_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+#endif
+
+    const SettingsType mobile_settings[] = {SettingsType::kDefaultApmMobile};
+
+    const int mobile_sample_rates[] = {8000, 16000};
+
+    for (auto sample_rate : mobile_sample_rates) {
+      for (auto settings : mobile_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    return simulation_configs;
+  }
+
+  std::string SettingsDescription() const {
+    std::string description;
+    switch (simulation_settings) {
+      case SettingsType::kDefaultApmMobile:
+        description = "DefaultApmMobile";
+        break;
+      case SettingsType::kDefaultApmDesktop:
+        description = "DefaultApmDesktop";
+        break;
+      case SettingsType::kDefaultApmDesktopAndBeamformer:
+        description = "DefaultApmDesktopAndBeamformer";
+        break;
+      case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer:
+        description = "DefaultApmDesktopAndIntelligibilityEnhancer";
+        break;
+      case SettingsType::kAllSubmodulesTurnedOff:
+        description = "AllSubmodulesOff";
+        break;
+      case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic:
+        description = "DefaultDesktopApmWithoutDelayAgnostic";
+        break;
+      case SettingsType::kDefaultDesktopApmWithoutExtendedFilter:
+        description = "DefaultDesktopApmWithoutExtendedFilter";
+        break;
+    }
+    return description;
+  }
+
+  int sample_rate_hz = 16000;
+  SettingsType simulation_settings = SettingsType::kDefaultApmDesktop;
+};
+
+// Handler for the frame counters.
+class FrameCounters {
+ public:
+  void IncreaseRenderCounter() {
+    rtc::CritScope cs(&crit_);
+    render_count_++;
+  }
+
+  void IncreaseCaptureCounter() {
+    rtc::CritScope cs(&crit_);
+    capture_count_++;
+  }
+
+  int GetCaptureCounter() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count_;
+  }
+
+  int GetRenderCounter() const {
+    rtc::CritScope cs(&crit_);
+    return render_count_;
+  }
+
+  int CaptureMinusRenderCounters() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count_ - render_count_;
+  }
+
+  int RenderMinusCaptureCounters() const {
+    return -CaptureMinusRenderCounters();
+  }
+
+  bool BothCountersExceedeThreshold(int threshold) const {
+    rtc::CritScope cs(&crit_);
+    return (render_count_ > threshold && capture_count_ > threshold);
+  }
+
+ private:
+  mutable rtc::CriticalSection crit_;
+  int render_count_ GUARDED_BY(crit_) = 0;
+  int capture_count_ GUARDED_BY(crit_) = 0;
+};
+
+// Class that protects a flag using a lock.
+class LockedFlag {
+ public:
+  bool get_flag() const {
+    rtc::CritScope cs(&crit_);
+    return flag_;
+  }
+
+  void set_flag() {
+    rtc::CritScope cs(&crit_);
+    flag_ = true;
+  }
+
+ private:
+  mutable rtc::CriticalSection crit_;
+  bool flag_ GUARDED_BY(crit_) = false;
+};
+
+// Parent class for the thread processors.
+class TimedThreadApiProcessor {
+ public:
+  TimedThreadApiProcessor(ProcessorType processor_type,
+                          Random* rand_gen,
+                          FrameCounters* shared_counters_state,
+                          LockedFlag* capture_call_checker,
+                          CallSimulator* test_framework,
+                          const SimulationConfig* simulation_config,
+                          AudioProcessing* apm,
+                          int num_durations_to_store,
+                          float input_level,
+                          int num_channels)
+      : rand_gen_(rand_gen),
+        frame_counters_(shared_counters_state),
+        capture_call_checker_(capture_call_checker),
+        test_(test_framework),
+        simulation_config_(simulation_config),
+        apm_(apm),
+        frame_data_(kMaxFrameSize),
+        clock_(webrtc::Clock::GetRealTimeClock()),
+        num_durations_to_store_(num_durations_to_store),
+        input_level_(input_level),
+        processor_type_(processor_type),
+        num_channels_(num_channels) {
+    api_call_durations_.reserve(num_durations_to_store_);
+  }
+
+  // Implements the callback functionality for the threads.
+  bool Process();
+
+  // Method for printing out the simulation statistics.
+  void print_processor_statistics(std::string processor_name) const {
+    const std::string modifier = "_api_call_duration";
+
+    // Lambda function for creating a test printout string.
+    auto create_mean_and_std_string = [](int64_t average,
+                                         int64_t standard_dev) {
+      std::string s = std::to_string(average);
+      s += ", ";
+      s += std::to_string(standard_dev);
+      return s;
+    };
+
+    const std::string sample_rate_name =
+        "_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
+
+    webrtc::test::PrintResultMeanAndError(
+        "apm_timing", sample_rate_name, processor_name,
+        create_mean_and_std_string(GetDurationAverage(),
+                                   GetDurationStandardDeviation()),
+        "us", false);
+
+    if (kPrintAllDurations) {
+      std::string value_string = "";
+      for (int64_t duration : api_call_durations_) {
+        value_string += std::to_string(duration) + ",";
+      }
+      webrtc::test::PrintResultList("apm_call_durations", sample_rate_name,
+                                    processor_name, value_string, "us", false);
+    }
+  }
+
+  void AddDuration(int64_t duration) {
+    if (api_call_durations_.size() < num_durations_to_store_) {
+      api_call_durations_.push_back(duration);
+    }
+  }
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const int kMaxFrameSize = 480;
+  static const int kNumInitializationFrames = 5;
+
+  int64_t GetDurationStandardDeviation() const {
+    double variance = 0;
+    const int64_t average_duration = GetDurationAverage();
+    for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
+         k++) {
+      int64_t tmp = api_call_durations_[k] - average_duration;
+      variance += static_cast<double>(tmp * tmp);
+    }
+    const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
+                            kNumInitializationFrames;
+    return (denominator > 0
+                ? rtc::checked_cast<int64_t>(sqrt(variance / denominator))
+                : -1);
+  }
+
+  int64_t GetDurationAverage() const {
+    int64_t average_duration = 0;
+    for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
+         k++) {
+      average_duration += api_call_durations_[k];
+    }
+    const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
+                            kNumInitializationFrames;
+    return (denominator > 0 ? average_duration / denominator : -1);
+  }
+
+  int ProcessCapture() {
+    // Set the stream delay.
+    apm_->set_stream_delay_ms(30);
+
+    // Call and time the specified capture side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+
+    frame_counters_->IncreaseCaptureCounter();
+
+    AddDuration(end_time - start_time);
+
+    if (first_process_call_) {
+      // Flag that the capture side has been called at least once
+      // (needed to ensure that a capture call has been done
+      // before the first render call is performed (implicitly
+      // required by the APM API).
+      capture_call_checker_->set_flag();
+      first_process_call_ = false;
+    }
+    return result;
+  }
+
+  bool ReadyToProcessCapture() {
+    return (frame_counters_->CaptureMinusRenderCounters() <=
+            kMaxCallDifference);
+  }
+
+  int ProcessRender() {
+    // Call and time the specified render side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessReverseStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+    frame_counters_->IncreaseRenderCounter();
+
+    AddDuration(end_time - start_time);
+
+    return result;
+  }
+
+  bool ReadyToProcessRender() {
+    // Do not process until at least one capture call has been done.
+    // (implicitly required by the APM API).
+    if (first_process_call_ && !capture_call_checker_->get_flag()) {
+      return false;
+    }
+
+    // Ensure that the number of render and capture calls do not differ too
+    // much.
+    if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
+      return false;
+    }
+
+    first_process_call_ = false;
+    return true;
+  }
+
+  void PrepareFrame() {
+    // Lambda function for populating a float multichannel audio frame
+    // with random data.
+    auto populate_audio_frame = [](float amplitude, size_t num_channels,
+                                   size_t samples_per_channel, Random* rand_gen,
+                                   float** frame) {
+      for (size_t ch = 0; ch < num_channels; ch++) {
+        for (size_t k = 0; k < samples_per_channel; k++) {
+          // Store random float number with a value between +-amplitude.
+          frame[ch][k] = amplitude * (2 * rand_gen->Rand<float>() - 1);
+        }
+      }
+    };
+
+    // Prepare the audio input data and metadata.
+    frame_data_.input_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.input_stream_config.set_num_channels(num_channels_);
+    frame_data_.input_stream_config.set_has_keyboard(false);
+    populate_audio_frame(input_level_, num_channels_,
+                         (simulation_config_->sample_rate_hz *
+                          AudioProcessing::kChunkSizeMs / 1000),
+                         rand_gen_, &frame_data_.input_frame[0]);
+
+    // Prepare the float audio output data and metadata.
+    frame_data_.output_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.output_stream_config.set_num_channels(1);
+    frame_data_.output_stream_config.set_has_keyboard(false);
+  }
+
+  bool ReadyToProcess() {
+    switch (processor_type_) {
+      case ProcessorType::kRender:
+        return ReadyToProcessRender();
+        break;
+      case ProcessorType::kCapture:
+        return ReadyToProcessCapture();
+        break;
+    }
+
+    // Should not be reached, but the return statement is needed for the code to
+    // build successfully on Android.
+    RTC_NOTREACHED();
+    return false;
+  }
+
+  Random* rand_gen_ = nullptr;
+  FrameCounters* frame_counters_ = nullptr;
+  LockedFlag* capture_call_checker_ = nullptr;
+  CallSimulator* test_ = nullptr;
+  const SimulationConfig* const simulation_config_ = nullptr;
+  AudioProcessing* apm_ = nullptr;
+  AudioFrameData frame_data_;
+  webrtc::Clock* clock_;
+  const size_t num_durations_to_store_;
+  std::vector<int64_t> api_call_durations_;
+  const float input_level_;
+  bool first_process_call_ = true;
+  const ProcessorType processor_type_;
+  const int num_channels_ = 1;
+};
+
+// Class for managing the test simulation.
+class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
+ public:
+  CallSimulator()
+      : test_complete_(EventWrapper::Create()),
+        render_thread_(
+            new rtc::PlatformThread(RenderProcessorThreadFunc, this, "render")),
+        capture_thread_(new rtc::PlatformThread(CaptureProcessorThreadFunc,
+                                                this,
+                                                "capture")),
+        rand_gen_(42U),
+        simulation_config_(static_cast<SimulationConfig>(GetParam())) {}
+
+  // Run the call simulation with a timeout.
+  EventTypeWrapper Run() {
+    StartThreads();
+
+    EventTypeWrapper result = test_complete_->Wait(kTestTimeout);
+
+    StopThreads();
+
+    render_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_render");
+    capture_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_capture");
+
+    return result;
+  }
+
+  // Tests whether all the required render and capture side calls have been
+  // done.
+  bool MaybeEndTest() {
+    if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
+      test_complete_->Set();
+      return true;
+    }
+    return false;
+  }
+
+ private:
+  static const float kCaptureInputFloatLevel;
+  static const float kRenderInputFloatLevel;
+  static const int kMinNumFramesToProcess = 150;
+  static const int32_t kTestTimeout = 3 * 10 * kMinNumFramesToProcess;
+
+  // ::testing::TestWithParam<> implementation.
+  void TearDown() override { StopThreads(); }
+
+  // Stop all running threads.
+  void StopThreads() {
+    render_thread_->Stop();
+    capture_thread_->Stop();
+  }
+
+  // Simulator and APM setup.
+  void SetUp() override {
+    // Lambda function for setting the default APM runtime settings for desktop.
+    auto set_default_desktop_apm_runtime_settings = [](AudioProcessing* apm) {
+      ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError,
+                apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(true));
+      ASSERT_EQ(apm->kNoError,
+                apm->echo_cancellation()->enable_delay_logging(true));
+    };
+
+    // Lambda function for setting the default APM runtime settings for mobile.
+    auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
+      ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError,
+                apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
+    };
+
+    // Lambda function for turning off all of the APM runtime settings
+    // submodules.
+    auto turn_off_default_apm_runtime_settings = [](AudioProcessing* apm) {
+      ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
+      ASSERT_EQ(apm->kNoError,
+                apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(false));
+      ASSERT_EQ(apm->kNoError,
+                apm->echo_cancellation()->enable_delay_logging(false));
+    };
+
+    // Lambda function for adding default desktop APM settings to a config.
+    auto add_default_desktop_config = [](Config* config) {
+      config->Set<ExtendedFilter>(new ExtendedFilter(true));
+      config->Set<DelayAgnostic>(new DelayAgnostic(true));
+    };
+
+    // Lambda function for adding beamformer settings to a config.
+    auto add_beamformer_config = [](Config* config) {
+      const size_t num_mics = 2;
+      const std::vector<Point> array_geometry =
+          ParseArrayGeometry("0 0 0 0.05 0 0", num_mics);
+      RTC_CHECK_EQ(array_geometry.size(), num_mics);
+
+      config->Set<Beamforming>(
+          new Beamforming(true, array_geometry,
+                          SphericalPointf(DegreesToRadians(90), 0.f, 1.f)));
+    };
+
+    int num_capture_channels = 1;
+    switch (simulation_config_.simulation_settings) {
+      case SettingsType::kDefaultApmMobile: {
+        apm_.reset(AudioProcessingImpl::Create());
+        ASSERT_TRUE(!!apm_);
+        set_default_mobile_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktop: {
+        Config config;
+        add_default_desktop_config(&config);
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopAndBeamformer: {
+        Config config;
+        add_beamformer_config(&config);
+        add_default_desktop_config(&config);
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        num_capture_channels = 2;
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer: {
+        Config config;
+        config.Set<Intelligibility>(new Intelligibility(true));
+        add_default_desktop_config(&config);
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+      case SettingsType::kAllSubmodulesTurnedOff: {
+        apm_.reset(AudioProcessingImpl::Create());
+        ASSERT_TRUE(!!apm_);
+        turn_off_default_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic: {
+        Config config;
+        config.Set<ExtendedFilter>(new ExtendedFilter(true));
+        config.Set<DelayAgnostic>(new DelayAgnostic(false));
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+      case SettingsType::kDefaultDesktopApmWithoutExtendedFilter: {
+        Config config;
+        config.Set<ExtendedFilter>(new ExtendedFilter(false));
+        config.Set<DelayAgnostic>(new DelayAgnostic(true));
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+    }
+
+    render_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kRender, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kRenderInputFloatLevel, 1));
+    capture_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kCapture, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kCaptureInputFloatLevel, num_capture_channels));
+  }
+
+  // Thread callback for the render thread.
+  static bool RenderProcessorThreadFunc(void* context) {
+    return reinterpret_cast<CallSimulator*>(context)
+        ->render_thread_state_->Process();
+  }
+
+  // Thread callback for the capture thread.
+  static bool CaptureProcessorThreadFunc(void* context) {
+    return reinterpret_cast<CallSimulator*>(context)
+        ->capture_thread_state_->Process();
+  }
+
+  // Start the threads used in the test.
+  void StartThreads() {
+    ASSERT_NO_FATAL_FAILURE(render_thread_->Start());
+    render_thread_->SetPriority(rtc::kRealtimePriority);
+    ASSERT_NO_FATAL_FAILURE(capture_thread_->Start());
+    capture_thread_->SetPriority(rtc::kRealtimePriority);
+  }
+
+  // Event handler for the test.
+  const rtc::scoped_ptr<EventWrapper> test_complete_;
+
+  // Thread related variables.
+  rtc::scoped_ptr<rtc::PlatformThread> render_thread_;
+  rtc::scoped_ptr<rtc::PlatformThread> capture_thread_;
+  Random rand_gen_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+  const SimulationConfig simulation_config_;
+  FrameCounters frame_counters_;
+  LockedFlag capture_call_checker_;
+  rtc::scoped_ptr<TimedThreadApiProcessor> render_thread_state_;
+  rtc::scoped_ptr<TimedThreadApiProcessor> capture_thread_state_;
+};
+
+// Implements the callback functionality for the threads.
+bool TimedThreadApiProcessor::Process() {
+  PrepareFrame();
+
+  // Wait in a spinlock manner until it is ok to start processing.
+  // Note that SleepMs is not applicable since it only allows sleeping
+  // on a millisecond basis which is too long.
+  while (!ReadyToProcess()) {
+  }
+
+  int result = AudioProcessing::kNoError;
+  switch (processor_type_) {
+    case ProcessorType::kRender:
+      result = ProcessRender();
+      break;
+    case ProcessorType::kCapture:
+      result = ProcessCapture();
+      break;
+  }
+
+  EXPECT_EQ(result, AudioProcessing::kNoError);
+
+  return !test_->MaybeEndTest();
+}
+
+const float CallSimulator::kRenderInputFloatLevel = 0.5f;
+const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
+}  // anonymous namespace
+
+TEST_P(CallSimulator, ApiCallDurationTest) {
+  // Run test and verify that it did not time out.
+  EXPECT_EQ(kEventSignaled, Run());
+}
+
+INSTANTIATE_TEST_CASE_P(
+    AudioProcessingPerformanceTest,
+    CallSimulator,
+    ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/audio_processing_tests.gypi b/webrtc/modules/audio_processing/audio_processing_tests.gypi
index 0314c69b04..523602baba 100644
--- a/webrtc/modules/audio_processing/audio_processing_tests.gypi
+++ b/webrtc/modules/audio_processing/audio_processing_tests.gypi
@@ -128,7 +128,11 @@
             '<(webrtc_root)/test/test.gyp:test_support',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
           ],
-          'sources': [ 'test/audioproc_float.cc', ],
+          'sources': [
+            'test/audio_file_processor.cc',
+            'test/audio_file_processor.h',
+            'test/audioproc_float.cc',
+          ],
         },
         {
           'target_name': 'unpack_aecdump',
diff --git a/webrtc/modules/audio_processing/beamformer/array_util.cc b/webrtc/modules/audio_processing/beamformer/array_util.cc
index 8aaeee9f59..6b1c474269 100644
--- a/webrtc/modules/audio_processing/beamformer/array_util.cc
+++ b/webrtc/modules/audio_processing/beamformer/array_util.cc
@@ -56,7 +56,7 @@ bool ArePerpendicular(const Point& a, const Point& b) {
   return std::abs(DotProduct(a, b)) < kMaxDotProduct;
 }
 
-rtc::Maybe<Point> GetDirectionIfLinear(
+rtc::Optional<Point> GetDirectionIfLinear(
     const std::vector<Point>& array_geometry) {
   RTC_DCHECK_GT(array_geometry.size(), 1u);
   const Point first_pair_direction =
@@ -65,13 +65,14 @@ rtc::Maybe<Point> GetDirectionIfLinear(
     const Point pair_direction =
         PairDirection(array_geometry[i - 1], array_geometry[i]);
     if (!AreParallel(first_pair_direction, pair_direction)) {
-      return rtc::Maybe<Point>();
+      return rtc::Optional<Point>();
     }
   }
-  return first_pair_direction;
+  return rtc::Optional<Point>(first_pair_direction);
 }
 
-rtc::Maybe<Point> GetNormalIfPlanar(const std::vector<Point>& array_geometry) {
+rtc::Optional<Point> GetNormalIfPlanar(
+    const std::vector<Point>& array_geometry) {
   RTC_DCHECK_GT(array_geometry.size(), 1u);
   const Point first_pair_direction =
       PairDirection(array_geometry[0], array_geometry[1]);
@@ -85,30 +86,30 @@ rtc::Maybe<Point> GetNormalIfPlanar(const std::vector<Point>& array_geometry) {
     }
   }
   if (is_linear) {
-    return rtc::Maybe<Point>();
+    return rtc::Optional<Point>();
   }
   const Point normal_direction =
       CrossProduct(first_pair_direction, pair_direction);
   for (; i < array_geometry.size(); ++i) {
     pair_direction = PairDirection(array_geometry[i - 1], array_geometry[i]);
     if (!ArePerpendicular(normal_direction, pair_direction)) {
-      return rtc::Maybe<Point>();
+      return rtc::Optional<Point>();
     }
   }
-  return normal_direction;
+  return rtc::Optional<Point>(normal_direction);
 }
 
-rtc::Maybe<Point> GetArrayNormalIfExists(
+rtc::Optional<Point> GetArrayNormalIfExists(
     const std::vector<Point>& array_geometry) {
-  const rtc::Maybe<Point> direction = GetDirectionIfLinear(array_geometry);
+  const rtc::Optional<Point> direction = GetDirectionIfLinear(array_geometry);
   if (direction) {
-    return Point(direction->y(), -direction->x(), 0.f);
+    return rtc::Optional<Point>(Point(direction->y(), -direction->x(), 0.f));
   }
-  const rtc::Maybe<Point> normal = GetNormalIfPlanar(array_geometry);
+  const rtc::Optional<Point> normal = GetNormalIfPlanar(array_geometry);
   if (normal && normal->z() < kMaxDotProduct) {
     return normal;
   }
-  return rtc::Maybe<Point>();
+  return rtc::Optional<Point>();
 }
 
 Point AzimuthToPoint(float azimuth) {
diff --git a/webrtc/modules/audio_processing/beamformer/array_util.h b/webrtc/modules/audio_processing/beamformer/array_util.h
index 7fff9735a1..f86ad5dee6 100644
--- a/webrtc/modules/audio_processing/beamformer/array_util.h
+++ b/webrtc/modules/audio_processing/beamformer/array_util.h
@@ -14,7 +14,7 @@
 #include <cmath>
 #include <vector>
 
-#include "webrtc/base/maybe.h"
+#include "webrtc/base/optional.h"
 
 namespace webrtc {
 
@@ -59,15 +59,16 @@ float GetMinimumSpacing(const std::vector<Point>& array_geometry);
 
 // If the given array geometry is linear it returns the direction without
 // normalizing.
-rtc::Maybe<Point> GetDirectionIfLinear(
+rtc::Optional<Point> GetDirectionIfLinear(
     const std::vector<Point>& array_geometry);
 
 // If the given array geometry is planar it returns the normal without
 // normalizing.
-rtc::Maybe<Point> GetNormalIfPlanar(const std::vector<Point>& array_geometry);
+rtc::Optional<Point> GetNormalIfPlanar(
+    const std::vector<Point>& array_geometry);
 
 // Returns the normal of an array if it has one and it is in the xy-plane.
-rtc::Maybe<Point> GetArrayNormalIfExists(
+rtc::Optional<Point> GetArrayNormalIfExists(
     const std::vector<Point>& array_geometry);
 
 // The resulting Point will be in the xy-plane.
diff --git a/webrtc/modules/audio_processing/beamformer/complex_matrix.h b/webrtc/modules/audio_processing/beamformer/complex_matrix.h
index bfa3563b89..707c51564b 100644
--- a/webrtc/modules/audio_processing/beamformer/complex_matrix.h
+++ b/webrtc/modules/audio_processing/beamformer/complex_matrix.h
@@ -27,10 +27,10 @@ class ComplexMatrix : public Matrix<complex<T> > {
  public:
   ComplexMatrix() : Matrix<complex<T> >() {}
 
-  ComplexMatrix(int num_rows, int num_columns)
+  ComplexMatrix(size_t num_rows, size_t num_columns)
       : Matrix<complex<T> >(num_rows, num_columns) {}
 
-  ComplexMatrix(const complex<T>* data, int num_rows, int num_columns)
+  ComplexMatrix(const complex<T>* data, size_t num_rows, size_t num_columns)
       : Matrix<complex<T> >(data, num_rows, num_columns) {}
 
   // Complex Matrix operations.
@@ -51,7 +51,7 @@ class ComplexMatrix : public Matrix<complex<T> > {
 
   ComplexMatrix& ConjugateTranspose() {
     this->CopyDataToScratch();
-    int num_rows = this->num_rows();
+    size_t num_rows = this->num_rows();
     this->SetNumRows(this->num_columns());
     this->SetNumColumns(num_rows);
     this->Resize();
@@ -82,8 +82,8 @@ class ComplexMatrix : public Matrix<complex<T> > {
  private:
   ComplexMatrix& ConjugateTranspose(const complex<T>* const* src) {
     complex<T>* const* elements = this->elements();
-    for (int i = 0; i < this->num_rows(); ++i) {
-      for (int j = 0; j < this->num_columns(); ++j) {
+    for (size_t i = 0; i < this->num_rows(); ++i) {
+      for (size_t j = 0; j < this->num_columns(); ++j) {
         elements[i][j] = conj(src[j][i]);
       }
     }
diff --git a/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc b/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
index d0728325fc..78f4df5ca9 100644
--- a/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
+++ b/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
@@ -27,7 +27,7 @@ float BesselJ0(float x) {
 
 // Calculates the Euclidean norm for a row vector.
 float Norm(const ComplexMatrix<float>& x) {
-  RTC_CHECK_EQ(1, x.num_rows());
+  RTC_CHECK_EQ(1u, x.num_rows());
   const size_t length = x.num_columns();
   const complex<float>* elems = x.elements()[0];
   float result = 0.f;
@@ -43,8 +43,8 @@ void CovarianceMatrixGenerator::UniformCovarianceMatrix(
     float wave_number,
     const std::vector<Point>& geometry,
     ComplexMatrix<float>* mat) {
-  RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_rows());
-  RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+  RTC_CHECK_EQ(geometry.size(), mat->num_rows());
+  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
 
   complex<float>* const* mat_els = mat->elements();
   for (size_t i = 0; i < geometry.size(); ++i) {
@@ -68,8 +68,8 @@ void CovarianceMatrixGenerator::AngledCovarianceMatrix(
     int sample_rate,
     const std::vector<Point>& geometry,
     ComplexMatrix<float>* mat) {
-  RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_rows());
-  RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+  RTC_CHECK_EQ(geometry.size(), mat->num_rows());
+  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
 
   ComplexMatrix<float> interf_cov_vector(1, geometry.size());
   ComplexMatrix<float> interf_cov_vector_transposed(geometry.size(), 1);
@@ -94,8 +94,8 @@ void CovarianceMatrixGenerator::PhaseAlignmentMasks(
     const std::vector<Point>& geometry,
     float angle,
     ComplexMatrix<float>* mat) {
-  RTC_CHECK_EQ(1, mat->num_rows());
-  RTC_CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+  RTC_CHECK_EQ(1u, mat->num_rows());
+  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
 
   float freq_in_hertz =
       (static_cast<float>(frequency_bin) / fft_size) * sample_rate;
diff --git a/webrtc/modules/audio_processing/beamformer/matrix.h b/webrtc/modules/audio_processing/beamformer/matrix.h
index 162aef1dac..51c1cece97 100644
--- a/webrtc/modules/audio_processing/beamformer/matrix.h
+++ b/webrtc/modules/audio_processing/beamformer/matrix.h
@@ -67,7 +67,7 @@ class Matrix {
   Matrix() : num_rows_(0), num_columns_(0) {}
 
   // Allocates space for the elements and initializes all values to zero.
-  Matrix(int num_rows, int num_columns)
+  Matrix(size_t num_rows, size_t num_columns)
       : num_rows_(num_rows), num_columns_(num_columns) {
     Resize();
     scratch_data_.resize(num_rows_ * num_columns_);
@@ -75,7 +75,7 @@ class Matrix {
   }
 
   // Copies |data| into the new Matrix.
-  Matrix(const T* data, int num_rows, int num_columns)
+  Matrix(const T* data, size_t num_rows, size_t num_columns)
       : num_rows_(0), num_columns_(0) {
     CopyFrom(data, num_rows, num_columns);
     scratch_data_.resize(num_rows_ * num_columns_);
@@ -90,23 +90,23 @@ class Matrix {
   }
 
   // Copy |data| into the Matrix. The current data is lost.
-  void CopyFrom(const T* const data, int num_rows, int num_columns) {
+  void CopyFrom(const T* const data, size_t num_rows, size_t num_columns) {
     Resize(num_rows, num_columns);
     memcpy(&data_[0], data, num_rows_ * num_columns_ * sizeof(data_[0]));
   }
 
   Matrix& CopyFromColumn(const T* const* src,
                          size_t column_index,
-                         int num_rows) {
+                         size_t num_rows) {
     Resize(1, num_rows);
-    for (int i = 0; i < num_columns_; ++i) {
+    for (size_t i = 0; i < num_columns_; ++i) {
       data_[i] = src[i][column_index];
     }
 
     return *this;
   }
 
-  void Resize(int num_rows, int num_columns) {
+  void Resize(size_t num_rows, size_t num_columns) {
     if (num_rows != num_rows_ || num_columns != num_columns_) {
       num_rows_ = num_rows;
       num_columns_ = num_columns;
@@ -115,8 +115,8 @@ class Matrix {
   }
 
   // Accessors and mutators.
-  int num_rows() const { return num_rows_; }
-  int num_columns() const { return num_columns_; }
+  size_t num_rows() const { return num_rows_; }
+  size_t num_columns() const { return num_columns_; }
   T* const* elements() { return &elements_[0]; }
   const T* const* elements() const { return &elements_[0]; }
 
@@ -124,7 +124,7 @@ class Matrix {
     RTC_CHECK_EQ(num_rows_, num_columns_);
 
     T trace = 0;
-    for (int i = 0; i < num_rows_; ++i) {
+    for (size_t i = 0; i < num_rows_; ++i) {
       trace += elements_[i][i];
     }
     return trace;
@@ -282,8 +282,8 @@ class Matrix {
     std::ostringstream ss;
     ss << std::endl << "Matrix" << std::endl;
 
-    for (int i = 0; i < num_rows_; ++i) {
-      for (int j = 0; j < num_columns_; ++j) {
+    for (size_t i = 0; i < num_rows_; ++i) {
+      for (size_t j = 0; j < num_columns_; ++j) {
         ss << elements_[i][j] << " ";
       }
       ss << std::endl;
@@ -294,8 +294,8 @@ class Matrix {
   }
 
  protected:
-  void SetNumRows(const int num_rows) { num_rows_ = num_rows; }
-  void SetNumColumns(const int num_columns) { num_columns_ = num_columns; }
+  void SetNumRows(const size_t num_rows) { num_rows_ = num_rows; }
+  void SetNumColumns(const size_t num_columns) { num_columns_ = num_columns; }
   T* data() { return &data_[0]; }
   const T* data() const { return &data_[0]; }
   const T* const* scratch_elements() const { return &scratch_elements_[0]; }
@@ -307,7 +307,7 @@ class Matrix {
     data_.resize(size);
     elements_.resize(num_rows_);
 
-    for (int i = 0; i < num_rows_; ++i) {
+    for (size_t i = 0; i < num_rows_; ++i) {
       elements_[i] = &data_[i * num_columns_];
     }
   }
@@ -317,14 +317,14 @@ class Matrix {
     scratch_data_ = data_;
     scratch_elements_.resize(num_rows_);
 
-    for (int i = 0; i < num_rows_; ++i) {
+    for (size_t i = 0; i < num_rows_; ++i) {
       scratch_elements_[i] = &scratch_data_[i * num_columns_];
     }
   }
 
  private:
-  int num_rows_;
-  int num_columns_;
+  size_t num_rows_;
+  size_t num_columns_;
   std::vector<T> data_;
   std::vector<T*> elements_;
 
@@ -336,8 +336,8 @@ class Matrix {
   // Helpers for Transpose and Multiply operations that unify in-place and
   // out-of-place solutions.
   Matrix& Transpose(const T* const* src) {
-    for (int i = 0; i < num_rows_; ++i) {
-      for (int j = 0; j < num_columns_; ++j) {
+    for (size_t i = 0; i < num_rows_; ++i) {
+      for (size_t j = 0; j < num_columns_; ++j) {
         elements_[i][j] = src[j][i];
       }
     }
@@ -345,11 +345,13 @@ class Matrix {
     return *this;
   }
 
-  Matrix& Multiply(const T* const* lhs, int num_rows_rhs, const T* const* rhs) {
-    for (int row = 0; row < num_rows_; ++row) {
-      for (int col = 0; col < num_columns_; ++col) {
+  Matrix& Multiply(const T* const* lhs,
+                   size_t num_rows_rhs,
+                   const T* const* rhs) {
+    for (size_t row = 0; row < num_rows_; ++row) {
+      for (size_t col = 0; col < num_columns_; ++col) {
         T cur_element = 0;
-        for (int i = 0; i < num_rows_rhs; ++i) {
+        for (size_t i = 0; i < num_rows_rhs; ++i) {
           cur_element += lhs[row][i] * rhs[i][col];
         }
 
diff --git a/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h b/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
index 7c58670068..9891a8220c 100644
--- a/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
+++ b/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
@@ -34,8 +34,8 @@ class MatrixTestHelpers {
 
     const T* const* expected_elements = expected.elements();
     const T* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_EQ(expected_elements[i][j], actual_elements[i][j]);
       }
     }
@@ -48,8 +48,8 @@ class MatrixTestHelpers {
 
     const float* const* expected_elements = expected.elements();
     const float* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_NEAR(expected_elements[i][j], actual_elements[i][j], kTolerance);
       }
     }
@@ -63,8 +63,8 @@ class MatrixTestHelpers {
 
     const complex<float>* const* expected_elements = expected.elements();
     const complex<float>* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_NEAR(expected_elements[i][j].real(),
                     actual_elements[i][j].real(),
                     kTolerance);
@@ -84,8 +84,8 @@ class MatrixTestHelpers {
 
     const complex<float>* const* expected_elements = expected.elements();
     const complex<float>* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_NEAR(expected_elements[i][j].real(),
                     actual_elements[i][j].real(),
                     tolerance);
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
index 029fa089fc..6ea7234f6f 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
@@ -79,7 +79,7 @@ const float kCompensationGain = 2.f;
 // The returned norm is clamped to be non-negative.
 float Norm(const ComplexMatrix<float>& mat,
            const ComplexMatrix<float>& norm_mat) {
-  RTC_CHECK_EQ(norm_mat.num_rows(), 1);
+  RTC_CHECK_EQ(1u, norm_mat.num_rows());
   RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_rows());
   RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_columns());
 
@@ -89,8 +89,8 @@ float Norm(const ComplexMatrix<float>& mat,
   const complex<float>* const* mat_els = mat.elements();
   const complex<float>* const* norm_mat_els = norm_mat.elements();
 
-  for (int i = 0; i < norm_mat.num_columns(); ++i) {
-    for (int j = 0; j < norm_mat.num_columns(); ++j) {
+  for (size_t i = 0; i < norm_mat.num_columns(); ++i) {
+    for (size_t j = 0; j < norm_mat.num_columns(); ++j) {
       first_product += conj(norm_mat_els[0][j]) * mat_els[j][i];
     }
     second_product += first_product * norm_mat_els[0][i];
@@ -102,15 +102,15 @@ float Norm(const ComplexMatrix<float>& mat,
 // Does conjugate(|lhs|) * |rhs| for row vectors |lhs| and |rhs|.
 complex<float> ConjugateDotProduct(const ComplexMatrix<float>& lhs,
                                    const ComplexMatrix<float>& rhs) {
-  RTC_CHECK_EQ(lhs.num_rows(), 1);
-  RTC_CHECK_EQ(rhs.num_rows(), 1);
+  RTC_CHECK_EQ(1u, lhs.num_rows());
+  RTC_CHECK_EQ(1u, rhs.num_rows());
   RTC_CHECK_EQ(lhs.num_columns(), rhs.num_columns());
 
   const complex<float>* const* lhs_elements = lhs.elements();
   const complex<float>* const* rhs_elements = rhs.elements();
 
   complex<float> result = complex<float>(0.f, 0.f);
-  for (int i = 0; i < lhs.num_columns(); ++i) {
+  for (size_t i = 0; i < lhs.num_columns(); ++i) {
     result += conj(lhs_elements[0][i]) * rhs_elements[0][i];
   }
 
@@ -126,8 +126,8 @@ size_t Round(float x) {
 float SumAbs(const ComplexMatrix<float>& mat) {
   float sum_abs = 0.f;
   const complex<float>* const* mat_els = mat.elements();
-  for (int i = 0; i < mat.num_rows(); ++i) {
-    for (int j = 0; j < mat.num_columns(); ++j) {
+  for (size_t i = 0; i < mat.num_rows(); ++i) {
+    for (size_t j = 0; j < mat.num_columns(); ++j) {
       sum_abs += std::abs(mat_els[i][j]);
     }
   }
@@ -138,8 +138,8 @@ float SumAbs(const ComplexMatrix<float>& mat) {
 float SumSquares(const ComplexMatrix<float>& mat) {
   float sum_squares = 0.f;
   const complex<float>* const* mat_els = mat.elements();
-  for (int i = 0; i < mat.num_rows(); ++i) {
-    for (int j = 0; j < mat.num_columns(); ++j) {
+  for (size_t i = 0; i < mat.num_rows(); ++i) {
+    for (size_t j = 0; j < mat.num_columns(); ++j) {
       float abs_value = std::abs(mat_els[i][j]);
       sum_squares += abs_value * abs_value;
     }
@@ -150,20 +150,20 @@ float SumSquares(const ComplexMatrix<float>& mat) {
 // Does |out| = |in|.' * conj(|in|) for row vector |in|.
 void TransposedConjugatedProduct(const ComplexMatrix<float>& in,
                                  ComplexMatrix<float>* out) {
-  RTC_CHECK_EQ(in.num_rows(), 1);
+  RTC_CHECK_EQ(1u, in.num_rows());
   RTC_CHECK_EQ(out->num_rows(), in.num_columns());
   RTC_CHECK_EQ(out->num_columns(), in.num_columns());
   const complex<float>* in_elements = in.elements()[0];
   complex<float>* const* out_elements = out->elements();
-  for (int i = 0; i < out->num_rows(); ++i) {
-    for (int j = 0; j < out->num_columns(); ++j) {
+  for (size_t i = 0; i < out->num_rows(); ++i) {
+    for (size_t j = 0; j < out->num_columns(); ++j) {
       out_elements[i][j] = in_elements[i] * conj(in_elements[j]);
     }
   }
 }
 
 std::vector<Point> GetCenteredArray(std::vector<Point> array_geometry) {
-  for (int dim = 0; dim < 3; ++dim) {
+  for (size_t dim = 0; dim < 3; ++dim) {
     float center = 0.f;
     for (size_t i = 0; i < array_geometry.size(); ++i) {
       center += array_geometry[i].c[dim];
@@ -379,7 +379,7 @@ void NonlinearBeamformer::ProcessChunk(const ChannelBuffer<float>& input,
       (high_pass_postfilter_mask_ - old_high_pass_mask) /
       input.num_frames_per_band();
   // Apply the smoothed high-pass mask to the first channel of each band.
-  // This can be done because the effct of the linear beamformer is negligible
+  // This can be done because the effect of the linear beamformer is negligible
   // compared to the post-filter.
   for (size_t i = 1; i < input.num_bands(); ++i) {
     float smoothed_mask = old_high_pass_mask;
@@ -408,13 +408,13 @@ bool NonlinearBeamformer::IsInBeam(const SphericalPointf& spherical_point) {
 }
 
 void NonlinearBeamformer::ProcessAudioBlock(const complex_f* const* input,
-                                            int num_input_channels,
+                                            size_t num_input_channels,
                                             size_t num_freq_bins,
-                                            int num_output_channels,
+                                            size_t num_output_channels,
                                             complex_f* const* output) {
-  RTC_CHECK_EQ(num_freq_bins, kNumFreqBins);
-  RTC_CHECK_EQ(num_input_channels, num_input_channels_);
-  RTC_CHECK_EQ(num_output_channels, 1);
+  RTC_CHECK_EQ(kNumFreqBins, num_freq_bins);
+  RTC_CHECK_EQ(num_input_channels_, num_input_channels);
+  RTC_CHECK_EQ(1u, num_output_channels);
 
   // Calculating the post-filter masks. Note that we need two for each
   // frequency bin to account for the positive and negative interferer
@@ -483,7 +483,7 @@ void NonlinearBeamformer::ApplyMasks(const complex_f* const* input,
 
     const complex_f* delay_sum_mask_els =
         normalized_delay_sum_masks_[f_ix].elements()[0];
-    for (int c_ix = 0; c_ix < num_input_channels_; ++c_ix) {
+    for (size_t c_ix = 0; c_ix < num_input_channels_; ++c_ix) {
       output_channel[f_ix] += input[c_ix][f_ix] * delay_sum_mask_els[c_ix];
     }
 
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
index 565c1f349f..29c416ca91 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
@@ -67,9 +67,9 @@ class NonlinearBeamformer
   // Process one frequency-domain block of audio. This is where the fun
   // happens. Implements LappedTransform::Callback.
   void ProcessAudioBlock(const complex<float>* const* input,
-                         int num_input_channels,
+                         size_t num_input_channels,
                          size_t num_freq_bins,
-                         int num_output_channels,
+                         size_t num_output_channels,
                          complex<float>* const* output) override;
 
  private:
@@ -129,12 +129,12 @@ class NonlinearBeamformer
   float window_[kFftSize];
 
   // Parameters exposed to the user.
-  const int num_input_channels_;
+  const size_t num_input_channels_;
   int sample_rate_hz_;
 
   const std::vector<Point> array_geometry_;
   // The normal direction of the array if it has one and it is in the xy-plane.
-  const rtc::Maybe<Point> array_normal_;
+  const rtc::Optional<Point> array_normal_;
 
   // Minimum spacing between microphone pairs.
   const float min_mic_spacing_;
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
index cc752485e9..d187552692 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
@@ -12,6 +12,7 @@
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
@@ -52,9 +53,9 @@ int main(int argc, char* argv[]) {
   NonlinearBeamformer bf(array_geometry);
   bf.Initialize(kChunkSizeMs, in_file.sample_rate());
 
-  printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+  printf("Input file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
          FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
-  printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+  printf("Output file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
          FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
 
   ChannelBuffer<float> in_buf(
diff --git a/webrtc/modules/audio_processing/common.h b/webrtc/modules/audio_processing/common.h
index ed8a0544c3..d4ddb92b50 100644
--- a/webrtc/modules/audio_processing/common.h
+++ b/webrtc/modules/audio_processing/common.h
@@ -17,7 +17,7 @@
 
 namespace webrtc {
 
-static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+static inline size_t ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
     case AudioProcessing::kMonoAndKeyboard:
@@ -27,7 +27,7 @@ static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
       return 2;
   }
   assert(false);
-  return -1;
+  return 0;
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index 56ee9e0fff..debc597c54 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -16,9 +16,8 @@
 extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -53,13 +52,22 @@ AudioProcessing::Error MapError(int err) {
       return AudioProcessing::kUnspecifiedError;
   }
 }
+
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
 }  // namespace
 
 EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm,
-                                           CriticalSectionWrapper* crit)
+                                           rtc::CriticalSection* crit_render,
+                                           rtc::CriticalSection* crit_capture)
     : ProcessingComponent(),
       apm_(apm),
-      crit_(crit),
+      crit_render_(crit_render),
+      crit_capture_(crit_capture),
       drift_compensation_enabled_(false),
       metrics_enabled_(false),
       suppression_level_(kModerateSuppression),
@@ -68,87 +76,131 @@ EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm,
       stream_has_echo_(false),
       delay_logging_enabled_(false),
       extended_filter_enabled_(false),
-      delay_agnostic_enabled_(false) {
+      delay_agnostic_enabled_(false),
+      render_queue_element_max_size_(0) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
 }
 
 EchoCancellationImpl::~EchoCancellationImpl() {}
 
 int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
+  rtc::CritScope cs_render(crit_render_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_reverse_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
+  render_queue_buffer_.clear();
+  for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+    for (size_t j = 0; j < audio->num_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAec_BufferFarend(
-          my_handle,
-          audio->split_bands_const_f(j)[kBand0To8kHz],
+      // Retrieve any error code produced by the buffering of the farend
+      // signal
+      err = WebRtcAec_GetBufferFarendError(
+          my_handle, audio->split_bands_const_f(j)[kBand0To8kHz],
           audio->num_frames_per_band());
 
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
+      if (err != AudioProcessing::kNoError) {
+        return MapError(err);  // TODO(ajm): warning possible?
       }
 
-      handle_index++;
+      // Buffer the samples in the render queue.
+      render_queue_buffer_.insert(render_queue_buffer_.end(),
+                                  audio->split_bands_const_f(j)[kBand0To8kHz],
+                                  (audio->split_bands_const_f(j)[kBand0To8kHz] +
+                                   audio->num_frames_per_band()));
     }
   }
 
-  return apm_->kNoError;
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    ReadQueuedRenderData();
+
+    // Retry the insert (should always work).
+    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+// Read chunks of data that were received and queued on the render side from
+// a queue. All the data chunks are buffered into the farend signal of the AEC.
+void EchoCancellationImpl::ReadQueuedRenderData() {
+  rtc::CritScope cs_capture(crit_capture_);
+  if (!is_component_enabled()) {
+    return;
+  }
+
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+    size_t handle_index = 0;
+    size_t buffer_index = 0;
+    const size_t num_frames_per_band =
+        capture_queue_buffer_.size() /
+        (apm_->num_output_channels() * apm_->num_reverse_channels());
+    for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+      for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
+        Handle* my_handle = static_cast<Handle*>(handle(handle_index));
+        WebRtcAec_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
+                               num_frames_per_band);
+
+        buffer_index += num_frames_per_band;
+        handle_index++;
+      }
+    }
+  }
 }
 
 int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs_capture(crit_capture_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   if (drift_compensation_enabled_ && !was_stream_drift_set_) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
-  assert(audio->num_channels() == apm_->num_output_channels());
+  assert(audio->num_channels() == apm_->num_proc_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   stream_has_echo_ = false;
-  for (int i = 0; i < audio->num_channels(); i++) {
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+  for (size_t i = 0; i < audio->num_channels(); i++) {
+    for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
       Handle* my_handle = handle(handle_index);
-      err = WebRtcAec_Process(
-          my_handle,
-          audio->split_bands_const_f(i),
-          audio->num_bands(),
-          audio->split_bands_f(i),
-          audio->num_frames_per_band(),
-          apm_->stream_delay_ms(),
-          stream_drift_samples_);
-
-      if (err != apm_->kNoError) {
-        err = GetHandleError(my_handle);
+      err = WebRtcAec_Process(my_handle, audio->split_bands_const_f(i),
+                              audio->num_bands(), audio->split_bands_f(i),
+                              audio->num_frames_per_band(),
+                              apm_->stream_delay_ms(), stream_drift_samples_);
+
+      if (err != AudioProcessing::kNoError) {
+        err = MapError(err);
         // TODO(ajm): Figure out how to return warnings properly.
-        if (err != apm_->kBadStreamParameterWarning) {
+        if (err != AudioProcessing::kBadStreamParameterWarning) {
           return err;
         }
       }
 
       int status = 0;
       err = WebRtcAec_get_echo_status(my_handle, &status);
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);
+      if (err != AudioProcessing::kNoError) {
+        return MapError(err);
       }
 
       if (status == 1) {
@@ -160,77 +212,92 @@ int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   }
 
   was_stream_drift_set_ = false;
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoCancellationImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   // Ensure AEC and AECM are not both enabled.
+  // The is_enabled call is safe from a deadlock perspective
+  // as both locks are already held in the correct order.
   if (enable && apm_->echo_control_mobile()->is_enabled()) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   return EnableComponent(enable);
 }
 
 bool EchoCancellationImpl::is_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }
 
 int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
+  {
+    if (MapSetting(level) == -1) {
+      return AudioProcessing::kBadParameterError;
+    }
+    rtc::CritScope cs(crit_capture_);
+    suppression_level_ = level;
   }
-
-  suppression_level_ = level;
   return Configure();
 }
 
 EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
     const {
+  rtc::CritScope cs(crit_capture_);
   return suppression_level_;
 }
 
 int EchoCancellationImpl::enable_drift_compensation(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  drift_compensation_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    drift_compensation_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::is_drift_compensation_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return drift_compensation_enabled_;
 }
 
 void EchoCancellationImpl::set_stream_drift_samples(int drift) {
+  rtc::CritScope cs(crit_capture_);
   was_stream_drift_set_ = true;
   stream_drift_samples_ = drift;
 }
 
 int EchoCancellationImpl::stream_drift_samples() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_drift_samples_;
 }
 
 int EchoCancellationImpl::enable_metrics(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  metrics_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    metrics_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::are_metrics_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return metrics_enabled_;
 }
 
 // TODO(ajm): we currently just use the metrics from the first AEC. Think more
 //            aboue the best way to extend this to multi-channel.
 int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (metrics == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
 
   if (!is_component_enabled() || !metrics_enabled_) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   AecMetrics my_metrics;
@@ -239,8 +306,8 @@ int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
 
   Handle* my_handle = static_cast<Handle*>(handle(0));
   int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
-  if (err != apm_->kNoError) {
-    return GetHandleError(my_handle);
+  if (err != AudioProcessing::kNoError) {
+    return MapError(err);
   }
 
   metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
@@ -263,62 +330,70 @@ int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
   metrics->a_nlp.maximum = my_metrics.aNlp.max;
   metrics->a_nlp.minimum = my_metrics.aNlp.min;
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 bool EchoCancellationImpl::stream_has_echo() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_has_echo_;
 }
 
 int EchoCancellationImpl::enable_delay_logging(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  delay_logging_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    delay_logging_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::is_delay_logging_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return delay_logging_enabled_;
 }
 
 bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return delay_agnostic_enabled_;
 }
 
 bool EchoCancellationImpl::is_extended_filter_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return extended_filter_enabled_;
 }
 
 // TODO(bjornv): How should we handle the multi-channel case?
 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
+  rtc::CritScope cs(crit_capture_);
   float fraction_poor_delays = 0;
   return GetDelayMetrics(median, std, &fraction_poor_delays);
 }
 
 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std,
                                           float* fraction_poor_delays) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (median == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
   if (std == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
 
   if (!is_component_enabled() || !delay_logging_enabled_) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   Handle* my_handle = static_cast<Handle*>(handle(0));
-  if (WebRtcAec_GetDelayMetrics(my_handle, median, std, fraction_poor_delays) !=
-      apm_->kNoError) {
-    return GetHandleError(my_handle);
+  const int err =
+      WebRtcAec_GetDelayMetrics(my_handle, median, std, fraction_poor_delays);
+  if (err != AudioProcessing::kNoError) {
+    return MapError(err);
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 struct AecCore* EchoCancellationImpl::aec_core() const {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (!is_component_enabled()) {
     return NULL;
   }
@@ -328,16 +403,51 @@ struct AecCore* EchoCancellationImpl::aec_core() const {
 
 int EchoCancellationImpl::Initialize() {
   int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
+  {
+    rtc::CritScope cs(crit_capture_);
+    if (err != AudioProcessing::kNoError || !is_component_enabled()) {
+      return err;
+    }
   }
 
-  return apm_->kNoError;
+  AllocateRenderQueue();
+
+  return AudioProcessing::kNoError;
+}
+
+void EchoCancellationImpl::AllocateRenderQueue() {
+  const size_t new_render_queue_element_max_size = std::max<size_t>(
+      static_cast<size_t>(1),
+      kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
+
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
+  // Reallocate the queue if the queue item size is too small to fit the
+  // data to put in the queue.
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
+
+    std::vector<float> template_queue_element(render_queue_element_max_size_);
+
+    render_signal_queue_.reset(
+        new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<float>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
+  } else {
+    render_signal_queue_->Clear();
+  }
 }
 
 void EchoCancellationImpl::SetExtraOptions(const Config& config) {
-  extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
-  delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
+  {
+    rtc::CritScope cs(crit_capture_);
+    extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
+    delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
+  }
   Configure();
 }
 
@@ -351,23 +461,25 @@ void EchoCancellationImpl::DestroyHandle(void* handle) const {
 }
 
 int EchoCancellationImpl::InitializeHandle(void* handle) const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+
   assert(handle != NULL);
   // TODO(ajm): Drift compensation is disabled in practice. If restored, it
   // should be managed internally and not depend on the hardware sample rate.
   // For now, just hardcode a 48 kHz value.
   return WebRtcAec_Init(static_cast<Handle*>(handle),
-                       apm_->proc_sample_rate_hz(),
-                       48000);
+                        apm_->proc_sample_rate_hz(), 48000);
 }
 
 int EchoCancellationImpl::ConfigureHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   assert(handle != NULL);
   AecConfig config;
   config.metricsMode = metrics_enabled_;
   config.nlpMode = MapSetting(suppression_level_);
   config.skewMode = drift_compensation_enabled_;
   config.delay_logging = delay_logging_enabled_;
-
   WebRtcAec_enable_extended_filter(
       WebRtcAec_aec_core(static_cast<Handle*>(handle)),
       extended_filter_enabled_ ? 1 : 0);
@@ -377,13 +489,14 @@ int EchoCancellationImpl::ConfigureHandle(void* handle) const {
   return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
 }
 
-int EchoCancellationImpl::num_handles_required() const {
-  return apm_->num_output_channels() *
-         apm_->num_reverse_channels();
+size_t EchoCancellationImpl::num_handles_required() const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+  return apm_->num_output_channels() * apm_->num_reverse_channels();
 }
 
 int EchoCancellationImpl::GetHandleError(void* handle) const {
+  // Not locked as it does not rely on anything in the state.
   assert(handle != NULL);
-  return MapError(WebRtcAec_get_error_code(static_cast<Handle*>(handle)));
+  return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.h b/webrtc/modules/audio_processing/echo_cancellation_impl.h
index 070dcabc5d..a40a267e32 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.h
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.h
@@ -11,19 +11,22 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
 class EchoCancellationImpl : public EchoCancellation,
                              public ProcessingComponent {
  public:
   EchoCancellationImpl(const AudioProcessing* apm,
-                       CriticalSectionWrapper* crit);
+                       rtc::CriticalSection* crit_render,
+                       rtc::CriticalSection* crit_capture);
   virtual ~EchoCancellationImpl();
 
   int ProcessRenderAudio(const AudioBuffer* audio);
@@ -38,10 +41,13 @@ class EchoCancellationImpl : public EchoCancellation,
   // ProcessingComponent implementation.
   int Initialize() override;
   void SetExtraOptions(const Config& config) override;
-
   bool is_delay_agnostic_enabled() const;
   bool is_extended_filter_enabled() const;
 
+  // Reads render side data that has been queued on the render call.
+  // Called holding the capture lock.
+  void ReadQueuedRenderData();
+
  private:
   // EchoCancellation implementation.
   int Enable(bool enable) override;
@@ -58,6 +64,7 @@ class EchoCancellationImpl : public EchoCancellation,
   int GetDelayMetrics(int* median,
                       int* std,
                       float* fraction_poor_delays) override;
+
   struct AecCore* aec_core() const override;
 
   // ProcessingComponent implementation.
@@ -65,20 +72,35 @@ class EchoCancellationImpl : public EchoCancellation,
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
+  size_t num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  void AllocateRenderQueue();
+
+  // Not guarded as its public API is thread safe.
   const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  bool drift_compensation_enabled_;
-  bool metrics_enabled_;
-  SuppressionLevel suppression_level_;
-  int stream_drift_samples_;
-  bool was_stream_drift_set_;
-  bool stream_has_echo_;
-  bool delay_logging_enabled_;
-  bool extended_filter_enabled_;
-  bool delay_agnostic_enabled_;
+
+  rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  rtc::CriticalSection* const crit_capture_;
+
+  bool drift_compensation_enabled_ GUARDED_BY(crit_capture_);
+  bool metrics_enabled_ GUARDED_BY(crit_capture_);
+  SuppressionLevel suppression_level_ GUARDED_BY(crit_capture_);
+  int stream_drift_samples_ GUARDED_BY(crit_capture_);
+  bool was_stream_drift_set_ GUARDED_BY(crit_capture_);
+  bool stream_has_echo_ GUARDED_BY(crit_capture_);
+  bool delay_logging_enabled_ GUARDED_BY(crit_capture_);
+  bool extended_filter_enabled_ GUARDED_BY(crit_capture_);
+  bool delay_agnostic_enabled_ GUARDED_BY(crit_capture_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+  std::vector<float> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<float> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  rtc::scoped_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
+      render_signal_queue_;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc b/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
index b2e11981fa..7f152bf942 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
@@ -14,7 +14,6 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
index 954aac7d4a..f2df5f7984 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -13,9 +13,8 @@
 #include <assert.h>
 #include <string.h>
 
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
@@ -56,6 +55,12 @@ AudioProcessing::Error MapError(int err) {
       return AudioProcessing::kUnspecifiedError;
   }
 }
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
 }  // namespace
 
 size_t EchoControlMobile::echo_path_size_bytes() {
@@ -63,13 +68,20 @@ size_t EchoControlMobile::echo_path_size_bytes() {
 }
 
 EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessing* apm,
-                                             CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    routing_mode_(kSpeakerphone),
-    comfort_noise_enabled_(true),
-    external_echo_path_(NULL) {}
+                                             rtc::CriticalSection* crit_render,
+                                             rtc::CriticalSection* crit_capture)
+    : ProcessingComponent(),
+      apm_(apm),
+      crit_render_(crit_render),
+      crit_capture_(crit_capture),
+      routing_mode_(kSpeakerphone),
+      comfort_noise_enabled_(true),
+      external_echo_path_(NULL),
+      render_queue_element_max_size_(0) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
+}
 
 EchoControlMobileImpl::~EchoControlMobileImpl() {
     if (external_echo_path_ != NULL) {
@@ -79,53 +91,98 @@ EchoControlMobileImpl::~EchoControlMobileImpl() {
 }
 
 int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) {
+  rtc::CritScope cs_render(crit_render_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_reverse_channels());
 
-  int err = apm_->kNoError;
-
+  int err = AudioProcessing::kNoError;
   // The ordering convention must be followed to pass to the correct AECM.
   size_t handle_index = 0;
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
+  render_queue_buffer_.clear();
+  for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+    for (size_t j = 0; j < audio->num_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAecm_BufferFarend(
-          my_handle,
-          audio->split_bands_const(j)[kBand0To8kHz],
+      err = WebRtcAecm_GetBufferFarendError(
+          my_handle, audio->split_bands_const(j)[kBand0To8kHz],
           audio->num_frames_per_band());
 
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
+      if (err != AudioProcessing::kNoError)
+        return MapError(err);  // TODO(ajm): warning possible?);
+
+      // Buffer the samples in the render queue.
+      render_queue_buffer_.insert(render_queue_buffer_.end(),
+                                  audio->split_bands_const(j)[kBand0To8kHz],
+                                  (audio->split_bands_const(j)[kBand0To8kHz] +
+                                   audio->num_frames_per_band()));
 
       handle_index++;
     }
   }
 
-  return apm_->kNoError;
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    ReadQueuedRenderData();
+
+    // Retry the insert (should always work).
+    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+// Read chunks of data that were received and queued on the render side from
+// a queue. All the data chunks are buffered into the farend signal of the AEC.
+void EchoControlMobileImpl::ReadQueuedRenderData() {
+  rtc::CritScope cs_capture(crit_capture_);
+
+  if (!is_component_enabled()) {
+    return;
+  }
+
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+    size_t handle_index = 0;
+    size_t buffer_index = 0;
+    const size_t num_frames_per_band =
+        capture_queue_buffer_.size() /
+        (apm_->num_output_channels() * apm_->num_reverse_channels());
+    for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+      for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
+        Handle* my_handle = static_cast<Handle*>(handle(handle_index));
+        WebRtcAecm_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
+                                num_frames_per_band);
+
+        buffer_index += num_frames_per_band;
+        handle_index++;
+      }
+    }
+  }
 }
 
 int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs_capture(crit_capture_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_output_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AECM.
   size_t handle_index = 0;
-  for (int i = 0; i < audio->num_channels(); i++) {
+  for (size_t i = 0; i < audio->num_channels(); i++) {
     // TODO(ajm): improve how this works, possibly inside AECM.
     //            This is kind of hacked up.
     const int16_t* noisy = audio->low_pass_reference(i);
@@ -134,7 +191,7 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
       noisy = clean;
       clean = NULL;
     }
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+    for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
       err = WebRtcAecm_Process(
           my_handle,
@@ -144,109 +201,158 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
           audio->num_frames_per_band(),
           apm_->stream_delay_ms());
 
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
+      if (err != AudioProcessing::kNoError)
+        return MapError(err);
 
       handle_index++;
     }
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoControlMobileImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
   // Ensure AEC and AECM are not both enabled.
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+  // The is_enabled call is safe from a deadlock perspective
+  // as both locks are allready held in the correct order.
   if (enable && apm_->echo_cancellation()->is_enabled()) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   return EnableComponent(enable);
 }
 
 bool EchoControlMobileImpl::is_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }
 
 int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
-  CriticalSectionScoped crit_scoped(crit_);
   if (MapSetting(mode) == -1) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
-  routing_mode_ = mode;
+  {
+    rtc::CritScope cs(crit_capture_);
+    routing_mode_ = mode;
+  }
   return Configure();
 }
 
 EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode()
     const {
+  rtc::CritScope cs(crit_capture_);
   return routing_mode_;
 }
 
 int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  comfort_noise_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    comfort_noise_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return comfort_noise_enabled_;
 }
 
 int EchoControlMobileImpl::SetEchoPath(const void* echo_path,
                                        size_t size_bytes) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (echo_path == NULL) {
-    return apm_->kNullPointerError;
-  }
-  if (size_bytes != echo_path_size_bytes()) {
-    // Size mismatch
-    return apm_->kBadParameterError;
-  }
+  {
+    rtc::CritScope cs_render(crit_render_);
+    rtc::CritScope cs_capture(crit_capture_);
+    if (echo_path == NULL) {
+      return AudioProcessing::kNullPointerError;
+    }
+    if (size_bytes != echo_path_size_bytes()) {
+      // Size mismatch
+      return AudioProcessing::kBadParameterError;
+    }
 
-  if (external_echo_path_ == NULL) {
-    external_echo_path_ = new unsigned char[size_bytes];
+    if (external_echo_path_ == NULL) {
+      external_echo_path_ = new unsigned char[size_bytes];
+    }
+    memcpy(external_echo_path_, echo_path, size_bytes);
   }
-  memcpy(external_echo_path_, echo_path, size_bytes);
 
   return Initialize();
 }
 
 int EchoControlMobileImpl::GetEchoPath(void* echo_path,
                                        size_t size_bytes) const {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (echo_path == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
   if (size_bytes != echo_path_size_bytes()) {
     // Size mismatch
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
   if (!is_component_enabled()) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   // Get the echo path from the first channel
   Handle* my_handle = static_cast<Handle*>(handle(0));
-  if (WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes) != 0) {
-      return GetHandleError(my_handle);
-  }
+  int32_t err = WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes);
+  if (err != 0)
+    return MapError(err);
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoControlMobileImpl::Initialize() {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+  {
+    rtc::CritScope cs_capture(crit_capture_);
+    if (!is_component_enabled()) {
+      return AudioProcessing::kNoError;
+    }
   }
 
-  if (apm_->proc_sample_rate_hz() > apm_->kSampleRate16kHz) {
+  if (apm_->proc_sample_rate_hz() > AudioProcessing::kSampleRate16kHz) {
     LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates";
-    return apm_->kBadSampleRateError;
+    return AudioProcessing::kBadSampleRateError;
   }
 
-  return ProcessingComponent::Initialize();
+  int err = ProcessingComponent::Initialize();
+  if (err != AudioProcessing::kNoError) {
+    return err;
+  }
+
+  AllocateRenderQueue();
+
+  return AudioProcessing::kNoError;
+}
+
+void EchoControlMobileImpl::AllocateRenderQueue() {
+  const size_t new_render_queue_element_max_size = std::max<size_t>(
+      static_cast<size_t>(1),
+      kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
+
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
+  // Reallocate the queue if the queue item size is too small to fit the
+  // data to put in the queue.
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
+
+    std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
+
+    render_signal_queue_.reset(
+        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
+  } else {
+    render_signal_queue_->Clear();
+  }
 }
 
 void* EchoControlMobileImpl::CreateHandle() const {
@@ -254,10 +360,14 @@ void* EchoControlMobileImpl::CreateHandle() const {
 }
 
 void EchoControlMobileImpl::DestroyHandle(void* handle) const {
+  // This method is only called in a non-concurrent manner during APM
+  // destruction.
   WebRtcAecm_Free(static_cast<Handle*>(handle));
 }
 
 int EchoControlMobileImpl::InitializeHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   assert(handle != NULL);
   Handle* my_handle = static_cast<Handle*>(handle);
   if (WebRtcAecm_Init(my_handle, apm_->proc_sample_rate_hz()) != 0) {
@@ -271,10 +381,12 @@ int EchoControlMobileImpl::InitializeHandle(void* handle) const {
     }
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   AecmConfig config;
   config.cngMode = comfort_noise_enabled_;
   config.echoMode = MapSetting(routing_mode_);
@@ -282,13 +394,14 @@ int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
   return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
 }
 
-int EchoControlMobileImpl::num_handles_required() const {
-  return apm_->num_output_channels() *
-         apm_->num_reverse_channels();
+size_t EchoControlMobileImpl::num_handles_required() const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+  return apm_->num_output_channels() * apm_->num_reverse_channels();
 }
 
 int EchoControlMobileImpl::GetHandleError(void* handle) const {
+  // Not locked as it does not rely on anything in the state.
   assert(handle != NULL);
-  return MapError(WebRtcAecm_get_error_code(static_cast<Handle*>(handle)));
+  return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.h b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
index da7022545f..4d6529d3ac 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.h
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
@@ -11,19 +11,23 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
 class EchoControlMobileImpl : public EchoControlMobile,
                               public ProcessingComponent {
  public:
   EchoControlMobileImpl(const AudioProcessing* apm,
-                        CriticalSectionWrapper* crit);
+                        rtc::CriticalSection* crit_render,
+                        rtc::CriticalSection* crit_capture);
+
   virtual ~EchoControlMobileImpl();
 
   int ProcessRenderAudio(const AudioBuffer* audio);
@@ -37,6 +41,9 @@ class EchoControlMobileImpl : public EchoControlMobile,
   // ProcessingComponent implementation.
   int Initialize() override;
 
+  // Reads render side data that has been queued on the render call.
+  void ReadQueuedRenderData();
+
  private:
   // EchoControlMobile implementation.
   int Enable(bool enable) override;
@@ -46,18 +53,37 @@ class EchoControlMobileImpl : public EchoControlMobile,
   int GetEchoPath(void* echo_path, size_t size_bytes) const override;
 
   // ProcessingComponent implementation.
+  // Called holding both the render and capture locks.
   void* CreateHandle() const override;
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
+  size_t num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  void AllocateRenderQueue();
+
+  // Not guarded as its public API is thread safe.
   const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  RoutingMode routing_mode_;
-  bool comfort_noise_enabled_;
-  unsigned char* external_echo_path_;
+
+  rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  rtc::CriticalSection* const crit_capture_;
+
+  RoutingMode routing_mode_ GUARDED_BY(crit_capture_);
+  bool comfort_noise_enabled_ GUARDED_BY(crit_capture_);
+  unsigned char* external_echo_path_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+
+  std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  rtc::scoped_ptr<
+      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+      render_signal_queue_;
 };
 }  // namespace webrtc
 
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 3b1537e796..04a6c7ba29 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -14,7 +14,6 @@
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -33,60 +32,113 @@ int16_t MapSetting(GainControl::Mode mode) {
   assert(false);
   return -1;
 }
+
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
+
 }  // namespace
 
 GainControlImpl::GainControlImpl(const AudioProcessing* apm,
-                                 CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    mode_(kAdaptiveAnalog),
-    minimum_capture_level_(0),
-    maximum_capture_level_(255),
-    limiter_enabled_(true),
-    target_level_dbfs_(3),
-    compression_gain_db_(9),
-    analog_capture_level_(0),
-    was_analog_level_set_(false),
-    stream_is_saturated_(false) {}
+                                 rtc::CriticalSection* crit_render,
+                                 rtc::CriticalSection* crit_capture)
+    : ProcessingComponent(),
+      apm_(apm),
+      crit_render_(crit_render),
+      crit_capture_(crit_capture),
+      mode_(kAdaptiveAnalog),
+      minimum_capture_level_(0),
+      maximum_capture_level_(255),
+      limiter_enabled_(true),
+      target_level_dbfs_(3),
+      compression_gain_db_(9),
+      analog_capture_level_(0),
+      was_analog_level_set_(false),
+      stream_is_saturated_(false),
+      render_queue_element_max_size_(0) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
+}
 
 GainControlImpl::~GainControlImpl() {}
 
 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_render_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
 
-  for (int i = 0; i < num_handles(); i++) {
+  render_queue_buffer_.resize(0);
+  for (size_t i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
-    int err = WebRtcAgc_AddFarend(
-        my_handle,
-        audio->mixed_low_pass_data(),
-        audio->num_frames_per_band());
+    int err =
+        WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());
 
-    if (err != apm_->kNoError) {
+    if (err != AudioProcessing::kNoError)
       return GetHandleError(my_handle);
-    }
+
+    // Buffer the samples in the render queue.
+    render_queue_buffer_.insert(
+        render_queue_buffer_.end(), audio->mixed_low_pass_data(),
+        (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
+  }
+
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    ReadQueuedRenderData();
+
+    // Retry the insert (should always work).
+    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+// Read chunks of data that were received and queued on the render side from
+// a queue. All the data chunks are buffered into the farend signal of the AGC.
+void GainControlImpl::ReadQueuedRenderData() {
+  rtc::CritScope cs(crit_capture_);
+
+  if (!is_component_enabled()) {
+    return;
   }
 
-  return apm_->kNoError;
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+    size_t buffer_index = 0;
+    const size_t num_frames_per_band =
+        capture_queue_buffer_.size() / num_handles();
+    for (size_t i = 0; i < num_handles(); i++) {
+      Handle* my_handle = static_cast<Handle*>(handle(i));
+      WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
+                          num_frames_per_band);
+
+      buffer_index += num_frames_per_band;
+    }
+  }
 }
 
 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_capture_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   if (mode_ == kAdaptiveAnalog) {
     capture_levels_.assign(num_handles(), analog_capture_level_);
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       err = WebRtcAgc_AddMic(
           my_handle,
@@ -94,13 +146,13 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
           audio->num_bands(),
           audio->num_frames_per_band());
 
-      if (err != apm_->kNoError) {
+      if (err != AudioProcessing::kNoError) {
         return GetHandleError(my_handle);
       }
     }
   } else if (mode_ == kAdaptiveDigital) {
 
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       int32_t capture_level_out = 0;
 
@@ -114,34 +166,38 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
 
       capture_levels_[i] = capture_level_out;
 
-      if (err != apm_->kNoError) {
+      if (err != AudioProcessing::kNoError) {
         return GetHandleError(my_handle);
       }
 
     }
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_capture_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());
 
   stream_is_saturated_ = false;
-  for (int i = 0; i < num_handles(); i++) {
+  for (size_t i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int32_t capture_level_out = 0;
     uint8_t saturation_warning = 0;
 
+    // The call to stream_has_echo() is ok from a deadlock perspective
+    // as the capture lock is allready held.
     int err = WebRtcAgc_Process(
         my_handle,
         audio->split_bands_const(i),
@@ -153,7 +209,7 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
         apm_->echo_cancellation()->stream_has_echo(),
         &saturation_warning);
 
-    if (err != apm_->kNoError) {
+    if (err != AudioProcessing::kNoError) {
       return GetHandleError(my_handle);
     }
 
@@ -166,7 +222,7 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   if (mode_ == kAdaptiveAnalog) {
     // Take the analog level to be the average across the handles.
     analog_capture_level_ = 0;
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       analog_capture_level_ += capture_levels_[i];
     }
 
@@ -174,22 +230,24 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   }
 
   was_analog_level_set_ = false;
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
+
   was_analog_level_set_ = true;
   if (level < minimum_capture_level_ || level > maximum_capture_level_) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
   analog_capture_level_ = level;
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int GainControlImpl::stream_analog_level() {
+  rtc::CritScope cs(crit_capture_);
   // TODO(ajm): enable this assertion?
   //assert(mode_ == kAdaptiveAnalog);
 
@@ -197,18 +255,21 @@ int GainControlImpl::stream_analog_level() {
 }
 
 int GainControlImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   return EnableComponent(enable);
 }
 
 bool GainControlImpl::is_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }
 
 int GainControlImpl::set_mode(Mode mode) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   if (MapSetting(mode) == -1) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   mode_ = mode;
@@ -216,22 +277,23 @@ int GainControlImpl::set_mode(Mode mode) {
 }
 
 GainControl::Mode GainControlImpl::mode() const {
+  rtc::CritScope cs(crit_capture_);
   return mode_;
 }
 
 int GainControlImpl::set_analog_level_limits(int minimum,
                                              int maximum) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (minimum < 0) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   if (maximum > 65535) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   if (maximum < minimum) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   minimum_capture_level_ = minimum;
@@ -241,21 +303,24 @@ int GainControlImpl::set_analog_level_limits(int minimum,
 }
 
 int GainControlImpl::analog_level_minimum() const {
+  rtc::CritScope cs(crit_capture_);
   return minimum_capture_level_;
 }
 
 int GainControlImpl::analog_level_maximum() const {
+  rtc::CritScope cs(crit_capture_);
   return maximum_capture_level_;
 }
 
 bool GainControlImpl::stream_is_saturated() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_is_saturated_;
 }
 
 int GainControlImpl::set_target_level_dbfs(int level) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (level > 31 || level < 0) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   target_level_dbfs_ = level;
@@ -263,13 +328,14 @@ int GainControlImpl::set_target_level_dbfs(int level) {
 }
 
 int GainControlImpl::target_level_dbfs() const {
+  rtc::CritScope cs(crit_capture_);
   return target_level_dbfs_;
 }
 
 int GainControlImpl::set_compression_gain_db(int gain) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (gain < 0 || gain > 90) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   compression_gain_db_ = gain;
@@ -277,27 +343,59 @@ int GainControlImpl::set_compression_gain_db(int gain) {
 }
 
 int GainControlImpl::compression_gain_db() const {
+  rtc::CritScope cs(crit_capture_);
   return compression_gain_db_;
 }
 
 int GainControlImpl::enable_limiter(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   limiter_enabled_ = enable;
   return Configure();
 }
 
 bool GainControlImpl::is_limiter_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return limiter_enabled_;
 }
 
 int GainControlImpl::Initialize() {
   int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
+  if (err != AudioProcessing::kNoError || !is_component_enabled()) {
     return err;
   }
 
-  capture_levels_.assign(num_handles(), analog_capture_level_);
-  return apm_->kNoError;
+  AllocateRenderQueue();
+
+  rtc::CritScope cs_capture(crit_capture_);
+  const int n = num_handles();
+  RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;
+
+  capture_levels_.assign(n, analog_capture_level_);
+  return AudioProcessing::kNoError;
+}
+
+void GainControlImpl::AllocateRenderQueue() {
+  const size_t new_render_queue_element_max_size =
+      std::max<size_t>(static_cast<size_t>(1),
+                       kMaxAllowedValuesOfSamplesPerFrame * num_handles());
+
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
+    std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
+
+    render_signal_queue_.reset(
+        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
+  } else {
+    render_signal_queue_->Clear();
+  }
 }
 
 void* GainControlImpl::CreateHandle() const {
@@ -309,6 +407,9 @@ void GainControlImpl::DestroyHandle(void* handle) const {
 }
 
 int GainControlImpl::InitializeHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
   return WebRtcAgc_Init(static_cast<Handle*>(handle),
                           minimum_capture_level_,
                           maximum_capture_level_,
@@ -317,6 +418,8 @@ int GainControlImpl::InitializeHandle(void* handle) const {
 }
 
 int GainControlImpl::ConfigureHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   WebRtcAgcConfig config;
   // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
   //            change the interface.
@@ -330,14 +433,15 @@ int GainControlImpl::ConfigureHandle(void* handle) const {
   return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
 }
 
-int GainControlImpl::num_handles_required() const {
-  return apm_->num_output_channels();
+size_t GainControlImpl::num_handles_required() const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+  return apm_->num_proc_channels();
 }
 
 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
-  return apm_->kUnspecifiedError;
+  return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h
index f24d200cf2..72789ba5e1 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/webrtc/modules/audio_processing/gain_control_impl.h
@@ -13,19 +13,23 @@
 
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
 class GainControlImpl : public GainControl,
                         public ProcessingComponent {
  public:
   GainControlImpl(const AudioProcessing* apm,
-                  CriticalSectionWrapper* crit);
+                  rtc::CriticalSection* crit_render,
+                  rtc::CriticalSection* crit_capture);
   virtual ~GainControlImpl();
 
   int ProcessRenderAudio(AudioBuffer* audio);
@@ -41,6 +45,9 @@ class GainControlImpl : public GainControl,
   bool is_limiter_enabled() const override;
   Mode mode() const override;
 
+  // Reads render side data that has been queued on the render call.
+  void ReadQueuedRenderData();
+
  private:
   // GainControl implementation.
   int Enable(bool enable) override;
@@ -61,21 +68,37 @@ class GainControlImpl : public GainControl,
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
+  size_t num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  void AllocateRenderQueue();
+
+  // Not guarded as its public API is thread safe.
   const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  Mode mode_;
-  int minimum_capture_level_;
-  int maximum_capture_level_;
-  bool limiter_enabled_;
-  int target_level_dbfs_;
-  int compression_gain_db_;
-  std::vector<int> capture_levels_;
-  int analog_capture_level_;
-  bool was_analog_level_set_;
-  bool stream_is_saturated_;
+
+  rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  rtc::CriticalSection* const crit_capture_;
+
+  Mode mode_ GUARDED_BY(crit_capture_);
+  int minimum_capture_level_ GUARDED_BY(crit_capture_);
+  int maximum_capture_level_ GUARDED_BY(crit_capture_);
+  bool limiter_enabled_ GUARDED_BY(crit_capture_);
+  int target_level_dbfs_ GUARDED_BY(crit_capture_);
+  int compression_gain_db_ GUARDED_BY(crit_capture_);
+  std::vector<int> capture_levels_ GUARDED_BY(crit_capture_);
+  int analog_capture_level_ GUARDED_BY(crit_capture_);
+  bool was_analog_level_set_ GUARDED_BY(crit_capture_);
+  bool stream_is_saturated_ GUARDED_BY(crit_capture_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+  std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  rtc::scoped_ptr<
+      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+      render_signal_queue_;
 };
 }  // namespace webrtc
 
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
index 29e482078e..375d58febb 100644
--- a/webrtc/modules/audio_processing/high_pass_filter_impl.cc
+++ b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
@@ -10,159 +10,125 @@
 
 #include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
 
-#include <assert.h>
-
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/typedefs.h"
-
 
 namespace webrtc {
 namespace {
-const int16_t kFilterCoefficients8kHz[5] =
-    {3798, -7596, 3798, 7807, -3733};
-
-const int16_t kFilterCoefficients[5] =
-    {4012, -8024, 4012, 8002, -3913};
-
-struct FilterState {
-  int16_t y[4];
-  int16_t x[2];
-  const int16_t* ba;
-};
-
-int InitializeFilter(FilterState* hpf, int sample_rate_hz) {
-  assert(hpf != NULL);
+const int16_t kFilterCoefficients8kHz[5] = {3798, -7596, 3798, 7807, -3733};
+const int16_t kFilterCoefficients[5] = {4012, -8024, 4012, 8002, -3913};
+}  // namespace
 
-  if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) {
-    hpf->ba = kFilterCoefficients8kHz;
-  } else {
-    hpf->ba = kFilterCoefficients;
+class HighPassFilterImpl::BiquadFilter {
+ public:
+  explicit BiquadFilter(int sample_rate_hz) :
+      ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz ?
+          kFilterCoefficients8kHz : kFilterCoefficients)
+  {
+    Reset();
   }
 
-  WebRtcSpl_MemSetW16(hpf->x, 0, 2);
-  WebRtcSpl_MemSetW16(hpf->y, 0, 4);
-
-  return AudioProcessing::kNoError;
-}
-
-int Filter(FilterState* hpf, int16_t* data, size_t length) {
-  assert(hpf != NULL);
-
-  int32_t tmp_int32 = 0;
-  int16_t* y = hpf->y;
-  int16_t* x = hpf->x;
-  const int16_t* ba = hpf->ba;
-
-  for (size_t i = 0; i < length; i++) {
-    //  y[i] = b[0] * x[i] + b[1] * x[i-1] + b[2] * x[i-2]
-    //         + -a[1] * y[i-1] + -a[2] * y[i-2];
-
-    tmp_int32 = y[1] * ba[3];  // -a[1] * y[i-1] (low part)
-    tmp_int32 += y[3] * ba[4];  // -a[2] * y[i-2] (low part)
-    tmp_int32 = (tmp_int32 >> 15);
-    tmp_int32 += y[0] * ba[3];  // -a[1] * y[i-1] (high part)
-    tmp_int32 += y[2] * ba[4];  // -a[2] * y[i-2] (high part)
-    tmp_int32 = (tmp_int32 << 1);
-
-    tmp_int32 += data[i] * ba[0];  // b[0]*x[0]
-    tmp_int32 += x[0] * ba[1];  // b[1]*x[i-1]
-    tmp_int32 += x[1] * ba[2];  // b[2]*x[i-2]
-
-    // Update state (input part)
-    x[1] = x[0];
-    x[0] = data[i];
-
-    // Update state (filtered part)
-    y[2] = y[0];
-    y[3] = y[1];
-    y[0] = static_cast<int16_t>(tmp_int32 >> 13);
-    y[1] = static_cast<int16_t>(
-        (tmp_int32 - (static_cast<int32_t>(y[0]) << 13)) << 2);
-
-    // Rounding in Q12, i.e. add 2^11
-    tmp_int32 += 2048;
-
-    // Saturate (to 2^27) so that the HP filtered signal does not overflow
-    tmp_int32 = WEBRTC_SPL_SAT(static_cast<int32_t>(134217727),
-                               tmp_int32,
-                               static_cast<int32_t>(-134217728));
-
-    // Convert back to Q0 and use rounding.
-    data[i] = (int16_t)(tmp_int32 >> 12);
+  void Reset() {
+    std::memset(x_, 0, sizeof(x_));
+    std::memset(y_, 0, sizeof(y_));
   }
 
-  return AudioProcessing::kNoError;
-}
-}  // namespace
+  void Process(int16_t* data, size_t length) {
+    const int16_t* const ba = ba_;
+    int16_t* x = x_;
+    int16_t* y = y_;
+    int32_t tmp_int32 = 0;
+
+    for (size_t i = 0; i < length; i++) {
+      //  y[i] = b[0] * x[i] +  b[1] * x[i-1] +  b[2] * x[i-2]
+      //                     + -a[1] * y[i-1] + -a[2] * y[i-2];
+
+      tmp_int32 = y[1] * ba[3];      // -a[1] * y[i-1] (low part)
+      tmp_int32 += y[3] * ba[4];     // -a[2] * y[i-2] (low part)
+      tmp_int32 = (tmp_int32 >> 15);
+      tmp_int32 += y[0] * ba[3];     // -a[1] * y[i-1] (high part)
+      tmp_int32 += y[2] * ba[4];     // -a[2] * y[i-2] (high part)
+      tmp_int32 = (tmp_int32 << 1);
+
+      tmp_int32 += data[i] * ba[0];  // b[0] * x[0]
+      tmp_int32 += x[0] * ba[1];     // b[1] * x[i-1]
+      tmp_int32 += x[1] * ba[2];     // b[2] * x[i-2]
+
+      // Update state (input part).
+      x[1] = x[0];
+      x[0] = data[i];
+
+      // Update state (filtered part).
+      y[2] = y[0];
+      y[3] = y[1];
+      y[0] = static_cast<int16_t>(tmp_int32 >> 13);
+      y[1] = static_cast<int16_t>(
+          (tmp_int32 - (static_cast<int32_t>(y[0]) << 13)) << 2);
+
+      // Rounding in Q12, i.e. add 2^11.
+      tmp_int32 += 2048;
+
+      // Saturate (to 2^27) so that the HP filtered signal does not overflow.
+      tmp_int32 = WEBRTC_SPL_SAT(static_cast<int32_t>(134217727),
+                                 tmp_int32,
+                                 static_cast<int32_t>(-134217728));
+
+      // Convert back to Q0 and use rounding.
+      data[i] = static_cast<int16_t>(tmp_int32 >> 12);
+    }
+  }
 
-typedef FilterState Handle;
+ private:
+  const int16_t* const ba_ = nullptr;
+  int16_t x_[2];
+  int16_t y_[4];
+};
 
-HighPassFilterImpl::HighPassFilterImpl(const AudioProcessing* apm,
-                                       CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit) {}
+HighPassFilterImpl::HighPassFilterImpl(rtc::CriticalSection* crit)
+    : crit_(crit) {
+  RTC_DCHECK(crit_);
+}
 
 HighPassFilterImpl::~HighPassFilterImpl() {}
 
-int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  int err = apm_->kNoError;
-
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+void HighPassFilterImpl::Initialize(size_t channels, int sample_rate_hz) {
+  std::vector<rtc::scoped_ptr<BiquadFilter>> new_filters(channels);
+  for (size_t i = 0; i < channels; i++) {
+    new_filters[i].reset(new BiquadFilter(sample_rate_hz));
   }
+  rtc::CritScope cs(crit_);
+  filters_.swap(new_filters);
+}
 
-  assert(audio->num_frames_per_band() <= 160);
-
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-    err = Filter(my_handle,
-                 audio->split_bands(i)[kBand0To8kHz],
-                 audio->num_frames_per_band());
-
-    if (err != apm_->kNoError) {
-      return GetHandleError(my_handle);
-    }
+void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
   }
 
-  return apm_->kNoError;
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
+  for (size_t i = 0; i < filters_.size(); i++) {
+    filters_[i]->Process(audio->split_bands(i)[kBand0To8kHz],
+                         audio->num_frames_per_band());
+  }
 }
 
 int HighPassFilterImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (!enabled_ && enable) {
+    for (auto& filter : filters_) {
+      filter->Reset();
+    }
+  }
+  enabled_ = enable;
+  return AudioProcessing::kNoError;
 }
 
 bool HighPassFilterImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-void* HighPassFilterImpl::CreateHandle() const {
-  return new FilterState;
-}
-
-void HighPassFilterImpl::DestroyHandle(void* handle) const {
-  delete static_cast<Handle*>(handle);
-}
-
-int HighPassFilterImpl::InitializeHandle(void* handle) const {
-  return InitializeFilter(static_cast<Handle*>(handle),
-                          apm_->proc_sample_rate_hz());
-}
-
-int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const {
-  return apm_->kNoError; // Not configurable.
-}
-
-int HighPassFilterImpl::num_handles_required() const {
-  return apm_->num_output_channels();
-}
-
-int HighPassFilterImpl::GetHandleError(void* handle) const {
-  // The component has no detailed errors.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.h b/webrtc/modules/audio_processing/high_pass_filter_impl.h
index 90b393e903..0e985bac7a 100644
--- a/webrtc/modules/audio_processing/high_pass_filter_impl.h
+++ b/webrtc/modules/audio_processing/high_pass_filter_impl.h
@@ -11,39 +11,34 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
-class HighPassFilterImpl : public HighPassFilter,
-                           public ProcessingComponent {
+class HighPassFilterImpl : public HighPassFilter {
  public:
-  HighPassFilterImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit);
-  virtual ~HighPassFilterImpl();
+  explicit HighPassFilterImpl(rtc::CriticalSection* crit);
+  ~HighPassFilterImpl() override;
 
-  int ProcessCaptureAudio(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize(size_t channels, int sample_rate_hz);
+  void ProcessCaptureAudio(AudioBuffer* audio);
 
   // HighPassFilter implementation.
+  int Enable(bool enable) override;
   bool is_enabled() const override;
 
  private:
-  // HighPassFilter implementation.
-  int Enable(bool enable) override;
-
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
+  class BiquadFilter;
+  rtc::CriticalSection* const crit_ = nullptr;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  std::vector<rtc::scoped_ptr<BiquadFilter>> filters_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
 };
 }  // namespace webrtc
 
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index c8ddc6a483..9a3a4b32d5 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -65,6 +65,7 @@ class VoiceDetection;
 struct ExtendedFilter {
   ExtendedFilter() : enabled(false) {}
   explicit ExtendedFilter(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kExtendedFilter;
   bool enabled;
 };
 
@@ -76,6 +77,7 @@ struct ExtendedFilter {
 struct DelayAgnostic {
   DelayAgnostic() : enabled(false) {}
   explicit DelayAgnostic(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kDelayAgnostic;
   bool enabled;
 };
 
@@ -96,6 +98,7 @@ struct ExperimentalAgc {
       : enabled(enabled), startup_min_volume(kAgcStartupMinVolume) {}
   ExperimentalAgc(bool enabled, int startup_min_volume)
       : enabled(enabled), startup_min_volume(startup_min_volume) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kExperimentalAgc;
   bool enabled;
   int startup_min_volume;
 };
@@ -105,6 +108,7 @@ struct ExperimentalAgc {
 struct ExperimentalNs {
   ExperimentalNs() : enabled(false) {}
   explicit ExperimentalNs(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kExperimentalNs;
   bool enabled;
 };
 
@@ -127,6 +131,7 @@ struct Beamforming {
       : enabled(enabled),
         array_geometry(array_geometry),
         target_direction(target_direction) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kBeamforming;
   const bool enabled;
   const std::vector<Point> array_geometry;
   const SphericalPointf target_direction;
@@ -141,6 +146,7 @@ struct Beamforming {
 struct Intelligibility {
   Intelligibility() : enabled(false) {}
   explicit Intelligibility(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kIntelligibility;
   bool enabled;
 };
 
@@ -279,13 +285,18 @@ class AudioProcessing {
   // ensures the options are applied immediately.
   virtual void SetExtraOptions(const Config& config) = 0;
 
+  // TODO(peah): Remove after voice engine no longer requires it to resample
+  // the reverse stream to the forward rate.
+  virtual int input_sample_rate_hz() const = 0;
+
   // TODO(ajm): Only intended for internal use. Make private and friend the
   // necessary classes?
   virtual int proc_sample_rate_hz() const = 0;
   virtual int proc_split_sample_rate_hz() const = 0;
-  virtual int num_input_channels() const = 0;
-  virtual int num_output_channels() const = 0;
-  virtual int num_reverse_channels() const = 0;
+  virtual size_t num_input_channels() const = 0;
+  virtual size_t num_proc_channels() const = 0;
+  virtual size_t num_output_channels() const = 0;
+  virtual size_t num_reverse_channels() const = 0;
 
   // Set to true when the output of AudioProcessing will be muted or in some
   // other way not used. Ideally, the captured audio would still be processed,
@@ -497,7 +508,7 @@ class StreamConfig {
   //               is true, the last channel in any corresponding list of
   //               channels is the keyboard channel.
   StreamConfig(int sample_rate_hz = 0,
-               int num_channels = 0,
+               size_t num_channels = 0,
                bool has_keyboard = false)
       : sample_rate_hz_(sample_rate_hz),
         num_channels_(num_channels),
@@ -508,14 +519,14 @@ class StreamConfig {
     sample_rate_hz_ = value;
     num_frames_ = calculate_frames(value);
   }
-  void set_num_channels(int value) { num_channels_ = value; }
+  void set_num_channels(size_t value) { num_channels_ = value; }
   void set_has_keyboard(bool value) { has_keyboard_ = value; }
 
   int sample_rate_hz() const { return sample_rate_hz_; }
 
   // The number of channels in the stream, not including the keyboard channel if
   // present.
-  int num_channels() const { return num_channels_; }
+  size_t num_channels() const { return num_channels_; }
 
   bool has_keyboard() const { return has_keyboard_; }
   size_t num_frames() const { return num_frames_; }
@@ -536,7 +547,7 @@ class StreamConfig {
   }
 
   int sample_rate_hz_;
-  int num_channels_;
+  size_t num_channels_;
   bool has_keyboard_;
   size_t num_frames_;
 };
diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
index 4ff52baf1c..9e1f2d5861 100644
--- a/webrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -201,11 +201,11 @@ class MockAudioProcessing : public AudioProcessing {
   MOCK_CONST_METHOD0(proc_split_sample_rate_hz,
       int());
   MOCK_CONST_METHOD0(num_input_channels,
-      int());
+      size_t());
   MOCK_CONST_METHOD0(num_output_channels,
-      int());
+      size_t());
   MOCK_CONST_METHOD0(num_reverse_channels,
-      int());
+      size_t());
   MOCK_METHOD1(set_output_will_be_muted,
       void(bool muted));
   MOCK_CONST_METHOD0(output_will_be_muted,
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
index d014ce060c..fe964aba8c 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
@@ -54,12 +54,12 @@ IntelligibilityEnhancer::TransformCallback::TransformCallback(
 
 void IntelligibilityEnhancer::TransformCallback::ProcessAudioBlock(
     const complex<float>* const* in_block,
-    int in_channels,
+    size_t in_channels,
     size_t frames,
-    int /* out_channels */,
+    size_t /* out_channels */,
     complex<float>* const* out_block) {
   RTC_DCHECK_EQ(parent_->freqs_, frames);
-  for (int i = 0; i < in_channels; ++i) {
+  for (size_t i = 0; i < in_channels; ++i) {
     parent_->DispatchAudio(source_, in_block[i], out_block[i]);
   }
 }
@@ -129,7 +129,7 @@ IntelligibilityEnhancer::IntelligibilityEnhancer(const Config& config)
 
 void IntelligibilityEnhancer::ProcessRenderAudio(float* const* audio,
                                                  int sample_rate_hz,
-                                                 int num_channels) {
+                                                 size_t num_channels) {
   RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
   RTC_CHECK_EQ(num_render_channels_, num_channels);
 
@@ -138,7 +138,7 @@ void IntelligibilityEnhancer::ProcessRenderAudio(float* const* audio,
   }
 
   if (active_) {
-    for (int i = 0; i < num_render_channels_; ++i) {
+    for (size_t i = 0; i < num_render_channels_; ++i) {
       memcpy(audio[i], temp_render_out_buffer_.channels()[i],
              chunk_length_ * sizeof(**audio));
     }
@@ -147,7 +147,7 @@ void IntelligibilityEnhancer::ProcessRenderAudio(float* const* audio,
 
 void IntelligibilityEnhancer::AnalyzeCaptureAudio(float* const* audio,
                                                   int sample_rate_hz,
-                                                  int num_channels) {
+                                                  size_t num_channels) {
   RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
   RTC_CHECK_EQ(num_capture_channels_, num_channels);
 
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
index 1e9e35ac2a..1eb22342ad 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -47,8 +47,8 @@ class IntelligibilityEnhancer {
           gain_change_limit(0.1f),
           rho(0.02f) {}
     int sample_rate_hz;
-    int num_capture_channels;
-    int num_render_channels;
+    size_t num_capture_channels;
+    size_t num_render_channels;
     intelligibility::VarianceArray::StepType var_type;
     float var_decay_rate;
     size_t var_window_size;
@@ -63,12 +63,12 @@ class IntelligibilityEnhancer {
   // Reads and processes chunk of noise stream in time domain.
   void AnalyzeCaptureAudio(float* const* audio,
                            int sample_rate_hz,
-                           int num_channels);
+                           size_t num_channels);
 
   // Reads chunk of speech in time domain and updates with modified signal.
   void ProcessRenderAudio(float* const* audio,
                           int sample_rate_hz,
-                          int num_channels);
+                          size_t num_channels);
   bool active() const;
 
  private:
@@ -85,9 +85,9 @@ class IntelligibilityEnhancer {
     // All in frequency domain, receives input |in_block|, applies
     // intelligibility enhancement, and writes result to |out_block|.
     void ProcessAudioBlock(const std::complex<float>* const* in_block,
-                           int in_channels,
+                           size_t in_channels,
                            size_t frames,
-                           int out_channels,
+                           size_t out_channels,
                            std::complex<float>* const* out_block) override;
 
    private:
@@ -144,8 +144,8 @@ class IntelligibilityEnhancer {
   const size_t bank_size_;     // Num ERB filters.
   const int sample_rate_hz_;
   const int erb_resolution_;
-  const int num_capture_channels_;
-  const int num_render_channels_;
+  const size_t num_capture_channels_;
+  const size_t num_render_channels_;
   const int analysis_rate_;    // Num blocks before gains recalculated.
 
   const bool active_;          // Whether render gains are being updated.
diff --git a/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc b/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
index 27d0ab48bb..4d2f5f4c5d 100644
--- a/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
+++ b/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
@@ -68,7 +68,7 @@ DEFINE_string(out_file,
               "Enhanced output. Use '-' to "
               "play through aplay immediately.");
 
-const int kNumChannels = 1;
+const size_t kNumChannels = 1;
 
 // void function for gtest
 void void_main(int argc, char* argv[]) {
diff --git a/webrtc/modules/audio_processing/level_estimator_impl.cc b/webrtc/modules/audio_processing/level_estimator_impl.cc
index 35fe697c2d..187873e33e 100644
--- a/webrtc/modules/audio_processing/level_estimator_impl.cc
+++ b/webrtc/modules/audio_processing/level_estimator_impl.cc
@@ -11,76 +11,55 @@
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
-LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessing* apm,
-                                       CriticalSectionWrapper* crit)
-    : ProcessingComponent(),
-      crit_(crit) {}
+LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit)
+    : crit_(crit), rms_(new RMSLevel()) {
+  RTC_DCHECK(crit);
+}
 
 LevelEstimatorImpl::~LevelEstimatorImpl() {}
 
-int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return AudioProcessing::kNoError;
-  }
+void LevelEstimatorImpl::Initialize() {
+  rtc::CritScope cs(crit_);
+  rms_->Reset();
+}
 
-  RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
-  for (int i = 0; i < audio->num_channels(); ++i) {
-    rms_level->Process(audio->channels_const()[i],
-                       audio->num_frames());
+void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
   }
 
-  return AudioProcessing::kNoError;
+  for (size_t i = 0; i < audio->num_channels(); i++) {
+    rms_->Process(audio->channels_const()[i], audio->num_frames());
+  }
 }
 
 int LevelEstimatorImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (enable && !enabled_) {
+    rms_->Reset();
+  }
+  enabled_ = enable;
+  return AudioProcessing::kNoError;
 }
 
 bool LevelEstimatorImpl::is_enabled() const {
-  return is_component_enabled();
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 
 int LevelEstimatorImpl::RMS() {
-  if (!is_component_enabled()) {
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
     return AudioProcessing::kNotEnabledError;
   }
 
-  RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
-  return rms_level->RMS();
-}
-
-// The ProcessingComponent implementation is pretty weird in this class since
-// we have only a single instance of the trivial underlying component.
-void* LevelEstimatorImpl::CreateHandle() const {
-  return new RMSLevel;
-}
-
-void LevelEstimatorImpl::DestroyHandle(void* handle) const {
-  delete static_cast<RMSLevel*>(handle);
+  return rms_->RMS();
 }
-
-int LevelEstimatorImpl::InitializeHandle(void* handle) const {
-  static_cast<RMSLevel*>(handle)->Reset();
-  return AudioProcessing::kNoError;
-}
-
-int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
-  return AudioProcessing::kNoError;
-}
-
-int LevelEstimatorImpl::num_handles_required() const {
-  return 1;
-}
-
-int LevelEstimatorImpl::GetHandleError(void* /*handle*/) const {
-  return AudioProcessing::kUnspecifiedError;
-}
-
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/level_estimator_impl.h b/webrtc/modules/audio_processing/level_estimator_impl.h
index 0d0050c7e7..4401da37e4 100644
--- a/webrtc/modules/audio_processing/level_estimator_impl.h
+++ b/webrtc/modules/audio_processing/level_estimator_impl.h
@@ -11,43 +11,36 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
-#include "webrtc/modules/audio_processing/rms_level.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
+class RMSLevel;
 
-class LevelEstimatorImpl : public LevelEstimator,
-                           public ProcessingComponent {
+class LevelEstimatorImpl : public LevelEstimator {
  public:
-  LevelEstimatorImpl(const AudioProcessing* apm,
-                     CriticalSectionWrapper* crit);
-  virtual ~LevelEstimatorImpl();
+  explicit LevelEstimatorImpl(rtc::CriticalSection* crit);
+  ~LevelEstimatorImpl() override;
 
-  int ProcessStream(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize();
+  void ProcessStream(AudioBuffer* audio);
 
   // LevelEstimator implementation.
-  bool is_enabled() const override;
-
- private:
-  // LevelEstimator implementation.
   int Enable(bool enable) override;
+  bool is_enabled() const override;
   int RMS() override;
 
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  CriticalSectionWrapper* crit_;
+ private:
+  rtc::CriticalSection* const crit_ = nullptr;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  rtc::scoped_ptr<RMSLevel> rms_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LevelEstimatorImpl);
 };
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
diff --git a/webrtc/modules/audio_processing/logging/aec_logging.h b/webrtc/modules/audio_processing/logging/aec_logging.h
index 3cf9ff89ed..b062913be2 100644
--- a/webrtc/modules/audio_processing/logging/aec_logging.h
+++ b/webrtc/modules/audio_processing/logging/aec_logging.h
@@ -43,6 +43,20 @@
     (void) fwrite(data, data_size, 1, file);           \
   } while (0)
 
+// Dumps a raw scalar int32 to file.
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_INT32(file, data)             \
+  do {                                                               \
+    int32_t value_to_store = data;                                   \
+    (void) fwrite(&value_to_store, sizeof(value_to_store), 1, file); \
+  } while (0)
+
+// Dumps a raw scalar double to file.
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_DOUBLE(file, data)            \
+  do {                                                               \
+    double value_to_store = data;                                    \
+    (void) fwrite(&value_to_store, sizeof(value_to_store), 1, file); \
+  } while (0)
+
 // Opens a raw data file for writing using the specified sample rate.
 #define RTC_AEC_DEBUG_RAW_OPEN(name, instance_counter, file) \
   do {                                                       \
@@ -73,6 +87,14 @@
   do {                                                 \
   } while (0)
 
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_INT32(file, data) \
+  do {                                                   \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_DOUBLE(file, data) \
+  do {                                                    \
+  } while (0)
+
 #define RTC_AEC_DEBUG_RAW_OPEN(file, name, instance_counter) \
   do {                                                       \
   } while (0)
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.cc b/webrtc/modules/audio_processing/noise_suppression_impl.cc
index 65ec3c445e..de7e856676 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -10,172 +10,166 @@
 
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 
-#include <assert.h>
-
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #if defined(WEBRTC_NS_FLOAT)
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
+#define NS_CREATE WebRtcNs_Create
+#define NS_FREE WebRtcNs_Free
+#define NS_INIT WebRtcNs_Init
+#define NS_SET_POLICY WebRtcNs_set_policy
+typedef NsHandle NsState;
 #elif defined(WEBRTC_NS_FIXED)
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
+#define NS_CREATE WebRtcNsx_Create
+#define NS_FREE WebRtcNsx_Free
+#define NS_INIT WebRtcNsx_Init
+#define NS_SET_POLICY WebRtcNsx_set_policy
+typedef NsxHandle NsState;
 #endif
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-
 
 namespace webrtc {
-
-#if defined(WEBRTC_NS_FLOAT)
-typedef NsHandle Handle;
-#elif defined(WEBRTC_NS_FIXED)
-typedef NsxHandle Handle;
-#endif
-
-namespace {
-int MapSetting(NoiseSuppression::Level level) {
-  switch (level) {
-    case NoiseSuppression::kLow:
-      return 0;
-    case NoiseSuppression::kModerate:
-      return 1;
-    case NoiseSuppression::kHigh:
-      return 2;
-    case NoiseSuppression::kVeryHigh:
-      return 3;
+class NoiseSuppressionImpl::Suppressor {
+ public:
+  explicit Suppressor(int sample_rate_hz) {
+    state_ = NS_CREATE();
+    RTC_CHECK(state_);
+    int error = NS_INIT(state_, sample_rate_hz);
+    RTC_DCHECK_EQ(0, error);
+  }
+  ~Suppressor() {
+    NS_FREE(state_);
   }
-  assert(false);
-  return -1;
+  NsState* state() { return state_; }
+ private:
+  NsState* state_ = nullptr;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Suppressor);
+};
+
+NoiseSuppressionImpl::NoiseSuppressionImpl(rtc::CriticalSection* crit)
+    : crit_(crit) {
+  RTC_DCHECK(crit);
 }
-}  // namespace
-
-NoiseSuppressionImpl::NoiseSuppressionImpl(const AudioProcessing* apm,
-                                           CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    level_(kModerate) {}
 
 NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
 
-int NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
-#if defined(WEBRTC_NS_FLOAT)
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
+  rtc::CritScope cs(crit_);
+  channels_ = channels;
+  sample_rate_hz_ = sample_rate_hz;
+  std::vector<rtc::scoped_ptr<Suppressor>> new_suppressors;
+  if (enabled_) {
+    new_suppressors.resize(channels);
+    for (size_t i = 0; i < channels; i++) {
+      new_suppressors[i].reset(new Suppressor(sample_rate_hz));
+    }
   }
-  assert(audio->num_frames_per_band() <= 160);
-  assert(audio->num_channels() == num_handles());
+  suppressors_.swap(new_suppressors);
+  set_level(level_);
+}
 
-  for (int i = 0; i < num_handles(); ++i) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
+void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+#if defined(WEBRTC_NS_FLOAT)
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
+  }
 
-    WebRtcNs_Analyze(my_handle, audio->split_bands_const_f(i)[kBand0To8kHz]);
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
+  for (size_t i = 0; i < suppressors_.size(); i++) {
+    WebRtcNs_Analyze(suppressors_[i]->state(),
+                     audio->split_bands_const_f(i)[kBand0To8kHz]);
   }
 #endif
-  return apm_->kNoError;
 }
 
-int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
   }
-  assert(audio->num_frames_per_band() <= 160);
-  assert(audio->num_channels() == num_handles());
 
-  for (int i = 0; i < num_handles(); ++i) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
+  for (size_t i = 0; i < suppressors_.size(); i++) {
 #if defined(WEBRTC_NS_FLOAT)
-    WebRtcNs_Process(my_handle,
+    WebRtcNs_Process(suppressors_[i]->state(),
                      audio->split_bands_const_f(i),
                      audio->num_bands(),
                      audio->split_bands_f(i));
 #elif defined(WEBRTC_NS_FIXED)
-    WebRtcNsx_Process(my_handle,
+    WebRtcNsx_Process(suppressors_[i]->state(),
                       audio->split_bands_const(i),
                       audio->num_bands(),
                       audio->split_bands(i));
 #endif
   }
-  return apm_->kNoError;
 }
 
 int NoiseSuppressionImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (enabled_ != enable) {
+    enabled_ = enable;
+    Initialize(channels_, sample_rate_hz_);
+  }
+  return AudioProcessing::kNoError;
 }
 
 bool NoiseSuppressionImpl::is_enabled() const {
-  return is_component_enabled();
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 
 int NoiseSuppressionImpl::set_level(Level level) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
+  int policy = 1;
+  switch (level) {
+    case NoiseSuppression::kLow:
+      policy = 0;
+      break;
+    case NoiseSuppression::kModerate:
+      policy = 1;
+      break;
+    case NoiseSuppression::kHigh:
+      policy = 2;
+      break;
+    case NoiseSuppression::kVeryHigh:
+      policy = 3;
+      break;
+    default:
+      RTC_NOTREACHED();
   }
-
+  rtc::CritScope cs(crit_);
   level_ = level;
-  return Configure();
+  for (auto& suppressor : suppressors_) {
+    int error = NS_SET_POLICY(suppressor->state(), policy);
+    RTC_DCHECK_EQ(0, error);
+  }
+  return AudioProcessing::kNoError;
 }
 
 NoiseSuppression::Level NoiseSuppressionImpl::level() const {
+  rtc::CritScope cs(crit_);
   return level_;
 }
 
 float NoiseSuppressionImpl::speech_probability() const {
+  rtc::CritScope cs(crit_);
 #if defined(WEBRTC_NS_FLOAT)
   float probability_average = 0.0f;
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-    probability_average += WebRtcNs_prior_speech_probability(my_handle);
+  for (auto& suppressor : suppressors_) {
+    probability_average +=
+        WebRtcNs_prior_speech_probability(suppressor->state());
   }
-  return probability_average / num_handles();
+  if (!suppressors_.empty()) {
+    probability_average /= suppressors_.size();
+  }
+  return probability_average;
 #elif defined(WEBRTC_NS_FIXED)
+  // TODO(peah): Returning error code as a float! Remove this.
   // Currently not available for the fixed point implementation.
-  return apm_->kUnsupportedFunctionError;
+  return AudioProcessing::kUnsupportedFunctionError;
 #endif
 }
-
-void* NoiseSuppressionImpl::CreateHandle() const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_Create();
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_Create();
-#endif
-}
-
-void NoiseSuppressionImpl::DestroyHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  WebRtcNs_Free(static_cast<Handle*>(handle));
-#elif defined(WEBRTC_NS_FIXED)
-  WebRtcNsx_Free(static_cast<Handle*>(handle));
-#endif
-}
-
-int NoiseSuppressionImpl::InitializeHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_Init(static_cast<Handle*>(handle),
-                       apm_->proc_sample_rate_hz());
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_Init(static_cast<Handle*>(handle),
-                        apm_->proc_sample_rate_hz());
-#endif
-}
-
-int NoiseSuppressionImpl::ConfigureHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_set_policy(static_cast<Handle*>(handle),
-                             MapSetting(level_));
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_set_policy(static_cast<Handle*>(handle),
-                              MapSetting(level_));
-#endif
-}
-
-int NoiseSuppressionImpl::num_handles_required() const {
-  return apm_->num_output_channels();
-}
-
-int NoiseSuppressionImpl::GetHandleError(void* handle) const {
-  // The NS has no get_error() function.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.h b/webrtc/modules/audio_processing/noise_suppression_impl.h
index 76a39b8e09..debbc61bc9 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.h
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.h
@@ -11,47 +11,42 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
-class NoiseSuppressionImpl : public NoiseSuppression,
-                             public ProcessingComponent {
+class NoiseSuppressionImpl : public NoiseSuppression {
  public:
-  NoiseSuppressionImpl(const AudioProcessing* apm,
-                       CriticalSectionWrapper* crit);
-  virtual ~NoiseSuppressionImpl();
+  explicit NoiseSuppressionImpl(rtc::CriticalSection* crit);
+  ~NoiseSuppressionImpl() override;
 
-  int AnalyzeCaptureAudio(AudioBuffer* audio);
-  int ProcessCaptureAudio(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize(size_t channels, int sample_rate_hz);
+  void AnalyzeCaptureAudio(AudioBuffer* audio);
+  void ProcessCaptureAudio(AudioBuffer* audio);
 
   // NoiseSuppression implementation.
+  int Enable(bool enable) override;
   bool is_enabled() const override;
-  float speech_probability() const override;
+  int set_level(Level level) override;
   Level level() const override;
+  float speech_probability() const override;
 
  private:
-  // NoiseSuppression implementation.
-  int Enable(bool enable) override;
-  int set_level(Level level) override;
-
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  Level level_;
+  class Suppressor;
+  rtc::CriticalSection* const crit_;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  Level level_ GUARDED_BY(crit_) = kModerate;
+  size_t channels_ GUARDED_BY(crit_) = 0;
+  int sample_rate_hz_ GUARDED_BY(crit_) = 0;
+  std::vector<rtc::scoped_ptr<Suppressor>> suppressors_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSuppressionImpl);
 };
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
diff --git a/webrtc/modules/audio_processing/ns/noise_suppression.c b/webrtc/modules/audio_processing/ns/noise_suppression.c
index 13f1b2d6dc..dd05e0ab3d 100644
--- a/webrtc/modules/audio_processing/ns/noise_suppression.c
+++ b/webrtc/modules/audio_processing/ns/noise_suppression.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
 
 #include <stdlib.h>
 #include <string.h>
diff --git a/webrtc/modules/audio_processing/ns/include/noise_suppression.h b/webrtc/modules/audio_processing/ns/noise_suppression.h
index 9dac56bdee..8018118b60 100644
--- a/webrtc/modules/audio_processing/ns/include/noise_suppression.h
+++ b/webrtc/modules/audio_processing/ns/noise_suppression.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_H_
 
 #include <stddef.h>
 
@@ -113,4 +113,4 @@ float WebRtcNs_prior_speech_probability(NsHandle* handle);
 }
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_H_
diff --git a/webrtc/modules/audio_processing/ns/noise_suppression_x.c b/webrtc/modules/audio_processing/ns/noise_suppression_x.c
index 150fe608dd..0a5ba13300 100644
--- a/webrtc/modules/audio_processing/ns/noise_suppression_x.c
+++ b/webrtc/modules/audio_processing/ns/noise_suppression_x.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 
 #include <stdlib.h>
 
diff --git a/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h b/webrtc/modules/audio_processing/ns/noise_suppression_x.h
index 88fe4cd635..02b44cc091 100644
--- a/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
+++ b/webrtc/modules/audio_processing/ns/noise_suppression_x.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_X_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_X_H_
 
 #include "webrtc/typedefs.h"
 
@@ -85,4 +85,4 @@ void WebRtcNsx_Process(NsxHandle* nsxInst,
 }
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_X_H_
diff --git a/webrtc/modules/audio_processing/ns/ns_core.c b/webrtc/modules/audio_processing/ns/ns_core.c
index 1d6091400e..5ce64cee29 100644
--- a/webrtc/modules/audio_processing/ns/ns_core.c
+++ b/webrtc/modules/audio_processing/ns/ns_core.c
@@ -15,7 +15,7 @@
 
 #include "webrtc/common_audio/fft4g.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
 #include "webrtc/modules/audio_processing/ns/ns_core.h"
 #include "webrtc/modules/audio_processing/ns/windows_private.h"
 
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index 71445792f5..25f16d26ab 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 
 #include <assert.h>
 #include <math.h>
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_c.c b/webrtc/modules/audio_processing/ns/nsx_core_c.c
index 14322d38cc..da7aa3d5db 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_c.c
@@ -10,7 +10,7 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/modules/audio_processing/ns/nsx_defines.h"
 
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_mips.c b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
index d99be8720b..7688d82d78 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_mips.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
@@ -11,7 +11,7 @@
 #include <assert.h>
 #include <string.h>
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 
 static const int16_t kIndicatorTable[17] = {
diff --git a/webrtc/modules/audio_processing/processing_component.cc b/webrtc/modules/audio_processing/processing_component.cc
index 9e16d7c4ee..7abd8e2100 100644
--- a/webrtc/modules/audio_processing/processing_component.cc
+++ b/webrtc/modules/audio_processing/processing_component.cc
@@ -55,12 +55,12 @@ bool ProcessingComponent::is_component_enabled() const {
   return enabled_;
 }
 
-void* ProcessingComponent::handle(int index) const {
+void* ProcessingComponent::handle(size_t index) const {
   assert(index < num_handles_);
   return handles_[index];
 }
 
-int ProcessingComponent::num_handles() const {
+size_t ProcessingComponent::num_handles() const {
   return num_handles_;
 }
 
@@ -70,12 +70,12 @@ int ProcessingComponent::Initialize() {
   }
 
   num_handles_ = num_handles_required();
-  if (num_handles_ > static_cast<int>(handles_.size())) {
+  if (num_handles_ > handles_.size()) {
     handles_.resize(num_handles_, NULL);
   }
 
-  assert(static_cast<int>(handles_.size()) >= num_handles_);
-  for (int i = 0; i < num_handles_; i++) {
+  assert(handles_.size() >= num_handles_);
+  for (size_t i = 0; i < num_handles_; i++) {
     if (handles_[i] == NULL) {
       handles_[i] = CreateHandle();
       if (handles_[i] == NULL) {
@@ -98,8 +98,8 @@ int ProcessingComponent::Configure() {
     return AudioProcessing::kNoError;
   }
 
-  assert(static_cast<int>(handles_.size()) >= num_handles_);
-  for (int i = 0; i < num_handles_; i++) {
+  assert(handles_.size() >= num_handles_);
+  for (size_t i = 0; i < num_handles_; i++) {
     int err = ConfigureHandle(handles_[i]);
     if (err != AudioProcessing::kNoError) {
       return GetHandleError(handles_[i]);
diff --git a/webrtc/modules/audio_processing/processing_component.h b/webrtc/modules/audio_processing/processing_component.h
index 8ee3ac6c7d..577f1570ad 100644
--- a/webrtc/modules/audio_processing/processing_component.h
+++ b/webrtc/modules/audio_processing/processing_component.h
@@ -17,6 +17,22 @@
 
 namespace webrtc {
 
+// Functor to use when supplying a verifier function for the queue item
+// verifcation.
+template <typename T>
+class RenderQueueItemVerifier {
+ public:
+  explicit RenderQueueItemVerifier(size_t minimum_capacity)
+      : minimum_capacity_(minimum_capacity) {}
+
+  bool operator()(const std::vector<T>& v) const {
+    return v.capacity() >= minimum_capacity_;
+  }
+
+ private:
+  size_t minimum_capacity_;
+};
+
 class ProcessingComponent {
  public:
   ProcessingComponent();
@@ -31,21 +47,21 @@ class ProcessingComponent {
  protected:
   virtual int Configure();
   int EnableComponent(bool enable);
-  void* handle(int index) const;
-  int num_handles() const;
+  void* handle(size_t index) const;
+  size_t num_handles() const;
 
  private:
   virtual void* CreateHandle() const = 0;
   virtual int InitializeHandle(void* handle) const = 0;
   virtual int ConfigureHandle(void* handle) const = 0;
   virtual void DestroyHandle(void* handle) const = 0;
-  virtual int num_handles_required() const = 0;
+  virtual size_t num_handles_required() const = 0;
   virtual int GetHandleError(void* handle) const = 0;
 
   std::vector<void*> handles_;
   bool initialized_;
   bool enabled_;
-  int num_handles_;
+  size_t num_handles_;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/splitting_filter.cc b/webrtc/modules/audio_processing/splitting_filter.cc
index 60427e2db6..46cc9352c2 100644
--- a/webrtc/modules/audio_processing/splitting_filter.cc
+++ b/webrtc/modules/audio_processing/splitting_filter.cc
@@ -16,7 +16,7 @@
 
 namespace webrtc {
 
-SplittingFilter::SplittingFilter(int num_channels,
+SplittingFilter::SplittingFilter(size_t num_channels,
                                  size_t num_bands,
                                  size_t num_frames)
     : num_bands_(num_bands) {
@@ -24,7 +24,7 @@ SplittingFilter::SplittingFilter(int num_channels,
   if (num_bands_ == 2) {
     two_bands_states_.resize(num_channels);
   } else if (num_bands_ == 3) {
-    for (int i = 0; i < num_channels; ++i) {
+    for (size_t i = 0; i < num_channels; ++i) {
       three_band_filter_banks_.push_back(new ThreeBandFilterBank(num_frames));
     }
   }
@@ -58,8 +58,7 @@ void SplittingFilter::Synthesis(const IFChannelBuffer* bands,
 
 void SplittingFilter::TwoBandsAnalysis(const IFChannelBuffer* data,
                                        IFChannelBuffer* bands) {
-  RTC_DCHECK_EQ(static_cast<int>(two_bands_states_.size()),
-                data->num_channels());
+  RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
   for (size_t i = 0; i < two_bands_states_.size(); ++i) {
     WebRtcSpl_AnalysisQMF(data->ibuf_const()->channels()[i],
                           data->num_frames(),
@@ -72,8 +71,7 @@ void SplittingFilter::TwoBandsAnalysis(const IFChannelBuffer* data,
 
 void SplittingFilter::TwoBandsSynthesis(const IFChannelBuffer* bands,
                                         IFChannelBuffer* data) {
-  RTC_DCHECK_EQ(static_cast<int>(two_bands_states_.size()),
-                data->num_channels());
+  RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
   for (size_t i = 0; i < two_bands_states_.size(); ++i) {
     WebRtcSpl_SynthesisQMF(bands->ibuf_const()->channels(0)[i],
                            bands->ibuf_const()->channels(1)[i],
@@ -86,8 +84,7 @@ void SplittingFilter::TwoBandsSynthesis(const IFChannelBuffer* bands,
 
 void SplittingFilter::ThreeBandsAnalysis(const IFChannelBuffer* data,
                                          IFChannelBuffer* bands) {
-  RTC_DCHECK_EQ(static_cast<int>(three_band_filter_banks_.size()),
-                data->num_channels());
+  RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
   for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
     three_band_filter_banks_[i]->Analysis(data->fbuf_const()->channels()[i],
                                           data->num_frames(),
@@ -97,8 +94,7 @@ void SplittingFilter::ThreeBandsAnalysis(const IFChannelBuffer* data,
 
 void SplittingFilter::ThreeBandsSynthesis(const IFChannelBuffer* bands,
                                           IFChannelBuffer* data) {
-  RTC_DCHECK_EQ(static_cast<int>(three_band_filter_banks_.size()),
-                data->num_channels());
+  RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
   for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
     three_band_filter_banks_[i]->Synthesis(bands->fbuf_const()->bands(i),
                                            bands->num_frames_per_band(),
diff --git a/webrtc/modules/audio_processing/splitting_filter.h b/webrtc/modules/audio_processing/splitting_filter.h
index 4698d3fe2b..6b81c2fb05 100644
--- a/webrtc/modules/audio_processing/splitting_filter.h
+++ b/webrtc/modules/audio_processing/splitting_filter.h
@@ -45,7 +45,7 @@ struct TwoBandsStates {
 // used.
 class SplittingFilter {
  public:
-  SplittingFilter(int num_channels, size_t num_bands, size_t num_frames);
+  SplittingFilter(size_t num_channels, size_t num_bands, size_t num_frames);
 
   void Analysis(const IFChannelBuffer* data, IFChannelBuffer* bands);
   void Synthesis(const IFChannelBuffer* bands, IFChannelBuffer* data);
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.cc b/webrtc/modules/audio_processing/test/audio_file_processor.cc
new file mode 100644
index 0000000000..56e9b4b96f
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.cc
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+
+using rtc::scoped_ptr;
+using rtc::CheckedDivExact;
+using std::vector;
+using webrtc::audioproc::Event;
+using webrtc::audioproc::Init;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+
+namespace webrtc {
+namespace {
+
+// Returns a StreamConfig corresponding to file.
+StreamConfig GetStreamConfig(const WavFile& file) {
+  return StreamConfig(file.sample_rate(), file.num_channels());
+}
+
+// Returns a ChannelBuffer corresponding to file.
+ChannelBuffer<float> GetChannelBuffer(const WavFile& file) {
+  return ChannelBuffer<float>(
+      CheckedDivExact(file.sample_rate(), AudioFileProcessor::kChunksPerSecond),
+      file.num_channels());
+}
+
+}  // namespace
+
+WavFileProcessor::WavFileProcessor(scoped_ptr<AudioProcessing> ap,
+                                   scoped_ptr<WavReader> in_file,
+                                   scoped_ptr<WavWriter> out_file)
+    : ap_(std::move(ap)),
+      in_buf_(GetChannelBuffer(*in_file)),
+      out_buf_(GetChannelBuffer(*out_file)),
+      input_config_(GetStreamConfig(*in_file)),
+      output_config_(GetStreamConfig(*out_file)),
+      buffer_reader_(std::move(in_file)),
+      buffer_writer_(std::move(out_file)) {}
+
+bool WavFileProcessor::ProcessChunk() {
+  if (!buffer_reader_.Read(&in_buf_)) {
+    return false;
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(kNoErr,
+                 ap_->ProcessStream(in_buf_.channels(), input_config_,
+                                    output_config_, out_buf_.channels()));
+  }
+  buffer_writer_.Write(out_buf_);
+  return true;
+}
+
+AecDumpFileProcessor::AecDumpFileProcessor(scoped_ptr<AudioProcessing> ap,
+                                           FILE* dump_file,
+                                           scoped_ptr<WavWriter> out_file)
+    : ap_(std::move(ap)),
+      dump_file_(dump_file),
+      out_buf_(GetChannelBuffer(*out_file)),
+      output_config_(GetStreamConfig(*out_file)),
+      buffer_writer_(std::move(out_file)) {
+  RTC_CHECK(dump_file_) << "Could not open dump file for reading.";
+}
+
+AecDumpFileProcessor::~AecDumpFileProcessor() {
+  fclose(dump_file_);
+}
+
+bool AecDumpFileProcessor::ProcessChunk() {
+  Event event_msg;
+
+  // Continue until we process our first Stream message.
+  do {
+    if (!ReadMessageFromFile(dump_file_, &event_msg)) {
+      return false;
+    }
+
+    if (event_msg.type() == Event::INIT) {
+      RTC_CHECK(event_msg.has_init());
+      HandleMessage(event_msg.init());
+
+    } else if (event_msg.type() == Event::STREAM) {
+      RTC_CHECK(event_msg.has_stream());
+      HandleMessage(event_msg.stream());
+
+    } else if (event_msg.type() == Event::REVERSE_STREAM) {
+      RTC_CHECK(event_msg.has_reverse_stream());
+      HandleMessage(event_msg.reverse_stream());
+    }
+  } while (event_msg.type() != Event::STREAM);
+
+  return true;
+}
+
+void AecDumpFileProcessor::HandleMessage(const Init& msg) {
+  RTC_CHECK(msg.has_sample_rate());
+  RTC_CHECK(msg.has_num_input_channels());
+  RTC_CHECK(msg.has_num_reverse_channels());
+
+  in_buf_.reset(new ChannelBuffer<float>(
+      CheckedDivExact(msg.sample_rate(), kChunksPerSecond),
+      msg.num_input_channels()));
+  const int reverse_sample_rate = msg.has_reverse_sample_rate()
+                                      ? msg.reverse_sample_rate()
+                                      : msg.sample_rate();
+  reverse_buf_.reset(new ChannelBuffer<float>(
+      CheckedDivExact(reverse_sample_rate, kChunksPerSecond),
+      msg.num_reverse_channels()));
+  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+  reverse_config_ =
+      StreamConfig(reverse_sample_rate, msg.num_reverse_channels());
+
+  const ProcessingConfig config = {
+      {input_config_, output_config_, reverse_config_, reverse_config_}};
+  RTC_CHECK_EQ(kNoErr, ap_->Initialize(config));
+}
+
+void AecDumpFileProcessor::HandleMessage(const Stream& msg) {
+  RTC_CHECK(!msg.has_input_data());
+  RTC_CHECK_EQ(in_buf_->num_channels(),
+               static_cast<size_t>(msg.input_channel_size()));
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+    RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                 msg.input_channel(i).size());
+    std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
+                msg.input_channel(i).size());
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(kNoErr, ap_->set_stream_delay_ms(msg.delay()));
+    ap_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+    if (msg.has_keypress()) {
+      ap_->set_stream_key_pressed(msg.keypress());
+    }
+    RTC_CHECK_EQ(kNoErr,
+                 ap_->ProcessStream(in_buf_->channels(), input_config_,
+                                    output_config_, out_buf_.channels()));
+  }
+
+  buffer_writer_.Write(out_buf_);
+}
+
+void AecDumpFileProcessor::HandleMessage(const ReverseStream& msg) {
+  RTC_CHECK(!msg.has_data());
+  RTC_CHECK_EQ(reverse_buf_->num_channels(),
+               static_cast<size_t>(msg.channel_size()));
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+    RTC_CHECK_EQ(reverse_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                 msg.channel(i).size());
+    std::memcpy(reverse_buf_->channels()[i], msg.channel(i).data(),
+                msg.channel(i).size());
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    // TODO(ajm): This currently discards the processed output, which is needed
+    // for e.g. intelligibility enhancement.
+    RTC_CHECK_EQ(kNoErr, ap_->ProcessReverseStream(
+                             reverse_buf_->channels(), reverse_config_,
+                             reverse_config_, reverse_buf_->channels()));
+  }
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.h b/webrtc/modules/audio_processing/test/audio_file_processor.h
new file mode 100644
index 0000000000..a3153b2244
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.h
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
+
+#include <algorithm>
+#include <limits>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+
+namespace webrtc {
+
+// Holds a few statistics about a series of TickIntervals.
+struct TickIntervalStats {
+  TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
+  TickInterval sum;
+  TickInterval max;
+  TickInterval min;
+};
+
+// Interface for processing an input file with an AudioProcessing instance and
+// dumping the results to an output file.
+class AudioFileProcessor {
+ public:
+  static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
+
+  virtual ~AudioFileProcessor() {}
+
+  // Processes one AudioProcessing::kChunkSizeMs of data from the input file and
+  // writes to the output file.
+  virtual bool ProcessChunk() = 0;
+
+  // Returns the execution time of all AudioProcessing calls.
+  const TickIntervalStats& proc_time() const { return proc_time_; }
+
+ protected:
+  // RAII class for execution time measurement. Updates the provided
+  // TickIntervalStats based on the time between ScopedTimer creation and
+  // leaving the enclosing scope.
+  class ScopedTimer {
+   public:
+    explicit ScopedTimer(TickIntervalStats* proc_time)
+        : proc_time_(proc_time), start_time_(TickTime::Now()) {}
+
+    ~ScopedTimer() {
+      TickInterval interval = TickTime::Now() - start_time_;
+      proc_time_->sum += interval;
+      proc_time_->max = std::max(proc_time_->max, interval);
+      proc_time_->min = std::min(proc_time_->min, interval);
+    }
+
+   private:
+    TickIntervalStats* const proc_time_;
+    TickTime start_time_;
+  };
+
+  TickIntervalStats* mutable_proc_time() { return &proc_time_; }
+
+ private:
+  TickIntervalStats proc_time_;
+};
+
+// Used to read from and write to WavFile objects.
+class WavFileProcessor final : public AudioFileProcessor {
+ public:
+  // Takes ownership of all parameters.
+  WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
+                   rtc::scoped_ptr<WavReader> in_file,
+                   rtc::scoped_ptr<WavWriter> out_file);
+  virtual ~WavFileProcessor() {}
+
+  // Processes one chunk from the WAV input and writes to the WAV output.
+  bool ProcessChunk() override;
+
+ private:
+  rtc::scoped_ptr<AudioProcessing> ap_;
+
+  ChannelBuffer<float> in_buf_;
+  ChannelBuffer<float> out_buf_;
+  const StreamConfig input_config_;
+  const StreamConfig output_config_;
+  ChannelBufferWavReader buffer_reader_;
+  ChannelBufferWavWriter buffer_writer_;
+};
+
+// Used to read from an aecdump file and write to a WavWriter.
+class AecDumpFileProcessor final : public AudioFileProcessor {
+ public:
+  // Takes ownership of all parameters.
+  AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
+                       FILE* dump_file,
+                       rtc::scoped_ptr<WavWriter> out_file);
+
+  virtual ~AecDumpFileProcessor();
+
+  // Processes messages from the aecdump file until the first Stream message is
+  // completed. Passes other data from the aecdump messages as appropriate.
+  bool ProcessChunk() override;
+
+ private:
+  void HandleMessage(const webrtc::audioproc::Init& msg);
+  void HandleMessage(const webrtc::audioproc::Stream& msg);
+  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
+
+  rtc::scoped_ptr<AudioProcessing> ap_;
+  FILE* dump_file_;
+
+  rtc::scoped_ptr<ChannelBuffer<float>> in_buf_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_;
+  ChannelBuffer<float> out_buf_;
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  const StreamConfig output_config_;
+  ChannelBufferWavWriter buffer_writer_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index c013a369fe..324cb7bec6 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -14,6 +14,7 @@
 #include <limits>
 #include <queue>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
@@ -23,11 +24,10 @@
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "gtest/gtest.h"
 #include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
@@ -53,11 +53,8 @@ namespace {
 // file. This is the typical case. When the file should be updated, it can
 // be set to true with the command-line switch --write_ref_data.
 bool write_ref_data = false;
-const int kChannels[] = {1, 2};
-const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
-
+const google::protobuf::int32 kChannels[] = {1, 2};
 const int kSampleRates[] = {8000, 16000, 32000, 48000};
-const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
 
 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
 // AECM doesn't support super-wb.
@@ -65,8 +62,6 @@ const int kProcessSampleRates[] = {8000, 16000};
 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
 const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
 #endif
-const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) /
-    sizeof(*kProcessSampleRates);
 
 enum StreamDirection { kForward = 0, kReverse };
 
@@ -77,7 +72,7 @@ void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
                cb->num_frames(),
                cb->num_channels(),
                cb_int.channels());
-  for (int i = 0; i < cb->num_channels(); ++i) {
+  for (size_t i = 0; i < cb->num_channels(); ++i) {
     S16ToFloat(cb_int.channels()[i],
                cb->num_frames(),
                cb->channels()[i]);
@@ -89,7 +84,7 @@ void ConvertToFloat(const AudioFrame& frame, ChannelBuffer<float>* cb) {
 }
 
 // Number of channels including the keyboard channel.
-int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+size_t TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
       return 1;
@@ -100,7 +95,7 @@ int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
       return 3;
   }
   assert(false);
-  return -1;
+  return 0;
 }
 
 int TruncateToMultipleOf10(int value) {
@@ -108,25 +103,25 @@ int TruncateToMultipleOf10(int value) {
 }
 
 void MixStereoToMono(const float* stereo, float* mono,
-                     int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; ++i)
+                     size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; ++i)
     mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2;
 }
 
 void MixStereoToMono(const int16_t* stereo, int16_t* mono,
-                     int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; ++i)
+                     size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; ++i)
     mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1;
 }
 
-void CopyLeftToRightChannel(int16_t* stereo, int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; i++) {
+void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     stereo[i * 2 + 1] = stereo[i * 2];
   }
 }
 
-void VerifyChannelsAreEqual(int16_t* stereo, int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; i++) {
+void VerifyChannelsAreEqual(int16_t* stereo, size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]);
   }
 }
@@ -139,7 +134,7 @@ void SetFrameTo(AudioFrame* frame, int16_t value) {
 }
 
 void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
-  ASSERT_EQ(2, frame->num_channels_);
+  ASSERT_EQ(2u, frame->num_channels_);
   for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     frame->data_[i] = left;
     frame->data_[i + 1] = right;
@@ -199,9 +194,9 @@ T AbsValue(T a) {
 }
 
 int16_t MaxAudioFrame(const AudioFrame& frame) {
-  const int length = frame.samples_per_channel_ * frame.num_channels_;
+  const size_t length = frame.samples_per_channel_ * frame.num_channels_;
   int16_t max_data = AbsValue(frame.data_[0]);
-  for (int i = 1; i < length; i++) {
+  for (size_t i = 1; i < length; i++) {
     max_data = std::max(max_data, AbsValue(frame.data_[i]));
   }
 
@@ -264,10 +259,10 @@ std::string OutputFilePath(std::string name,
                            int output_rate,
                            int reverse_input_rate,
                            int reverse_output_rate,
-                           int num_input_channels,
-                           int num_output_channels,
-                           int num_reverse_input_channels,
-                           int num_reverse_output_channels,
+                           size_t num_input_channels,
+                           size_t num_output_channels,
+                           size_t num_reverse_input_channels,
+                           size_t num_reverse_output_channels,
                            StreamDirection file_direction) {
   std::ostringstream ss;
   ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir"
@@ -362,9 +357,9 @@ class ApmTest : public ::testing::Test {
   void Init(int sample_rate_hz,
             int output_sample_rate_hz,
             int reverse_sample_rate_hz,
-            int num_input_channels,
-            int num_output_channels,
-            int num_reverse_channels,
+            size_t num_input_channels,
+            size_t num_output_channels,
+            size_t num_reverse_channels,
             bool open_output_file);
   void Init(AudioProcessing* ap);
   void EnableAllComponents();
@@ -377,12 +372,12 @@ class ApmTest : public ::testing::Test {
   void ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
                                     int delay_min, int delay_max);
   void TestChangingChannelsInt16Interface(
-      int num_channels,
+      size_t num_channels,
       AudioProcessing::Error expected_return);
-  void TestChangingForwardChannels(int num_in_channels,
-                                   int num_out_channels,
+  void TestChangingForwardChannels(size_t num_in_channels,
+                                   size_t num_out_channels,
                                    AudioProcessing::Error expected_return);
-  void TestChangingReverseChannels(int num_rev_channels,
+  void TestChangingReverseChannels(size_t num_rev_channels,
                                    AudioProcessing::Error expected_return);
   void RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate);
   void RunManualVolumeChangeIsPossibleTest(int sample_rate);
@@ -403,7 +398,7 @@ class ApmTest : public ::testing::Test {
   rtc::scoped_ptr<ChannelBuffer<float> > float_cb_;
   rtc::scoped_ptr<ChannelBuffer<float> > revfloat_cb_;
   int output_sample_rate_hz_;
-  int num_output_channels_;
+  size_t num_output_channels_;
   FILE* far_file_;
   FILE* near_file_;
   FILE* out_file_;
@@ -487,9 +482,9 @@ void ApmTest::Init(AudioProcessing* ap) {
 void ApmTest::Init(int sample_rate_hz,
                    int output_sample_rate_hz,
                    int reverse_sample_rate_hz,
-                   int num_input_channels,
-                   int num_output_channels,
-                   int num_reverse_channels,
+                   size_t num_input_channels,
+                   size_t num_output_channels,
+                   size_t num_reverse_channels,
                    bool open_output_file) {
   SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_);
   output_sample_rate_hz_ = output_sample_rate_hz;
@@ -821,7 +816,7 @@ TEST_F(ApmTest, DelayOffsetWithLimitsIsSetProperly) {
 }
 
 void ApmTest::TestChangingChannelsInt16Interface(
-    int num_channels,
+    size_t num_channels,
     AudioProcessing::Error expected_return) {
   frame_->num_channels_ = num_channels;
   EXPECT_EQ(expected_return, apm_->ProcessStream(frame_));
@@ -829,8 +824,8 @@ void ApmTest::TestChangingChannelsInt16Interface(
 }
 
 void ApmTest::TestChangingForwardChannels(
-    int num_in_channels,
-    int num_out_channels,
+    size_t num_in_channels,
+    size_t num_out_channels,
     AudioProcessing::Error expected_return) {
   const StreamConfig input_stream = {frame_->sample_rate_hz_, num_in_channels};
   const StreamConfig output_stream = {output_sample_rate_hz_, num_out_channels};
@@ -841,7 +836,7 @@ void ApmTest::TestChangingForwardChannels(
 }
 
 void ApmTest::TestChangingReverseChannels(
-    int num_rev_channels,
+    size_t num_rev_channels,
     AudioProcessing::Error expected_return) {
   const ProcessingConfig processing_config = {
       {{frame_->sample_rate_hz_, apm_->num_input_channels()},
@@ -862,11 +857,11 @@ TEST_F(ApmTest, ChannelsInt16Interface) {
 
   TestChangingChannelsInt16Interface(0, apm_->kBadNumberChannelsError);
 
-  for (int i = 1; i < 4; i++) {
+  for (size_t i = 1; i < 4; i++) {
     TestChangingChannelsInt16Interface(i, kNoErr);
     EXPECT_EQ(i, apm_->num_input_channels());
     // We always force the number of reverse channels used for processing to 1.
-    EXPECT_EQ(1, apm_->num_reverse_channels());
+    EXPECT_EQ(1u, apm_->num_reverse_channels());
   }
 }
 
@@ -877,8 +872,8 @@ TEST_F(ApmTest, Channels) {
   TestChangingForwardChannels(0, 1, apm_->kBadNumberChannelsError);
   TestChangingReverseChannels(0, apm_->kBadNumberChannelsError);
 
-  for (int i = 1; i < 4; ++i) {
-    for (int j = 0; j < 1; ++j) {
+  for (size_t i = 1; i < 4; ++i) {
+    for (size_t j = 0; j < 1; ++j) {
       // Output channels much be one or match input channels.
       if (j == 1 || i == j) {
         TestChangingForwardChannels(i, j, kNoErr);
@@ -887,7 +882,7 @@ TEST_F(ApmTest, Channels) {
         EXPECT_EQ(i, apm_->num_input_channels());
         EXPECT_EQ(j, apm_->num_output_channels());
         // The number of reverse channels used for processing to is always 1.
-        EXPECT_EQ(1, apm_->num_reverse_channels());
+        EXPECT_EQ(1u, apm_->num_reverse_channels());
       } else {
         TestChangingForwardChannels(i, j,
                                     AudioProcessing::kBadNumberChannelsError);
@@ -902,7 +897,7 @@ TEST_F(ApmTest, SampleRatesInt) {
   EXPECT_EQ(apm_->kBadSampleRateError, ProcessStreamChooser(kIntFormat));
   // Testing valid sample rates
   int fs[] = {8000, 16000, 32000, 48000};
-  for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
+  for (size_t i = 0; i < arraysize(fs); i++) {
     SetContainerFormat(fs[i], 2, frame_, &float_cb_);
     EXPECT_NOERR(ProcessStreamChooser(kIntFormat));
   }
@@ -921,7 +916,7 @@ TEST_F(ApmTest, EchoCancellation) {
     EchoCancellation::kModerateSuppression,
     EchoCancellation::kHighSuppression,
   };
-  for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+  for (size_t i = 0; i < arraysize(level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->echo_cancellation()->set_suppression_level(level[i]));
     EXPECT_EQ(level[i],
@@ -998,7 +993,7 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
   // Test a couple of corner cases and verify that the estimated delay is
   // within a valid region (set to +-1.5 blocks). Note that these cases are
   // sampling frequency dependent.
-  for (size_t i = 0; i < kProcessSampleRatesSize; i++) {
+  for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
     Init(kProcessSampleRates[i],
          kProcessSampleRates[i],
          kProcessSampleRates[i],
@@ -1070,7 +1065,7 @@ TEST_F(ApmTest, EchoControlMobile) {
       EchoControlMobile::kSpeakerphone,
       EchoControlMobile::kLoudSpeakerphone,
   };
-  for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+  for (size_t i = 0; i < arraysize(mode); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->echo_control_mobile()->set_routing_mode(mode[i]));
     EXPECT_EQ(mode[i],
@@ -1135,7 +1130,7 @@ TEST_F(ApmTest, GainControl) {
     GainControl::kAdaptiveDigital,
     GainControl::kFixedDigital
   };
-  for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+  for (size_t i = 0; i < arraysize(mode); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_mode(mode[i]));
     EXPECT_EQ(mode[i], apm_->gain_control()->mode());
@@ -1151,7 +1146,7 @@ TEST_F(ApmTest, GainControl) {
       apm_->gain_control()->target_level_dbfs()));
 
   int level_dbfs[] = {0, 6, 31};
-  for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) {
+  for (size_t i = 0; i < arraysize(level_dbfs); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
     EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
@@ -1169,7 +1164,7 @@ TEST_F(ApmTest, GainControl) {
       apm_->gain_control()->compression_gain_db()));
 
   int gain_db[] = {0, 10, 90};
-  for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) {
+  for (size_t i = 0; i < arraysize(gain_db); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_compression_gain_db(gain_db[i]));
     EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
@@ -1200,14 +1195,14 @@ TEST_F(ApmTest, GainControl) {
       apm_->gain_control()->analog_level_maximum()));
 
   int min_level[] = {0, 255, 1024};
-  for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+  for (size_t i = 0; i < arraysize(min_level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
     EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
   }
 
   int max_level[] = {0, 1024, 65535};
-  for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+  for (size_t i = 0; i < arraysize(min_level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
     EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
@@ -1246,7 +1241,7 @@ void ApmTest::RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate) {
 // Verifies that despite volume slider quantization, the AGC can continue to
 // increase its volume.
 TEST_F(ApmTest, QuantizedVolumeDoesNotGetStuck) {
-  for (size_t i = 0; i < kSampleRatesSize; ++i) {
+  for (size_t i = 0; i < arraysize(kSampleRates); ++i) {
     RunQuantizedVolumeDoesNotGetStuckTest(kSampleRates[i]);
   }
 }
@@ -1291,7 +1286,7 @@ void ApmTest::RunManualVolumeChangeIsPossibleTest(int sample_rate) {
 }
 
 TEST_F(ApmTest, ManualVolumeChangeIsPossible) {
-  for (size_t i = 0; i < kSampleRatesSize; ++i) {
+  for (size_t i = 0; i < arraysize(kSampleRates); ++i) {
     RunManualVolumeChangeIsPossibleTest(kSampleRates[i]);
   }
 }
@@ -1299,11 +1294,11 @@ TEST_F(ApmTest, ManualVolumeChangeIsPossible) {
 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
 TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
   const int kSampleRateHz = 16000;
-  const int kSamplesPerChannel =
-      AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000;
-  const int kNumInputChannels = 2;
-  const int kNumOutputChannels = 1;
-  const int kNumChunks = 700;
+  const size_t kSamplesPerChannel =
+      static_cast<size_t>(AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000);
+  const size_t kNumInputChannels = 2;
+  const size_t kNumOutputChannels = 1;
+  const size_t kNumChunks = 700;
   const float kScaleFactor = 0.25f;
   Config config;
   std::vector<webrtc::Point> geometry;
@@ -1317,8 +1312,8 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
   EXPECT_EQ(kNoErr, apm->gain_control()->Enable(true));
   ChannelBuffer<float> src_buf(kSamplesPerChannel, kNumInputChannels);
   ChannelBuffer<float> dest_buf(kSamplesPerChannel, kNumOutputChannels);
-  const int max_length = kSamplesPerChannel * std::max(kNumInputChannels,
-                                                       kNumOutputChannels);
+  const size_t max_length = kSamplesPerChannel * std::max(kNumInputChannels,
+                                                          kNumOutputChannels);
   rtc::scoped_ptr<int16_t[]> int_data(new int16_t[max_length]);
   rtc::scoped_ptr<float[]> float_data(new float[max_length]);
   std::string filename = ResourceFilePath("far", kSampleRateHz);
@@ -1330,13 +1325,13 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
   bool is_target = false;
   EXPECT_CALL(*beamformer, is_target_present())
       .WillRepeatedly(testing::ReturnPointee(&is_target));
-  for (int i = 0; i < kNumChunks; ++i) {
+  for (size_t i = 0; i < kNumChunks; ++i) {
     ASSERT_TRUE(ReadChunk(far_file,
                           int_data.get(),
                           float_data.get(),
                           &src_buf));
-    for (int j = 0; j < kNumInputChannels; ++j) {
-      for (int k = 0; k < kSamplesPerChannel; ++k) {
+    for (size_t j = 0; j < kNumInputChannels; ++j) {
+      for (size_t k = 0; k < kSamplesPerChannel; ++k) {
         src_buf.channels()[j][k] *= kScaleFactor;
       }
     }
@@ -1355,13 +1350,13 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
             apm->gain_control()->compression_gain_db());
   rewind(far_file);
   is_target = true;
-  for (int i = 0; i < kNumChunks; ++i) {
+  for (size_t i = 0; i < kNumChunks; ++i) {
     ASSERT_TRUE(ReadChunk(far_file,
                           int_data.get(),
                           float_data.get(),
                           &src_buf));
-    for (int j = 0; j < kNumInputChannels; ++j) {
-      for (int k = 0; k < kSamplesPerChannel; ++k) {
+    for (size_t j = 0; j < kNumInputChannels; ++j) {
+      for (size_t k = 0; k < kSamplesPerChannel; ++k) {
         src_buf.channels()[j][k] *= kScaleFactor;
       }
     }
@@ -1390,7 +1385,7 @@ TEST_F(ApmTest, NoiseSuppression) {
     NoiseSuppression::kHigh,
     NoiseSuppression::kVeryHigh
   };
-  for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+  for (size_t i = 0; i < arraysize(level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->noise_suppression()->set_level(level[i]));
     EXPECT_EQ(level[i], apm_->noise_suppression()->level());
@@ -1492,7 +1487,7 @@ TEST_F(ApmTest, VoiceDetection) {
       VoiceDetection::kModerateLikelihood,
       VoiceDetection::kHighLikelihood
   };
-  for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) {
+  for (size_t i = 0; i < arraysize(likelihood); i++) {
     EXPECT_EQ(apm_->kNoError,
               apm_->voice_detection()->set_likelihood(likelihood[i]));
     EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
@@ -1524,7 +1519,7 @@ TEST_F(ApmTest, VoiceDetection) {
       AudioFrame::kVadPassive,
       AudioFrame::kVadUnknown
   };
-  for (size_t i = 0; i < sizeof(activity)/sizeof(*activity); i++) {
+  for (size_t i = 0; i < arraysize(activity); i++) {
     frame_->vad_activity_ = activity[i];
     EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
     EXPECT_EQ(activity[i], frame_->vad_activity_);
@@ -1550,7 +1545,7 @@ TEST_F(ApmTest, AllProcessingDisabledByDefault) {
 }
 
 TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
-  for (size_t i = 0; i < kSampleRatesSize; i++) {
+  for (size_t i = 0; i < arraysize(kSampleRates); i++) {
     Init(kSampleRates[i], kSampleRates[i], kSampleRates[i], 2, 2, 2, false);
     SetFrameTo(frame_, 1000, 2000);
     AudioFrame frame_copy;
@@ -1602,7 +1597,7 @@ TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledFloat) {
 TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) {
   EnableAllComponents();
 
-  for (size_t i = 0; i < kProcessSampleRatesSize; i++) {
+  for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
     Init(kProcessSampleRates[i],
          kProcessSampleRates[i],
          kProcessSampleRates[i],
@@ -1751,7 +1746,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
       const audioproc::ReverseStream msg = event_msg.reverse_stream();
 
       if (msg.channel_size() > 0) {
-        ASSERT_EQ(revframe_->num_channels_, msg.channel_size());
+        ASSERT_EQ(revframe_->num_channels_,
+                  static_cast<size_t>(msg.channel_size()));
         for (int i = 0; i < msg.channel_size(); ++i) {
            memcpy(revfloat_cb_->channels()[i],
                   msg.channel(i).data(),
@@ -1781,7 +1777,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
       }
 
       if (msg.input_channel_size() > 0) {
-        ASSERT_EQ(frame_->num_channels_, msg.input_channel_size());
+        ASSERT_EQ(frame_->num_channels_,
+                  static_cast<size_t>(msg.input_channel_size()));
         for (int i = 0; i < msg.input_channel_size(); ++i) {
            memcpy(float_cb_->channels()[i],
                   msg.input_channel(i).data(),
@@ -1939,11 +1936,14 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
     if (test->num_input_channels() != test->num_output_channels())
       continue;
 
-    const int num_render_channels = test->num_reverse_channels();
-    const int num_input_channels = test->num_input_channels();
-    const int num_output_channels = test->num_output_channels();
-    const int samples_per_channel = test->sample_rate() *
-        AudioProcessing::kChunkSizeMs / 1000;
+    const size_t num_render_channels =
+        static_cast<size_t>(test->num_reverse_channels());
+    const size_t num_input_channels =
+        static_cast<size_t>(test->num_input_channels());
+    const size_t num_output_channels =
+        static_cast<size_t>(test->num_output_channels());
+    const size_t samples_per_channel = static_cast<size_t>(
+        test->sample_rate() * AudioProcessing::kChunkSizeMs / 1000);
 
     Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
          num_input_channels, num_output_channels, num_render_channels, true);
@@ -1984,7 +1984,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
           test->sample_rate(),
           LayoutFromChannels(num_output_channels),
           float_cb_->channels()));
-      for (int j = 0; j < num_output_channels; ++j) {
+      for (size_t j = 0; j < num_output_channels; ++j) {
         FloatToS16(float_cb_->channels()[j],
                    samples_per_channel,
                    output_cb.channels()[j]);
@@ -2017,7 +2017,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
                   0.01);
 
       // Reset in case of downmixing.
-      frame_->num_channels_ = test->num_input_channels();
+      frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
     }
     rewind(far_file_);
     rewind(near_file_);
@@ -2035,9 +2035,9 @@ TEST_F(ApmTest, Process) {
     OpenFileAndReadMessage(ref_filename_, &ref_data);
   } else {
     // Write the desired tests to the protobuf reference file.
-    for (size_t i = 0; i < kChannelsSize; i++) {
-      for (size_t j = 0; j < kChannelsSize; j++) {
-        for (size_t l = 0; l < kProcessSampleRatesSize; l++) {
+    for (size_t i = 0; i < arraysize(kChannels); i++) {
+      for (size_t j = 0; j < arraysize(kChannels); j++) {
+        for (size_t l = 0; l < arraysize(kProcessSampleRates); l++) {
           audioproc::Test* test = ref_data.add_test();
           test->set_num_reverse_channels(kChannels[i]);
           test->set_num_input_channels(kChannels[j]);
@@ -2078,9 +2078,9 @@ TEST_F(ApmTest, Process) {
     Init(test->sample_rate(),
          test->sample_rate(),
          test->sample_rate(),
-         test->num_input_channels(),
-         test->num_output_channels(),
-         test->num_reverse_channels(),
+         static_cast<size_t>(test->num_input_channels()),
+         static_cast<size_t>(test->num_output_channels()),
+         static_cast<size_t>(test->num_reverse_channels()),
          true);
 
     int frame_count = 0;
@@ -2105,7 +2105,8 @@ TEST_F(ApmTest, Process) {
       EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
 
       // Ensure the frame was downmixed properly.
-      EXPECT_EQ(test->num_output_channels(), frame_->num_channels_);
+      EXPECT_EQ(static_cast<size_t>(test->num_output_channels()),
+                frame_->num_channels_);
 
       max_output_average += MaxAudioFrame(*frame_);
 
@@ -2135,7 +2136,7 @@ TEST_F(ApmTest, Process) {
       ASSERT_EQ(frame_size, write_count);
 
       // Reset in case of downmixing.
-      frame_->num_channels_ = test->num_input_channels();
+      frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
       frame_count++;
     }
     max_output_average /= frame_count;
@@ -2264,12 +2265,11 @@ TEST_F(ApmTest, NoErrorsWithKeyboardChannel) {
     {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kMono},
     {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kStereo},
   };
-  size_t channel_format_size = sizeof(cf) / sizeof(*cf);
 
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create());
   // Enable one component just to ensure some processing takes place.
   ap->noise_suppression()->Enable(true);
-  for (size_t i = 0; i < channel_format_size; ++i) {
+  for (size_t i = 0; i < arraysize(cf); ++i) {
     const int in_rate = 44100;
     const int out_rate = 48000;
     ChannelBuffer<float> in_cb(SamplesFromRate(in_rate),
@@ -2296,7 +2296,7 @@ TEST_F(ApmTest, NoErrorsWithKeyboardChannel) {
 // error results to the supplied accumulators.
 void UpdateBestSNR(const float* ref,
                    const float* test,
-                   int length,
+                   size_t length,
                    int expected_delay,
                    double* variance_acc,
                    double* sq_error_acc) {
@@ -2308,7 +2308,7 @@ void UpdateBestSNR(const float* ref,
        ++delay) {
     double sq_error = 0;
     double variance = 0;
-    for (int i = 0; i < length - delay; ++i) {
+    for (size_t i = 0; i < length - delay; ++i) {
       double error = test[i + delay] - ref[i];
       sq_error += error * error;
       variance += ref[i] * ref[i];
@@ -2360,14 +2360,10 @@ class AudioProcessingTest
   static void SetUpTestCase() {
     // Create all needed output reference files.
     const int kNativeRates[] = {8000, 16000, 32000, 48000};
-    const size_t kNativeRatesSize =
-        sizeof(kNativeRates) / sizeof(*kNativeRates);
-    const int kNumChannels[] = {1, 2};
-    const size_t kNumChannelsSize =
-        sizeof(kNumChannels) / sizeof(*kNumChannels);
-    for (size_t i = 0; i < kNativeRatesSize; ++i) {
-      for (size_t j = 0; j < kNumChannelsSize; ++j) {
-        for (size_t k = 0; k < kNumChannelsSize; ++k) {
+    const size_t kNumChannels[] = {1, 2};
+    for (size_t i = 0; i < arraysize(kNativeRates); ++i) {
+      for (size_t j = 0; j < arraysize(kNumChannels); ++j) {
+        for (size_t k = 0; k < arraysize(kNumChannels); ++k) {
           // The reference files always have matching input and output channels.
           ProcessFormat(kNativeRates[i], kNativeRates[i], kNativeRates[i],
                         kNativeRates[i], kNumChannels[j], kNumChannels[j],
@@ -2388,10 +2384,10 @@ class AudioProcessingTest
                             int output_rate,
                             int reverse_input_rate,
                             int reverse_output_rate,
-                            int num_input_channels,
-                            int num_output_channels,
-                            int num_reverse_input_channels,
-                            int num_reverse_output_channels,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
+                            size_t num_reverse_input_channels,
+                            size_t num_reverse_output_channels,
                             std::string output_file_prefix) {
     Config config;
     config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
@@ -2466,18 +2462,19 @@ class AudioProcessingTest
       // Dump forward output to file.
       Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(),
                  float_data.get());
-      int out_length = out_cb.num_channels() * out_cb.num_frames();
+      size_t out_length = out_cb.num_channels() * out_cb.num_frames();
 
-      ASSERT_EQ(static_cast<size_t>(out_length),
+      ASSERT_EQ(out_length,
                 fwrite(float_data.get(), sizeof(float_data[0]),
                        out_length, out_file));
 
       // Dump reverse output to file.
       Interleave(rev_out_cb.channels(), rev_out_cb.num_frames(),
                  rev_out_cb.num_channels(), float_data.get());
-      int rev_out_length = rev_out_cb.num_channels() * rev_out_cb.num_frames();
+      size_t rev_out_length =
+          rev_out_cb.num_channels() * rev_out_cb.num_frames();
 
-      ASSERT_EQ(static_cast<size_t>(rev_out_length),
+      ASSERT_EQ(rev_out_length,
                 fwrite(float_data.get(), sizeof(float_data[0]), rev_out_length,
                        rev_out_file));
 
@@ -2513,9 +2510,8 @@ TEST_P(AudioProcessingTest, Formats) {
       {2, 2, 1, 1},
       {2, 2, 2, 2},
   };
-  size_t channel_format_size = sizeof(cf) / sizeof(*cf);
 
-  for (size_t i = 0; i < channel_format_size; ++i) {
+  for (size_t i = 0; i < arraysize(cf); ++i) {
     ProcessFormat(input_rate_, output_rate_, reverse_input_rate_,
                   reverse_output_rate_, cf[i].num_input, cf[i].num_output,
                   cf[i].num_reverse_input, cf[i].num_reverse_output, "out");
@@ -2565,8 +2561,8 @@ TEST_P(AudioProcessingTest, Formats) {
       ASSERT_TRUE(out_file != NULL);
       ASSERT_TRUE(ref_file != NULL);
 
-      const int ref_length = SamplesFromRate(ref_rate) * out_num;
-      const int out_length = SamplesFromRate(out_rate) * out_num;
+      const size_t ref_length = SamplesFromRate(ref_rate) * out_num;
+      const size_t out_length = SamplesFromRate(out_rate) * out_num;
       // Data from the reference file.
       rtc::scoped_ptr<float[]> ref_data(new float[ref_length]);
       // Data from the output file.
@@ -2606,8 +2602,9 @@ TEST_P(AudioProcessingTest, Formats) {
         if (out_rate != ref_rate) {
           // Resample the output back to its internal processing rate if
           // necssary.
-          ASSERT_EQ(ref_length, resampler.Resample(out_ptr, out_length,
-                                                   cmp_data.get(), ref_length));
+          ASSERT_EQ(ref_length,
+                    static_cast<size_t>(resampler.Resample(
+                        out_ptr, out_length, cmp_data.get(), ref_length)));
           out_ptr = cmp_data.get();
         }
 
@@ -2752,9 +2749,5 @@ INSTANTIATE_TEST_CASE_P(
                     std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 #endif
 
-// TODO(henrike): re-implement functionality lost when removing the old main
-//                function. See
-//                https://code.google.com/p/webrtc/issues/detail?id=1981
-
 }  // namespace
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index 811e9070fa..a489d255c8 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -9,35 +9,50 @@
  */
 
 #include <stdio.h>
+#include <iostream>
 #include <sstream>
 #include <string>
+#include <utility>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 
-DEFINE_string(dump, "", "The name of the debug dump file to read from.");
-DEFINE_string(i, "", "The name of the input file to read from.");
-DEFINE_string(i_rev, "", "The name of the reverse input file to read from.");
-DEFINE_string(o, "out.wav", "Name of the output file to write to.");
-DEFINE_string(o_rev,
-              "out_rev.wav",
-              "Name of the reverse output file to write to.");
-DEFINE_int32(out_channels, 0, "Number of output channels. Defaults to input.");
-DEFINE_int32(out_sample_rate, 0,
-             "Output sample rate in Hz. Defaults to input.");
+namespace {
+
+bool ValidateOutChannels(const char* flagname, int32_t value) {
+  return value >= 0;
+}
+
+}  // namespace
+
+DEFINE_string(dump, "", "Name of the aecdump debug file to read from.");
+DEFINE_string(i, "", "Name of the capture input stream file to read from.");
+DEFINE_string(
+    o,
+    "out.wav",
+    "Name of the output file to write the processed capture stream to.");
+DEFINE_int32(out_channels, 1, "Number of output channels.");
+const bool out_channels_dummy =
+    google::RegisterFlagValidator(&FLAGS_out_channels, &ValidateOutChannels);
+DEFINE_int32(out_sample_rate, 48000, "Output sample rate in Hz.");
 DEFINE_string(mic_positions, "",
     "Space delimited cartesian coordinates of microphones in meters. "
     "The coordinates of each point are contiguous. "
     "For a two element array: \"x1 y1 z1 x2 y2 z2\"");
-DEFINE_double(target_angle_degrees, 90, "The azimuth of the target in radians");
+DEFINE_double(
+    target_angle_degrees,
+    90,
+    "The azimuth of the target in degrees. Only applies to beamforming.");
 
 DEFINE_bool(aec, false, "Enable echo cancellation.");
 DEFINE_bool(agc, false, "Enable automatic gain control.");
@@ -64,15 +79,6 @@ const char kUsage[] =
     "All components are disabled by default. If any bi-directional components\n"
     "are enabled, only debug dump files are permitted.";
 
-// Returns a StreamConfig corresponding to wav_file if it's non-nullptr.
-// Otherwise returns a default initialized StreamConfig.
-StreamConfig MakeStreamConfig(const WavFile* wav_file) {
-  if (wav_file) {
-    return {wav_file->sample_rate(), wav_file->num_channels()};
-  }
-  return {};
-}
-
 }  // namespace
 
 int main(int argc, char* argv[]) {
@@ -84,158 +90,75 @@ int main(int argc, char* argv[]) {
             "An input file must be specified with either -i or -dump.\n");
     return 1;
   }
-  if (!FLAGS_dump.empty()) {
-    fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
+  if (FLAGS_dump.empty() && (FLAGS_aec || FLAGS_ie)) {
+    fprintf(stderr, "-aec and -ie require a -dump file.\n");
+    return 1;
+  }
+  if (FLAGS_ie) {
+    fprintf(stderr,
+            "FIXME(ajm): The intelligibility enhancer output is not dumped.\n");
     return 1;
   }
 
   test::TraceToStderr trace_to_stderr(true);
-  WavReader in_file(FLAGS_i);
-  // If the output format is uninitialized, use the input format.
-  const int out_channels =
-      FLAGS_out_channels ? FLAGS_out_channels : in_file.num_channels();
-  const int out_sample_rate =
-      FLAGS_out_sample_rate ? FLAGS_out_sample_rate : in_file.sample_rate();
-  WavWriter out_file(FLAGS_o, out_sample_rate, out_channels);
-
   Config config;
-  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
-  config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
-
   if (FLAGS_bf || FLAGS_all) {
-    const size_t num_mics = in_file.num_channels();
-    const std::vector<Point> array_geometry =
-        ParseArrayGeometry(FLAGS_mic_positions, num_mics);
-    RTC_CHECK_EQ(array_geometry.size(), num_mics);
-
+    if (FLAGS_mic_positions.empty()) {
+      fprintf(stderr, "-mic_positions must be specified when -bf is used.\n");
+      return 1;
+    }
     config.Set<Beamforming>(new Beamforming(
-        true, array_geometry,
+        true, ParseArrayGeometry(FLAGS_mic_positions),
         SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f,
                         1.f)));
   }
+  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
+  config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
 
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
-  if (!FLAGS_dump.empty()) {
-    RTC_CHECK_EQ(kNoErr,
-                 ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
-  } else if (FLAGS_aec) {
-    fprintf(stderr, "-aec requires a -dump file.\n");
-    return -1;
-  }
-  bool process_reverse = !FLAGS_i_rev.empty();
+  RTC_CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
   RTC_CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
-  RTC_CHECK_EQ(kNoErr,
-               ap->gain_control()->set_mode(GainControl::kFixedDigital));
   RTC_CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
   RTC_CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
-  if (FLAGS_ns_level != -1)
+  if (FLAGS_ns_level != -1) {
     RTC_CHECK_EQ(kNoErr,
                  ap->noise_suppression()->set_level(
                      static_cast<NoiseSuppression::Level>(FLAGS_ns_level)));
-
-  printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-         FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
-  printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-         FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
-
-  ChannelBuffer<float> in_buf(
-      rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
-      in_file.num_channels());
-  ChannelBuffer<float> out_buf(
-      rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
-      out_file.num_channels());
-
-  std::vector<float> in_interleaved(in_buf.size());
-  std::vector<float> out_interleaved(out_buf.size());
-
-  rtc::scoped_ptr<WavReader> in_rev_file;
-  rtc::scoped_ptr<WavWriter> out_rev_file;
-  rtc::scoped_ptr<ChannelBuffer<float>> in_rev_buf;
-  rtc::scoped_ptr<ChannelBuffer<float>> out_rev_buf;
-  std::vector<float> in_rev_interleaved;
-  std::vector<float> out_rev_interleaved;
-  if (process_reverse) {
-    in_rev_file.reset(new WavReader(FLAGS_i_rev));
-    out_rev_file.reset(new WavWriter(FLAGS_o_rev, in_rev_file->sample_rate(),
-                                     in_rev_file->num_channels()));
-    printf("In rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-           FLAGS_i_rev.c_str(), in_rev_file->num_channels(),
-           in_rev_file->sample_rate());
-    printf("Out rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-           FLAGS_o_rev.c_str(), out_rev_file->num_channels(),
-           out_rev_file->sample_rate());
-    in_rev_buf.reset(new ChannelBuffer<float>(
-        rtc::CheckedDivExact(in_rev_file->sample_rate(), kChunksPerSecond),
-        in_rev_file->num_channels()));
-    in_rev_interleaved.resize(in_rev_buf->size());
-    out_rev_buf.reset(new ChannelBuffer<float>(
-        rtc::CheckedDivExact(out_rev_file->sample_rate(), kChunksPerSecond),
-        out_rev_file->num_channels()));
-    out_rev_interleaved.resize(out_rev_buf->size());
+  }
+  ap->set_stream_key_pressed(FLAGS_ts);
+
+  rtc::scoped_ptr<AudioFileProcessor> processor;
+  auto out_file = rtc_make_scoped_ptr(new WavWriter(
+      FLAGS_o, FLAGS_out_sample_rate, static_cast<size_t>(FLAGS_out_channels)));
+  std::cout << FLAGS_o << ": " << out_file->FormatAsString() << std::endl;
+  if (FLAGS_dump.empty()) {
+    auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i));
+    std::cout << FLAGS_i << ": " << in_file->FormatAsString() << std::endl;
+    processor.reset(new WavFileProcessor(std::move(ap), std::move(in_file),
+                                         std::move(out_file)));
+
+  } else {
+    processor.reset(new AecDumpFileProcessor(
+        std::move(ap), fopen(FLAGS_dump.c_str(), "rb"), std::move(out_file)));
   }
 
-  TickTime processing_start_time;
-  TickInterval accumulated_time;
   int num_chunks = 0;
-
-  const auto input_config = MakeStreamConfig(&in_file);
-  const auto output_config = MakeStreamConfig(&out_file);
-  const auto reverse_input_config = MakeStreamConfig(in_rev_file.get());
-  const auto reverse_output_config = MakeStreamConfig(out_rev_file.get());
-
-  while (in_file.ReadSamples(in_interleaved.size(),
-                             &in_interleaved[0]) == in_interleaved.size()) {
-    // Have logs display the file time rather than wallclock time.
+  while (processor->ProcessChunk()) {
     trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond);
-    FloatS16ToFloat(&in_interleaved[0], in_interleaved.size(),
-                    &in_interleaved[0]);
-    Deinterleave(&in_interleaved[0], in_buf.num_frames(),
-                 in_buf.num_channels(), in_buf.channels());
-    if (process_reverse) {
-      in_rev_file->ReadSamples(in_rev_interleaved.size(),
-                               in_rev_interleaved.data());
-      FloatS16ToFloat(in_rev_interleaved.data(), in_rev_interleaved.size(),
-                      in_rev_interleaved.data());
-      Deinterleave(in_rev_interleaved.data(), in_rev_buf->num_frames(),
-                   in_rev_buf->num_channels(), in_rev_buf->channels());
-    }
-
-    if (FLAGS_perf) {
-      processing_start_time = TickTime::Now();
-    }
-    RTC_CHECK_EQ(kNoErr, ap->ProcessStream(in_buf.channels(), input_config,
-                                           output_config, out_buf.channels()));
-    if (process_reverse) {
-      RTC_CHECK_EQ(kNoErr, ap->ProcessReverseStream(
-                               in_rev_buf->channels(), reverse_input_config,
-                               reverse_output_config, out_rev_buf->channels()));
-    }
-    if (FLAGS_perf) {
-      accumulated_time += TickTime::Now() - processing_start_time;
-    }
-
-    Interleave(out_buf.channels(), out_buf.num_frames(),
-               out_buf.num_channels(), &out_interleaved[0]);
-    FloatToFloatS16(&out_interleaved[0], out_interleaved.size(),
-                    &out_interleaved[0]);
-    out_file.WriteSamples(&out_interleaved[0], out_interleaved.size());
-    if (process_reverse) {
-      Interleave(out_rev_buf->channels(), out_rev_buf->num_frames(),
-                 out_rev_buf->num_channels(), out_rev_interleaved.data());
-      FloatToFloatS16(out_rev_interleaved.data(), out_rev_interleaved.size(),
-                      out_rev_interleaved.data());
-      out_rev_file->WriteSamples(out_rev_interleaved.data(),
-                                 out_rev_interleaved.size());
-    }
-    num_chunks++;
+    ++num_chunks;
   }
+
   if (FLAGS_perf) {
-    int64_t execution_time_ms = accumulated_time.Milliseconds();
-    printf("\nExecution time: %.3f s\nFile time: %.2f s\n"
-           "Time per chunk: %.3f ms\n",
-           execution_time_ms * 0.001f, num_chunks * 1.f / kChunksPerSecond,
-           execution_time_ms * 1.f / num_chunks);
+    const auto& proc_time = processor->proc_time();
+    int64_t exec_time_us = proc_time.sum.Microseconds();
+    printf(
+        "\nExecution time: %.3f s, File time: %.2f s\n"
+        "Time per chunk (mean, max, min):\n%.0f us, %.0f us, %.0f us\n",
+        exec_time_us * 1e-6, num_chunks * 1.f / kChunksPerSecond,
+        exec_time_us * 1.f / num_chunks, 1.f * proc_time.max.Microseconds(),
+        1.f * proc_time.min.Microseconds());
   }
+
   return 0;
 }
 
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc
new file mode 100644
index 0000000000..005faa0f44
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -0,0 +1,612 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>  // size_t
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+void MaybeResetBuffer(rtc::scoped_ptr<ChannelBuffer<float>>* buffer,
+                      const StreamConfig& config) {
+  auto& buffer_ref = *buffer;
+  if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
+      buffer_ref->num_channels() != config.num_channels()) {
+    buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(),
+                                             config.num_channels()));
+  }
+}
+
+class DebugDumpGenerator {
+ public:
+  DebugDumpGenerator(const std::string& input_file_name,
+                     int input_file_rate_hz,
+                     int input_channels,
+                     const std::string& reverse_file_name,
+                     int reverse_file_rate_hz,
+                     int reverse_channels,
+                     const Config& config,
+                     const std::string& dump_file_name);
+
+  // Constructor that uses default input files.
+  explicit DebugDumpGenerator(const Config& config);
+
+  ~DebugDumpGenerator();
+
+  // Changes the sample rate of the input audio to the APM.
+  void SetInputRate(int rate_hz);
+
+  // Sets if converts stereo input signal to mono by discarding other channels.
+  void ForceInputMono(bool mono);
+
+  // Changes the sample rate of the reverse audio to the APM.
+  void SetReverseRate(int rate_hz);
+
+  // Sets if converts stereo reverse signal to mono by discarding other
+  // channels.
+  void ForceReverseMono(bool mono);
+
+  // Sets the required sample rate of the APM output.
+  void SetOutputRate(int rate_hz);
+
+  // Sets the required channels of the APM output.
+  void SetOutputChannels(int channels);
+
+  std::string dump_file_name() const { return dump_file_name_; }
+
+  void StartRecording();
+  void Process(size_t num_blocks);
+  void StopRecording();
+  AudioProcessing* apm() const { return apm_.get(); }
+
+ private:
+  static void ReadAndDeinterleave(ResampleInputAudioFile* audio, int channels,
+                                  const StreamConfig& config,
+                                  float* const* buffer);
+
+  // APM input/output settings.
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  StreamConfig output_config_;
+
+  // Input file format.
+  const std::string input_file_name_;
+  ResampleInputAudioFile input_audio_;
+  const int input_file_channels_;
+
+  // Reverse file format.
+  const std::string reverse_file_name_;
+  ResampleInputAudioFile reverse_audio_;
+  const int reverse_file_channels_;
+
+  // Buffer for APM input/output.
+  rtc::scoped_ptr<ChannelBuffer<float>> input_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
+  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+
+  const std::string dump_file_name_;
+};
+
+DebugDumpGenerator::DebugDumpGenerator(const std::string& input_file_name,
+                                       int input_rate_hz,
+                                       int input_channels,
+                                       const std::string& reverse_file_name,
+                                       int reverse_rate_hz,
+                                       int reverse_channels,
+                                       const Config& config,
+                                       const std::string& dump_file_name)
+    : input_config_(input_rate_hz, input_channels),
+      reverse_config_(reverse_rate_hz, reverse_channels),
+      output_config_(input_rate_hz, input_channels),
+      input_audio_(input_file_name, input_rate_hz, input_rate_hz),
+      input_file_channels_(input_channels),
+      reverse_audio_(reverse_file_name, reverse_rate_hz, reverse_rate_hz),
+      reverse_file_channels_(reverse_channels),
+      input_(new ChannelBuffer<float>(input_config_.num_frames(),
+                                      input_config_.num_channels())),
+      reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(),
+                                        reverse_config_.num_channels())),
+      output_(new ChannelBuffer<float>(output_config_.num_frames(),
+                                       output_config_.num_channels())),
+      apm_(AudioProcessing::Create(config)),
+      dump_file_name_(dump_file_name) {
+}
+
+DebugDumpGenerator::DebugDumpGenerator(const Config& config)
+  : DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"), 32000, 2,
+                       ResourcePath("far32_stereo", "pcm"), 32000, 2,
+                       config,
+                       TempFilename(OutputPath(), "debug_aec")) {
+}
+
+DebugDumpGenerator::~DebugDumpGenerator() {
+  remove(dump_file_name_.c_str());
+}
+
+void DebugDumpGenerator::SetInputRate(int rate_hz) {
+  input_audio_.set_output_rate_hz(rate_hz);
+  input_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&input_, input_config_);
+}
+
+void DebugDumpGenerator::ForceInputMono(bool mono) {
+  const int channels = mono ? 1 : input_file_channels_;
+  input_config_.set_num_channels(channels);
+  MaybeResetBuffer(&input_, input_config_);
+}
+
+void DebugDumpGenerator::SetReverseRate(int rate_hz) {
+  reverse_audio_.set_output_rate_hz(rate_hz);
+  reverse_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+void DebugDumpGenerator::ForceReverseMono(bool mono) {
+  const int channels = mono ? 1 : reverse_file_channels_;
+  reverse_config_.set_num_channels(channels);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+void DebugDumpGenerator::SetOutputRate(int rate_hz) {
+  output_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&output_, output_config_);
+}
+
+void DebugDumpGenerator::SetOutputChannels(int channels) {
+  output_config_.set_num_channels(channels);
+  MaybeResetBuffer(&output_, output_config_);
+}
+
+void DebugDumpGenerator::StartRecording() {
+  apm_->StartDebugRecording(dump_file_name_.c_str());
+}
+
+void DebugDumpGenerator::Process(size_t num_blocks) {
+  for (size_t i = 0; i < num_blocks; ++i) {
+    ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_,
+                        reverse_config_, reverse_->channels());
+    ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
+                        input_->channels());
+    RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
+    apm_->set_stream_key_pressed(i % 10 == 9);
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 apm_->ProcessStream(input_->channels(), input_config_,
+                                     output_config_, output_->channels()));
+
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 apm_->ProcessReverseStream(reverse_->channels(),
+                                            reverse_config_,
+                                            reverse_config_,
+                                            reverse_->channels()));
+  }
+}
+
+void DebugDumpGenerator::StopRecording() {
+  apm_->StopDebugRecording();
+}
+
+void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio,
+                                             int channels,
+                                             const StreamConfig& config,
+                                             float* const* buffer) {
+  const size_t num_frames = config.num_frames();
+  const int out_channels = config.num_channels();
+
+  std::vector<int16_t> signal(channels * num_frames);
+
+  audio->Read(num_frames * channels, &signal[0]);
+
+  // We only allow reducing number of channels by discarding some channels.
+  RTC_CHECK_LE(out_channels, channels);
+  for (int channel = 0; channel < out_channels; ++channel) {
+    for (size_t i = 0; i < num_frames; ++i) {
+      buffer[channel][i] = S16ToFloat(signal[i * channels + channel]);
+    }
+  }
+}
+
+}  // namespace
+
+class DebugDumpTest : public ::testing::Test {
+ public:
+  DebugDumpTest();
+
+  // VerifyDebugDump replays a debug dump using APM and verifies that the result
+  // is bit-exact-identical to the output channel in the dump. This is only
+  // guaranteed if the debug dump is started on the first frame.
+  void VerifyDebugDump(const std::string& dump_file_name);
+
+ private:
+  // Following functions are facilities for replaying debug dumps.
+  void OnInitEvent(const audioproc::Init& msg);
+  void OnStreamEvent(const audioproc::Stream& msg);
+  void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
+  void OnConfigEvent(const audioproc::Config& msg);
+
+  void MaybeRecreateApm(const audioproc::Config& msg);
+  void ConfigureApm(const audioproc::Config& msg);
+
+  // Buffer for APM input/output.
+  rtc::scoped_ptr<ChannelBuffer<float>> input_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
+  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  StreamConfig output_config_;
+};
+
+DebugDumpTest::DebugDumpTest()
+    : input_(nullptr),  // will be created upon usage.
+      reverse_(nullptr),
+      output_(nullptr),
+      apm_(nullptr) {
+}
+
+void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) {
+  FILE* in_file = fopen(in_filename.c_str(), "rb");
+  ASSERT_TRUE(in_file);
+  audioproc::Event event_msg;
+
+  while (ReadMessageFromFile(in_file, &event_msg)) {
+    switch (event_msg.type()) {
+      case audioproc::Event::INIT:
+        OnInitEvent(event_msg.init());
+        break;
+      case audioproc::Event::STREAM:
+        OnStreamEvent(event_msg.stream());
+        break;
+      case audioproc::Event::REVERSE_STREAM:
+        OnReverseStreamEvent(event_msg.reverse_stream());
+        break;
+      case audioproc::Event::CONFIG:
+        OnConfigEvent(event_msg.config());
+        break;
+      case audioproc::Event::UNKNOWN_EVENT:
+        // We do not expect receive UNKNOWN event currently.
+        FAIL();
+    }
+  }
+  fclose(in_file);
+}
+
+// OnInitEvent reset the input/output/reserve channel format.
+void DebugDumpTest::OnInitEvent(const audioproc::Init& msg) {
+  ASSERT_TRUE(msg.has_num_input_channels());
+  ASSERT_TRUE(msg.has_output_sample_rate());
+  ASSERT_TRUE(msg.has_num_output_channels());
+  ASSERT_TRUE(msg.has_reverse_sample_rate());
+  ASSERT_TRUE(msg.has_num_reverse_channels());
+
+  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+  output_config_ =
+      StreamConfig(msg.output_sample_rate(), msg.num_output_channels());
+  reverse_config_ =
+      StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
+
+  MaybeResetBuffer(&input_, input_config_);
+  MaybeResetBuffer(&output_, output_config_);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+// OnStreamEvent replays an input signal and verifies the output.
+void DebugDumpTest::OnStreamEvent(const audioproc::Stream& msg) {
+  // APM should have been created.
+  ASSERT_TRUE(apm_.get());
+
+  EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(msg.level()));
+  EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
+  apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+  if (msg.has_keypress())
+    apm_->set_stream_key_pressed(msg.keypress());
+  else
+    apm_->set_stream_key_pressed(true);
+
+  ASSERT_EQ(input_config_.num_channels(),
+            static_cast<size_t>(msg.input_channel_size()));
+  ASSERT_EQ(input_config_.num_frames() * sizeof(float),
+            msg.input_channel(0).size());
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+     memcpy(input_->channels()[i], msg.input_channel(i).data(),
+            msg.input_channel(i).size());
+  }
+
+  ASSERT_EQ(AudioProcessing::kNoError,
+            apm_->ProcessStream(input_->channels(), input_config_,
+                                output_config_, output_->channels()));
+
+  // Check that output of APM is bit-exact to the output in the dump.
+  ASSERT_EQ(output_config_.num_channels(),
+            static_cast<size_t>(msg.output_channel_size()));
+  ASSERT_EQ(output_config_.num_frames() * sizeof(float),
+            msg.output_channel(0).size());
+  for (int i = 0; i < msg.output_channel_size(); ++i) {
+    ASSERT_EQ(0, memcmp(output_->channels()[i], msg.output_channel(i).data(),
+                        msg.output_channel(i).size()));
+  }
+}
+
+void DebugDumpTest::OnReverseStreamEvent(const audioproc::ReverseStream& msg) {
+  // APM should have been created.
+  ASSERT_TRUE(apm_.get());
+
+  ASSERT_GT(msg.channel_size(), 0);
+  ASSERT_EQ(reverse_config_.num_channels(),
+            static_cast<size_t>(msg.channel_size()));
+  ASSERT_EQ(reverse_config_.num_frames() * sizeof(float),
+            msg.channel(0).size());
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+     memcpy(reverse_->channels()[i], msg.channel(i).data(),
+            msg.channel(i).size());
+  }
+
+  ASSERT_EQ(AudioProcessing::kNoError,
+            apm_->ProcessReverseStream(reverse_->channels(),
+                                       reverse_config_,
+                                       reverse_config_,
+                                       reverse_->channels()));
+}
+
+void DebugDumpTest::OnConfigEvent(const audioproc::Config& msg) {
+  MaybeRecreateApm(msg);
+  ConfigureApm(msg);
+}
+
+void DebugDumpTest::MaybeRecreateApm(const audioproc::Config& msg) {
+  // These configurations cannot be changed on the fly.
+  Config config;
+  ASSERT_TRUE(msg.has_aec_delay_agnostic_enabled());
+  config.Set<DelayAgnostic>(
+      new DelayAgnostic(msg.aec_delay_agnostic_enabled()));
+
+  ASSERT_TRUE(msg.has_noise_robust_agc_enabled());
+  config.Set<ExperimentalAgc>(
+       new ExperimentalAgc(msg.noise_robust_agc_enabled()));
+
+  ASSERT_TRUE(msg.has_transient_suppression_enabled());
+  config.Set<ExperimentalNs>(
+      new ExperimentalNs(msg.transient_suppression_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_extended_filter_enabled());
+  config.Set<ExtendedFilter>(new ExtendedFilter(
+      msg.aec_extended_filter_enabled()));
+
+  // We only create APM once, since changes on these fields should not
+  // happen in current implementation.
+  if (!apm_.get()) {
+    apm_.reset(AudioProcessing::Create(config));
+  }
+}
+
+void DebugDumpTest::ConfigureApm(const audioproc::Config& msg) {
+  // AEC configs.
+  ASSERT_TRUE(msg.has_aec_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->Enable(msg.aec_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_drift_compensation_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->enable_drift_compensation(
+                msg.aec_drift_compensation_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_suppression_level());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->set_suppression_level(
+                static_cast<EchoCancellation::SuppressionLevel>(
+                    msg.aec_suppression_level())));
+
+  // AECM configs.
+  ASSERT_TRUE(msg.has_aecm_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->Enable(msg.aecm_enabled()));
+
+  ASSERT_TRUE(msg.has_aecm_comfort_noise_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->enable_comfort_noise(
+                msg.aecm_comfort_noise_enabled()));
+
+  ASSERT_TRUE(msg.has_aecm_routing_mode());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->set_routing_mode(
+                static_cast<EchoControlMobile::RoutingMode>(
+                    msg.aecm_routing_mode())));
+
+  // AGC configs.
+  ASSERT_TRUE(msg.has_agc_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->Enable(msg.agc_enabled()));
+
+  ASSERT_TRUE(msg.has_agc_mode());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->set_mode(
+                static_cast<GainControl::Mode>(msg.agc_mode())));
+
+  ASSERT_TRUE(msg.has_agc_limiter_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->enable_limiter(msg.agc_limiter_enabled()));
+
+  // HPF configs.
+  ASSERT_TRUE(msg.has_hpf_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->high_pass_filter()->Enable(msg.hpf_enabled()));
+
+  // NS configs.
+  ASSERT_TRUE(msg.has_ns_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->noise_suppression()->Enable(msg.ns_enabled()));
+
+  ASSERT_TRUE(msg.has_ns_level());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->noise_suppression()->set_level(
+                static_cast<NoiseSuppression::Level>(msg.ns_level())));
+}
+
+TEST_F(DebugDumpTest, SimpleCase) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeInputFormat) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetInputRate(48000);
+
+  generator.ForceInputMono(true);
+  // Number of output channel should not be larger than that of input. APM will
+  // fail otherwise.
+  generator.SetOutputChannels(1);
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeReverseFormat) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetReverseRate(48000);
+  generator.ForceReverseMono(true);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeOutputFormat) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetOutputRate(48000);
+  generator.SetOutputChannels(1);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleAec) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleDelayAgnosticAec) {
+  Config config;
+  config.Set<DelayAgnostic>(new DelayAgnostic(true));
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleAecLevel) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(true));
+  EXPECT_EQ(AudioProcessing::kNoError,
+            aec->set_suppression_level(EchoCancellation::kLowSuppression));
+  generator.StartRecording();
+  generator.Process(100);
+
+  EXPECT_EQ(AudioProcessing::kNoError,
+            aec->set_suppression_level(EchoCancellation::kHighSuppression));
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+#if defined(WEBRTC_ANDROID)
+// AGC may not be supported on Android.
+#define MAYBE_ToggleAgc DISABLED_ToggleAgc
+#else
+#define MAYBE_ToggleAgc ToggleAgc
+#endif
+TEST_F(DebugDumpTest, MAYBE_ToggleAgc) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  GainControl* agc = generator.apm()->gain_control();
+  EXPECT_EQ(AudioProcessing::kNoError, agc->Enable(!agc->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleNs) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  NoiseSuppression* ns = generator.apm()->noise_suppression();
+  EXPECT_EQ(AudioProcessing::kNoError, ns->Enable(!ns->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, TransientSuppressionOn) {
+  Config config;
+  config.Set<ExperimentalNs>(new ExperimentalNs(true));
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index 43165404c8..6e20a787e7 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -17,12 +17,13 @@
 
 #include <algorithm>
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -159,9 +160,9 @@ void void_main(int argc, char* argv[]) {
 
   int32_t sample_rate_hz = 16000;
 
-  int num_capture_input_channels = 1;
-  int num_capture_output_channels = 1;
-  int num_render_channels = 1;
+  size_t num_capture_input_channels = 1;
+  size_t num_capture_output_channels = 1;
+  size_t num_render_channels = 1;
 
   int samples_per_channel = sample_rate_hz / 100;
 
@@ -207,14 +208,14 @@ void void_main(int argc, char* argv[]) {
     } else if (strcmp(argv[i], "-ch") == 0) {
       i++;
       ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
+      ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_capture_input_channels));
       i++;
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
+      ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_capture_output_channels));
 
     } else if (strcmp(argv[i], "-rch") == 0) {
       i++;
       ASSERT_LT(i, argc) << "Specify number of channels after -rch";
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
+      ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_render_channels));
 
     } else if (strcmp(argv[i], "-aec") == 0) {
       ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
@@ -447,10 +448,10 @@ void void_main(int argc, char* argv[]) {
 
   if (verbose) {
     printf("Sample rate: %d Hz\n", sample_rate_hz);
-    printf("Primary channels: %d (in), %d (out)\n",
+    printf("Primary channels: %" PRIuS " (in), %" PRIuS " (out)\n",
            num_capture_input_channels,
            num_capture_output_channels);
-    printf("Reverse channels: %d \n", num_render_channels);
+    printf("Reverse channels: %" PRIuS "\n", num_render_channels);
   }
 
   const std::string out_path = webrtc::test::OutputPath();
@@ -601,14 +602,18 @@ void void_main(int argc, char* argv[]) {
         if (msg.has_output_sample_rate()) {
           output_sample_rate = msg.output_sample_rate();
         }
-        output_layout = LayoutFromChannels(msg.num_output_channels());
-        ASSERT_EQ(kNoErr, apm->Initialize(
-                              msg.sample_rate(),
-                              output_sample_rate,
-                              reverse_sample_rate,
-                              LayoutFromChannels(msg.num_input_channels()),
-                              output_layout,
-                              LayoutFromChannels(msg.num_reverse_channels())));
+        output_layout =
+            LayoutFromChannels(static_cast<size_t>(msg.num_output_channels()));
+        ASSERT_EQ(kNoErr,
+                  apm->Initialize(
+                      msg.sample_rate(),
+                      output_sample_rate,
+                      reverse_sample_rate,
+                      LayoutFromChannels(
+                          static_cast<size_t>(msg.num_input_channels())),
+                      output_layout,
+                      LayoutFromChannels(
+                          static_cast<size_t>(msg.num_reverse_channels()))));
 
         samples_per_channel = msg.sample_rate() / 100;
         far_frame.sample_rate_hz_ = reverse_sample_rate;
@@ -636,11 +641,11 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (!raw_output) {
-          // The WAV file needs to be reset every time, because it cant change
-          // it's sample rate or number of channels.
-          output_wav_file.reset(new WavWriter(out_filename + ".wav",
-                                              output_sample_rate,
-                                              msg.num_output_channels()));
+          // The WAV file needs to be reset every time, because it can't change
+          // its sample rate or number of channels.
+          output_wav_file.reset(new WavWriter(
+              out_filename + ".wav", output_sample_rate,
+              static_cast<size_t>(msg.num_output_channels())));
         }
 
       } else if (event_msg.type() == Event::REVERSE_STREAM) {
@@ -1049,7 +1054,9 @@ void void_main(int argc, char* argv[]) {
       }
     }
   }
-  printf("100%% complete\r");
+  if (progress) {
+    printf("100%% complete\r");
+  }
 
   if (aecm_echo_path_out_file != NULL) {
     const size_t path_size =
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 1b9ac3ce4c..0bd70126ae 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <utility>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 
@@ -31,6 +33,35 @@ void RawFile::WriteSamples(const float* samples, size_t num_samples) {
   fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 }
 
+ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
+    : file_(std::move(file)) {}
+
+bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
+  interleaved_.resize(buffer->size());
+  if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
+      interleaved_.size()) {
+    return false;
+  }
+
+  FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
+               buffer->channels());
+  return true;
+}
+
+ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
+    : file_(std::move(file)) {}
+
+void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
+  interleaved_.resize(buffer.size());
+  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+             &interleaved_[0]);
+  FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  file_->WriteSamples(&interleaved_[0], interleaved_.size());
+}
+
 void WriteIntData(const int16_t* data,
                   size_t length,
                   WavWriter* wav_file,
@@ -44,8 +75,8 @@ void WriteIntData(const int16_t* data,
 }
 
 void WriteFloatData(const float* const* data,
-                    int samples_per_channel,
-                    int num_channels,
+                    size_t samples_per_channel,
+                    size_t num_channels,
                     WavWriter* wav_file,
                     RawFile* raw_file) {
   size_t length = num_channels * samples_per_channel;
@@ -74,8 +105,8 @@ FILE* OpenFile(const std::string& filename, const char* mode) {
   return file;
 }
 
-int SamplesFromRate(int rate) {
-  return AudioProcessing::kChunkSizeMs * rate / 1000;
+size_t SamplesFromRate(int rate) {
+  return static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate / 1000);
 }
 
 void SetFrameSampleRate(AudioFrame* frame,
@@ -85,35 +116,39 @@ void SetFrameSampleRate(AudioFrame* frame,
       sample_rate_hz / 1000;
 }
 
-AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels) {
+AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
   switch (num_channels) {
     case 1:
       return AudioProcessing::kMono;
     case 2:
       return AudioProcessing::kStereo;
     default:
-      assert(false);
+      RTC_CHECK(false);
       return AudioProcessing::kMono;
   }
 }
 
-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
-                                      size_t num_mics) {
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) {
   const std::vector<float> values = ParseList<float>(mic_positions);
-  RTC_CHECK_EQ(values.size(), 3 * num_mics)
-      << "Could not parse mic_positions or incorrect number of points.";
+  const size_t num_mics =
+      rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
+  RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough.";
 
   std::vector<Point> result;
   result.reserve(num_mics);
   for (size_t i = 0; i < values.size(); i += 3) {
-    double x = values[i + 0];
-    double y = values[i + 1];
-    double z = values[i + 2];
-    result.push_back(Point(x, y, z));
+    result.push_back(Point(values[i + 0], values[i + 1], values[i + 2]));
   }
 
   return result;
 }
 
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
+                                      size_t num_mics) {
+  std::vector<Point> result = ParseArrayGeometry(mic_positions);
+  RTC_CHECK_EQ(result.size(), num_mics)
+      << "Could not parse mic_positions or incorrect number of points.";
+  return result;
+}
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index 8dd380b15d..e23beb66f4 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -22,7 +22,7 @@
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -43,28 +43,57 @@ class RawFile final {
   RTC_DISALLOW_COPY_AND_ASSIGN(RawFile);
 };
 
+// Reads ChannelBuffers from a provided WavReader.
+class ChannelBufferWavReader final {
+ public:
+  explicit ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file);
+
+  // Reads data from the file according to the |buffer| format. Returns false if
+  // a full buffer can't be read from the file.
+  bool Read(ChannelBuffer<float>* buffer);
+
+ private:
+  rtc::scoped_ptr<WavReader> file_;
+  std::vector<float> interleaved_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader);
+};
+
+// Writes ChannelBuffers to a provided WavWriter.
+class ChannelBufferWavWriter final {
+ public:
+  explicit ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file);
+  void Write(const ChannelBuffer<float>& buffer);
+
+ private:
+  rtc::scoped_ptr<WavWriter> file_;
+  std::vector<float> interleaved_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter);
+};
+
 void WriteIntData(const int16_t* data,
                   size_t length,
                   WavWriter* wav_file,
                   RawFile* raw_file);
 
 void WriteFloatData(const float* const* data,
-                    int samples_per_channel,
-                    int num_channels,
+                    size_t samples_per_channel,
+                    size_t num_channels,
                     WavWriter* wav_file,
                     RawFile* raw_file);
 
 // Exits on failure; do not use in unit tests.
 FILE* OpenFile(const std::string& filename, const char* mode);
 
-int SamplesFromRate(int rate);
+size_t SamplesFromRate(int rate);
 
 void SetFrameSampleRate(AudioFrame* frame,
                         int sample_rate_hz);
 
 template <typename T>
 void SetContainerFormat(int sample_rate_hz,
-                        int num_channels,
+                        size_t num_channels,
                         AudioFrame* frame,
                         rtc::scoped_ptr<ChannelBuffer<T> >* cb) {
   SetFrameSampleRate(frame, sample_rate_hz);
@@ -72,14 +101,14 @@ void SetContainerFormat(int sample_rate_hz,
   cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
 }
 
-AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels);
+AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels);
 
 template <typename T>
-float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
+float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
   float mse = 0;
   float mean = 0;
   *variance = 0;
-  for (int i = 0; i < length; ++i) {
+  for (size_t i = 0; i < length; ++i) {
     T error = ref[i] - test[i];
     mse += error * error;
     *variance += ref[i] * ref[i];
@@ -118,6 +147,9 @@ std::vector<T> ParseList(const std::string& to_parse) {
 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
                                       size_t num_mics);
 
+// Same as above, but without the num_mics check for when it isn't available.
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions);
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
diff --git a/webrtc/modules/audio_processing/test/unpack.cc b/webrtc/modules/audio_processing/test/unpack.cc
index 24578e240c..8b2b082f97 100644
--- a/webrtc/modules/audio_processing/test/unpack.cc
+++ b/webrtc/modules/audio_processing/test/unpack.cc
@@ -17,6 +17,7 @@
 
 #include "gflags/gflags.h"
 #include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
@@ -76,12 +77,12 @@ int do_main(int argc, char* argv[]) {
 
   Event event_msg;
   int frame_count = 0;
-  int reverse_samples_per_channel = 0;
-  int input_samples_per_channel = 0;
-  int output_samples_per_channel = 0;
-  int num_reverse_channels = 0;
-  int num_input_channels = 0;
-  int num_output_channels = 0;
+  size_t reverse_samples_per_channel = 0;
+  size_t input_samples_per_channel = 0;
+  size_t output_samples_per_channel = 0;
+  size_t num_reverse_channels = 0;
+  size_t num_input_channels = 0;
+  size_t num_output_channels = 0;
   rtc::scoped_ptr<WavWriter> reverse_wav_file;
   rtc::scoped_ptr<WavWriter> input_wav_file;
   rtc::scoped_ptr<WavWriter> output_wav_file;
@@ -117,7 +118,7 @@ int do_main(int argc, char* argv[]) {
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_reverse_channels]);
-        for (int i = 0; i < num_reverse_channels; ++i) {
+        for (size_t i = 0; i < num_reverse_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
         }
         WriteFloatData(data.get(),
@@ -148,7 +149,7 @@ int do_main(int argc, char* argv[]) {
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_input_channels]);
-        for (int i = 0; i < num_input_channels; ++i) {
+        for (size_t i = 0; i < num_input_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
         }
         WriteFloatData(data.get(),
@@ -172,7 +173,7 @@ int do_main(int argc, char* argv[]) {
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_output_channels]);
-        for (int i = 0; i < num_output_channels; ++i) {
+        for (size_t i = 0; i < num_output_channels; ++i) {
           data[i] =
               reinterpret_cast<const float*>(msg.output_channel(i).data());
         }
@@ -268,11 +269,14 @@ int do_main(int argc, char* argv[]) {
               "  Reverse sample rate: %d\n",
               reverse_sample_rate);
       num_input_channels = msg.num_input_channels();
-      fprintf(settings_file, "  Input channels: %d\n", num_input_channels);
+      fprintf(settings_file, "  Input channels: %" PRIuS "\n",
+              num_input_channels);
       num_output_channels = msg.num_output_channels();
-      fprintf(settings_file, "  Output channels: %d\n", num_output_channels);
+      fprintf(settings_file, "  Output channels: %" PRIuS "\n",
+              num_output_channels);
       num_reverse_channels = msg.num_reverse_channels();
-      fprintf(settings_file, "  Reverse channels: %d\n", num_reverse_channels);
+      fprintf(settings_file, "  Reverse channels: %" PRIuS "\n",
+              num_reverse_channels);
 
       fprintf(settings_file, "\n");
 
@@ -283,9 +287,12 @@ int do_main(int argc, char* argv[]) {
         output_sample_rate = input_sample_rate;
       }
 
-      reverse_samples_per_channel = reverse_sample_rate / 100;
-      input_samples_per_channel = input_sample_rate / 100;
-      output_samples_per_channel = output_sample_rate / 100;
+      reverse_samples_per_channel =
+          static_cast<size_t>(reverse_sample_rate / 100);
+      input_samples_per_channel =
+          static_cast<size_t>(input_sample_rate / 100);
+      output_samples_per_channel =
+          static_cast<size_t>(output_sample_rate / 100);
 
       if (!FLAGS_raw) {
         // The WAV files need to be reset every time, because they cant change
diff --git a/webrtc/modules/audio_processing/transient/file_utils_unittest.cc b/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
index 7a035d2b41..7fb7d2d6a9 100644
--- a/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
+++ b/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
@@ -17,7 +17,6 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -59,7 +58,12 @@ class TransientFileUtilsTest: public ::testing::Test {
   const std::string kTestFileNamef;
 };
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToFloat)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToFloat DISABLED_ConvertByteArrayToFloat
+#else
+#define MAYBE_ConvertByteArrayToFloat ConvertByteArrayToFloat
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToFloat) {
   float value = 0.0;
 
   EXPECT_EQ(0, ConvertByteArrayToFloat(kPiBytesf, &value));
@@ -72,7 +76,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToFloat)) {
   EXPECT_FLOAT_EQ(kAvogadro, value);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToDouble)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToDouble DISABLED_ConvertByteArrayToDouble
+#else
+#define MAYBE_ConvertByteArrayToDouble ConvertByteArrayToDouble
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToDouble) {
   double value = 0.0;
 
   EXPECT_EQ(0, ConvertByteArrayToDouble(kPiBytes, &value));
@@ -85,7 +94,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertByteArrayToDouble)) {
   EXPECT_DOUBLE_EQ(kAvogadro, value);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertFloatToByteArray)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertFloatToByteArray DISABLED_ConvertFloatToByteArray
+#else
+#define MAYBE_ConvertFloatToByteArray ConvertFloatToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertFloatToByteArray) {
   rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[4]);
 
   EXPECT_EQ(0, ConvertFloatToByteArray(kPi, bytes.get()));
@@ -98,7 +112,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertFloatToByteArray)) {
   EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytesf, 4));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertDoubleToByteArray)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertDoubleToByteArray DISABLED_ConvertDoubleToByteArray
+#else
+#define MAYBE_ConvertDoubleToByteArray ConvertDoubleToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertDoubleToByteArray) {
   rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[8]);
 
   EXPECT_EQ(0, ConvertDoubleToByteArray(kPi, bytes.get()));
@@ -111,7 +130,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ConvertDoubleToByteArray)) {
   EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytes, 8));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadInt16BufferFromFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16BufferFromFile DISABLED_ReadInt16BufferFromFile
+#else
+#define MAYBE_ReadInt16BufferFromFile ReadInt16BufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16BufferFromFile) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -149,8 +173,13 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadInt16BufferFromFile)) {
   EXPECT_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest,
-       DISABLED_ON_IOS(ReadInt16FromFileToFloatBuffer)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToFloatBuffer \
+  DISABLED_ReadInt16FromFileToFloatBuffer
+#else
+#define MAYBE_ReadInt16FromFileToFloatBuffer ReadInt16FromFileToFloatBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToFloatBuffer) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -191,8 +220,13 @@ TEST_F(TransientFileUtilsTest,
   EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest,
-       DISABLED_ON_IOS(ReadInt16FromFileToDoubleBuffer)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToDoubleBuffer \
+  DISABLED_ReadInt16FromFileToDoubleBuffer
+#else
+#define MAYBE_ReadInt16FromFileToDoubleBuffer ReadInt16FromFileToDoubleBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToDoubleBuffer) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -232,7 +266,12 @@ TEST_F(TransientFileUtilsTest,
   EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadFloatBufferFromFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadFloatBufferFromFile DISABLED_ReadFloatBufferFromFile
+#else
+#define MAYBE_ReadFloatBufferFromFile ReadFloatBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadFloatBufferFromFile) {
   std::string test_filename = kTestFileNamef;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -269,7 +308,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadFloatBufferFromFile)) {
   EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadDoubleBufferFromFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadDoubleBufferFromFile DISABLED_ReadDoubleBufferFromFile
+#else
+#define MAYBE_ReadDoubleBufferFromFile ReadDoubleBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -306,7 +350,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ReadDoubleBufferFromFile)) {
   EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteInt16BufferToFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteInt16BufferToFile DISABLED_WriteInt16BufferToFile
+#else
+#define MAYBE_WriteInt16BufferToFile WriteInt16BufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -348,7 +397,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteInt16BufferToFile)) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteFloatBufferToFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteFloatBufferToFile DISABLED_WriteFloatBufferToFile
+#else
+#define MAYBE_WriteFloatBufferToFile WriteFloatBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -390,7 +444,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteFloatBufferToFile)) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteDoubleBufferToFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteDoubleBufferToFile DISABLED_WriteDoubleBufferToFile
+#else
+#define MAYBE_WriteDoubleBufferToFile WriteDoubleBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -432,7 +491,12 @@ TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(WriteDoubleBufferToFile)) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, DISABLED_ON_IOS(ExpectedErrorReturnValues)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ExpectedErrorReturnValues DISABLED_ExpectedErrorReturnValues
+#else
+#define MAYBE_ExpectedErrorReturnValues ExpectedErrorReturnValues
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
   std::string test_filename = kTestFileName;
 
   double value;
diff --git a/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc b/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
index 6a70a3f92c..b60077510b 100644
--- a/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
+++ b/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
@@ -19,8 +19,7 @@
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
- #include "webrtc/typedefs.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 
@@ -37,7 +36,11 @@ static const size_t kNumberOfSampleRates =
 // The files contain all the results in double precision (Little endian).
 // The audio files used with different sample rates are stored in the same
 // directory.
-TEST(TransientDetectorTest, DISABLED_ON_IOS(CorrectnessBasedOnFiles)) {
+#if defined(WEBRTC_IOS)
+TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) {
+#else
+TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
+#endif
   for (size_t i = 0; i < kNumberOfSampleRates; ++i) {
     int sample_rate_hz = kSampleRatesHz[i];
 
diff --git a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
index 506abaf203..b7b7595abf 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
@@ -19,7 +19,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc b/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
index 7c99f4f161..e4e9048f88 100644
--- a/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
+++ b/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
@@ -19,7 +19,6 @@
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -69,7 +68,11 @@ TEST(WPDTreeTest, Construction) {
 // It also writes the results in its own set of files in the out directory.
 // Matlab and output files contain all the results in double precision (Little
 // endian) appended.
-TEST(WPDTreeTest, DISABLED_ON_IOS(CorrectnessBasedOnMatlabFiles)) {
+#if defined(WEBRTC_IOS)
+TEST(WPDTreeTest, DISABLED_CorrectnessBasedOnMatlabFiles) {
+#else
+TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
+#endif
   // 10 ms at 16000 Hz.
   const size_t kTestBufferSize = 160;
   const int kLevels = 3;
diff --git a/webrtc/modules/audio_processing/typing_detection.h b/webrtc/modules/audio_processing/typing_detection.h
index 5fa6456e9e..40608f885d 100644
--- a/webrtc/modules/audio_processing/typing_detection.h
+++ b/webrtc/modules/audio_processing/typing_detection.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
index 39ec37e6ec..fce144de6b 100644
--- a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
+++ b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
@@ -18,7 +18,7 @@
 #include "webrtc/modules/audio_processing/vad/common.h"
 #include "webrtc/modules/audio_processing/vad/noise_gmm_tables.h"
 #include "webrtc/modules/audio_processing/vad/voice_gmm_tables.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/vad/standalone_vad.cc b/webrtc/modules/audio_processing/vad/standalone_vad.cc
index 468b8ff3f0..1209526a92 100644
--- a/webrtc/modules/audio_processing/vad/standalone_vad.cc
+++ b/webrtc/modules/audio_processing/vad/standalone_vad.cc
@@ -12,8 +12,8 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
index 942008e733..1d1dcc7066 100644
--- a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
+++ b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -14,9 +14,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -55,7 +54,11 @@ TEST(StandaloneVadTest, Api) {
   EXPECT_EQ(kMode, vad->mode());
 }
 
-TEST(StandaloneVadTest, DISABLED_ON_IOS(ActivityDetection)) {
+#if defined(WEBRTC_IOS)
+TEST(StandaloneVadTest, DISABLED_ActivityDetection) {
+#else
+TEST(StandaloneVadTest, ActivityDetection) {
+#endif
   rtc::scoped_ptr<StandaloneVad> vad(StandaloneVad::Create());
   const size_t kDataLength = kLength10Ms;
   int16_t data[kDataLength] = {0};
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
index 8535d1ff57..1a595597b6 100644
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
@@ -23,7 +23,7 @@ extern "C" {
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h"
 }
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
index f509af476f..a8a4ead2e3 100644
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
@@ -21,7 +21,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_processing/vad/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/voice_activity_detector.cc b/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
index ef56a3574c..fc9d103918 100644
--- a/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
+++ b/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
@@ -18,7 +18,7 @@ namespace webrtc {
 namespace {
 
 const size_t kMaxLength = 320;
-const int kNumChannels = 1;
+const size_t kNumChannels = 1;
 
 const double kDefaultVoiceValue = 1.0;
 const double kNeutralProbability = 0.5;
diff --git a/webrtc/modules/audio_processing/voice_detection_impl.cc b/webrtc/modules/audio_processing/voice_detection_impl.cc
index 374189e709..22d218c371 100644
--- a/webrtc/modules/audio_processing/voice_detection_impl.cc
+++ b/webrtc/modules/audio_processing/voice_detection_impl.cc
@@ -10,61 +10,61 @@
 
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 
-#include <assert.h>
-
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
-
-typedef VadInst Handle;
-
-namespace {
-int MapSetting(VoiceDetection::Likelihood likelihood) {
-  switch (likelihood) {
-    case VoiceDetection::kVeryLowLikelihood:
-      return 3;
-    case VoiceDetection::kLowLikelihood:
-      return 2;
-    case VoiceDetection::kModerateLikelihood:
-      return 1;
-    case VoiceDetection::kHighLikelihood:
-      return 0;
+class VoiceDetectionImpl::Vad {
+ public:
+  Vad() {
+    state_ = WebRtcVad_Create();
+    RTC_CHECK(state_);
+    int error = WebRtcVad_Init(state_);
+    RTC_DCHECK_EQ(0, error);
+  }
+  ~Vad() {
+    WebRtcVad_Free(state_);
   }
-  assert(false);
-  return -1;
+  VadInst* state() { return state_; }
+ private:
+  VadInst* state_ = nullptr;
+  RTC_DISALLOW_COPY_AND_ASSIGN(Vad);
+};
+
+VoiceDetectionImpl::VoiceDetectionImpl(rtc::CriticalSection* crit)
+    : crit_(crit) {
+  RTC_DCHECK(crit);
 }
-}  // namespace
-
-VoiceDetectionImpl::VoiceDetectionImpl(const AudioProcessing* apm,
-                                       CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    stream_has_voice_(false),
-    using_external_vad_(false),
-    likelihood_(kLowLikelihood),
-    frame_size_ms_(10),
-    frame_size_samples_(0) {}
 
 VoiceDetectionImpl::~VoiceDetectionImpl() {}
 
-int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+void VoiceDetectionImpl::Initialize(int sample_rate_hz) {
+  rtc::CritScope cs(crit_);
+  sample_rate_hz_ = sample_rate_hz;
+  rtc::scoped_ptr<Vad> new_vad;
+  if (enabled_) {
+    new_vad.reset(new Vad());
   }
+  vad_.swap(new_vad);
+  using_external_vad_ = false;
+  frame_size_samples_ =
+      static_cast<size_t>(frame_size_ms_ * sample_rate_hz_) / 1000;
+  set_likelihood(likelihood_);
+}
 
+void VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
+  }
   if (using_external_vad_) {
     using_external_vad_ = false;
-    return apm_->kNoError;
+    return;
   }
-  assert(audio->num_frames_per_band() <= 160);
 
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
   // TODO(ajm): concatenate data in frame buffer here.
-
-  int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
-                                  apm_->proc_split_sample_rate_hz(),
+  int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
                                   audio->mixed_low_pass_data(),
                                   frame_size_samples_);
   if (vad_ret == 0) {
@@ -74,103 +74,81 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
     stream_has_voice_ = true;
     audio->set_activity(AudioFrame::kVadActive);
   } else {
-    return apm_->kUnspecifiedError;
+    RTC_NOTREACHED();
   }
-
-  return apm_->kNoError;
 }
 
 int VoiceDetectionImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (enabled_ != enable) {
+    enabled_ = enable;
+    Initialize(sample_rate_hz_);
+  }
+  return AudioProcessing::kNoError;
 }
 
 bool VoiceDetectionImpl::is_enabled() const {
-  return is_component_enabled();
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 
 int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
+  rtc::CritScope cs(crit_);
   using_external_vad_ = true;
   stream_has_voice_ = has_voice;
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 bool VoiceDetectionImpl::stream_has_voice() const {
+  rtc::CritScope cs(crit_);
   // TODO(ajm): enable this assertion?
   //assert(using_external_vad_ || is_component_enabled());
   return stream_has_voice_;
 }
 
 int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (MapSetting(likelihood) == -1) {
-    return apm_->kBadParameterError;
-  }
-
+  rtc::CritScope cs(crit_);
   likelihood_ = likelihood;
-  return Configure();
+  if (enabled_) {
+    int mode = 2;
+    switch (likelihood) {
+      case VoiceDetection::kVeryLowLikelihood:
+        mode = 3;
+        break;
+      case VoiceDetection::kLowLikelihood:
+        mode = 2;
+        break;
+      case VoiceDetection::kModerateLikelihood:
+        mode = 1;
+        break;
+      case VoiceDetection::kHighLikelihood:
+        mode = 0;
+        break;
+      default:
+        RTC_NOTREACHED();
+        break;
+    }
+    int error = WebRtcVad_set_mode(vad_->state(), mode);
+    RTC_DCHECK_EQ(0, error);
+  }
+  return AudioProcessing::kNoError;
 }
 
 VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
+  rtc::CritScope cs(crit_);
   return likelihood_;
 }
 
 int VoiceDetectionImpl::set_frame_size_ms(int size) {
-  CriticalSectionScoped crit_scoped(crit_);
-  assert(size == 10); // TODO(ajm): remove when supported.
-  if (size != 10 &&
-      size != 20 &&
-      size != 30) {
-    return apm_->kBadParameterError;
-  }
-
+  rtc::CritScope cs(crit_);
+  RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported.
   frame_size_ms_ = size;
-
-  return Initialize();
+  Initialize(sample_rate_hz_);
+  return AudioProcessing::kNoError;
 }
 
 int VoiceDetectionImpl::frame_size_ms() const {
+  rtc::CritScope cs(crit_);
   return frame_size_ms_;
 }
-
-int VoiceDetectionImpl::Initialize() {
-  int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
-  }
-
-  using_external_vad_ = false;
-  frame_size_samples_ = static_cast<size_t>(
-      frame_size_ms_ * apm_->proc_split_sample_rate_hz() / 1000);
-  // TODO(ajm): intialize frame buffer here.
-
-  return apm_->kNoError;
-}
-
-void* VoiceDetectionImpl::CreateHandle() const {
-  return WebRtcVad_Create();
-}
-
-void VoiceDetectionImpl::DestroyHandle(void* handle) const {
-  WebRtcVad_Free(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::InitializeHandle(void* handle) const {
-  return WebRtcVad_Init(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::ConfigureHandle(void* handle) const {
-  return WebRtcVad_set_mode(static_cast<Handle*>(handle),
-                            MapSetting(likelihood_));
-}
-
-int VoiceDetectionImpl::num_handles_required() const {
-  return 1;
-}
-
-int VoiceDetectionImpl::GetHandleError(void* handle) const {
-  // The VAD has no get_error() function.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/voice_detection_impl.h b/webrtc/modules/audio_processing/voice_detection_impl.h
index b18808316e..0d6d8cf14a 100644
--- a/webrtc/modules/audio_processing/voice_detection_impl.h
+++ b/webrtc/modules/audio_processing/voice_detection_impl.h
@@ -11,31 +11,27 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
-class VoiceDetectionImpl : public VoiceDetection,
-                           public ProcessingComponent {
+class VoiceDetectionImpl : public VoiceDetection {
  public:
-  VoiceDetectionImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit);
-  virtual ~VoiceDetectionImpl();
+  explicit VoiceDetectionImpl(rtc::CriticalSection* crit);
+  ~VoiceDetectionImpl() override;
 
-  int ProcessCaptureAudio(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize(int sample_rate_hz);
+  void ProcessCaptureAudio(AudioBuffer* audio);
 
   // VoiceDetection implementation.
-  bool is_enabled() const override;
-
-  // ProcessingComponent implementation.
-  int Initialize() override;
-
- private:
-  // VoiceDetection implementation.
   int Enable(bool enable) override;
+  bool is_enabled() const override;
   int set_stream_has_voice(bool has_voice) override;
   bool stream_has_voice() const override;
   int set_likelihood(Likelihood likelihood) override;
@@ -43,21 +39,18 @@ class VoiceDetectionImpl : public VoiceDetection,
   int set_frame_size_ms(int size) override;
   int frame_size_ms() const override;
 
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  bool stream_has_voice_;
-  bool using_external_vad_;
-  Likelihood likelihood_;
-  int frame_size_ms_;
-  size_t frame_size_samples_;
+ private:
+  class Vad;
+  rtc::CriticalSection* const crit_;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  bool stream_has_voice_ GUARDED_BY(crit_) = false;
+  bool using_external_vad_ GUARDED_BY(crit_) = false;
+  Likelihood likelihood_ GUARDED_BY(crit_) = kLowLikelihood;
+  int frame_size_ms_ GUARDED_BY(crit_) = 10;
+  size_t frame_size_samples_ GUARDED_BY(crit_) = 0;
+  int sample_rate_hz_ GUARDED_BY(crit_) = 0;
+  rtc::scoped_ptr<Vad> vad_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VoiceDetectionImpl);
 };
 }  // namespace webrtc
 
diff --git a/webrtc/modules/bitrate_controller/BUILD.gn b/webrtc/modules/bitrate_controller/BUILD.gn
index 4ef536b572..5e3741ba93 100644
--- a/webrtc/modules/bitrate_controller/BUILD.gn
+++ b/webrtc/modules/bitrate_controller/BUILD.gn
@@ -10,7 +10,6 @@ import("../../build/webrtc.gni")
 
 source_set("bitrate_controller") {
   sources = [
-    "bitrate_allocator.cc",
     "bitrate_controller_impl.cc",
     "bitrate_controller_impl.h",
     "include/bitrate_allocator.h",
diff --git a/webrtc/modules/bitrate_controller/bitrate_allocator.cc b/webrtc/modules/bitrate_controller/bitrate_allocator.cc
deleted file mode 100644
index 0aec528cde..0000000000
--- a/webrtc/modules/bitrate_controller/bitrate_allocator.cc
+++ /dev/null
@@ -1,190 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- *
- */
-
-#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
-
-#include <algorithm>
-#include <utility>
-
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-
-namespace webrtc {
-
-// Allow packets to be transmitted in up to 2 times max video bitrate if the
-// bandwidth estimate allows it.
-const int kTransmissionMaxBitrateMultiplier = 2;
-const int kDefaultBitrateBps = 300000;
-
-BitrateAllocator::BitrateAllocator()
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      bitrate_observers_(),
-      enforce_min_bitrate_(true),
-      last_bitrate_bps_(kDefaultBitrateBps),
-      last_fraction_loss_(0),
-      last_rtt_(0) {
-}
-
-
-void BitrateAllocator::OnNetworkChanged(uint32_t bitrate,
-                                        uint8_t fraction_loss,
-                                        int64_t rtt) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  last_bitrate_bps_ = bitrate;
-  last_fraction_loss_ = fraction_loss;
-  last_rtt_ = rtt;
-  ObserverBitrateMap allocation = AllocateBitrates();
-  for (const auto& kv : allocation)
-    kv.first->OnNetworkChanged(kv.second, last_fraction_loss_, last_rtt_);
-}
-
-BitrateAllocator::ObserverBitrateMap BitrateAllocator::AllocateBitrates() {
-  if (bitrate_observers_.empty())
-    return ObserverBitrateMap();
-
-  uint32_t sum_min_bitrates = 0;
-  for (const auto& observer : bitrate_observers_)
-    sum_min_bitrates += observer.second.min_bitrate;
-  if (last_bitrate_bps_ <= sum_min_bitrates)
-    return LowRateAllocation(last_bitrate_bps_);
-  else
-    return NormalRateAllocation(last_bitrate_bps_, sum_min_bitrates);
-}
-
-int BitrateAllocator::AddBitrateObserver(BitrateObserver* observer,
-                                         uint32_t min_bitrate_bps,
-                                         uint32_t max_bitrate_bps) {
-  CriticalSectionScoped lock(crit_sect_.get());
-
-  BitrateObserverConfList::iterator it =
-      FindObserverConfigurationPair(observer);
-
-  // Allow the max bitrate to be exceeded for FEC and retransmissions.
-  // TODO(holmer): We have to get rid of this hack as it makes it difficult to
-  // properly allocate bitrate. The allocator should instead distribute any
-  // extra bitrate after all streams have maxed out.
-  max_bitrate_bps *= kTransmissionMaxBitrateMultiplier;
-  if (it != bitrate_observers_.end()) {
-    // Update current configuration.
-    it->second.min_bitrate = min_bitrate_bps;
-    it->second.max_bitrate = max_bitrate_bps;
-  } else {
-    // Add new settings.
-    bitrate_observers_.push_back(BitrateObserverConfiguration(
-        observer, BitrateConfiguration(min_bitrate_bps, max_bitrate_bps)));
-    bitrate_observers_modified_ = true;
-  }
-
-  ObserverBitrateMap allocation = AllocateBitrates();
-  int new_observer_bitrate_bps = 0;
-  for (auto& kv : allocation) {
-    kv.first->OnNetworkChanged(kv.second, last_fraction_loss_, last_rtt_);
-    if (kv.first == observer)
-      new_observer_bitrate_bps = kv.second;
-  }
-  return new_observer_bitrate_bps;
-}
-
-void BitrateAllocator::RemoveBitrateObserver(BitrateObserver* observer) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  BitrateObserverConfList::iterator it =
-      FindObserverConfigurationPair(observer);
-  if (it != bitrate_observers_.end()) {
-    bitrate_observers_.erase(it);
-    bitrate_observers_modified_ = true;
-  }
-}
-
-void BitrateAllocator::GetMinMaxBitrateSumBps(int* min_bitrate_sum_bps,
-                                              int* max_bitrate_sum_bps) const {
-  *min_bitrate_sum_bps = 0;
-  *max_bitrate_sum_bps = 0;
-
-  CriticalSectionScoped lock(crit_sect_.get());
-  for (const auto& observer : bitrate_observers_) {
-    *min_bitrate_sum_bps += observer.second.min_bitrate;
-    *max_bitrate_sum_bps += observer.second.max_bitrate;
-  }
-}
-
-BitrateAllocator::BitrateObserverConfList::iterator
-BitrateAllocator::FindObserverConfigurationPair(
-    const BitrateObserver* observer) {
-  for (auto it = bitrate_observers_.begin(); it != bitrate_observers_.end();
-       ++it) {
-    if (it->first == observer)
-      return it;
-  }
-  return bitrate_observers_.end();
-}
-
-void BitrateAllocator::EnforceMinBitrate(bool enforce_min_bitrate) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  enforce_min_bitrate_ = enforce_min_bitrate;
-}
-
-BitrateAllocator::ObserverBitrateMap BitrateAllocator::NormalRateAllocation(
-    uint32_t bitrate,
-    uint32_t sum_min_bitrates) {
-  uint32_t number_of_observers = bitrate_observers_.size();
-  uint32_t bitrate_per_observer =
-      (bitrate - sum_min_bitrates) / number_of_observers;
-  // Use map to sort list based on max bitrate.
-  ObserverSortingMap list_max_bitrates;
-  for (const auto& observer : bitrate_observers_) {
-    list_max_bitrates.insert(std::pair<uint32_t, ObserverConfiguration>(
-        observer.second.max_bitrate,
-        ObserverConfiguration(observer.first, observer.second.min_bitrate)));
-  }
-  ObserverBitrateMap allocation;
-  ObserverSortingMap::iterator max_it = list_max_bitrates.begin();
-  while (max_it != list_max_bitrates.end()) {
-    number_of_observers--;
-    uint32_t observer_allowance =
-        max_it->second.min_bitrate + bitrate_per_observer;
-    if (max_it->first < observer_allowance) {
-      // We have more than enough for this observer.
-      // Carry the remainder forward.
-      uint32_t remainder = observer_allowance - max_it->first;
-      if (number_of_observers != 0) {
-        bitrate_per_observer += remainder / number_of_observers;
-      }
-      allocation[max_it->second.observer] = max_it->first;
-    } else {
-      allocation[max_it->second.observer] = observer_allowance;
-    }
-    list_max_bitrates.erase(max_it);
-    // Prepare next iteration.
-    max_it = list_max_bitrates.begin();
-  }
-  return allocation;
-}
-
-BitrateAllocator::ObserverBitrateMap BitrateAllocator::LowRateAllocation(
-    uint32_t bitrate) {
-  ObserverBitrateMap allocation;
-  if (enforce_min_bitrate_) {
-    // Min bitrate to all observers.
-    for (const auto& observer : bitrate_observers_)
-      allocation[observer.first] = observer.second.min_bitrate;
-  } else {
-    // Allocate up to |min_bitrate| to one observer at a time, until
-    // |bitrate| is depleted.
-    uint32_t remainder = bitrate;
-    for (const auto& observer : bitrate_observers_) {
-      uint32_t allocated_bitrate =
-          std::min(remainder, observer.second.min_bitrate);
-      allocation[observer.first] = allocated_bitrate;
-      remainder -= allocated_bitrate;
-    }
-  }
-  return allocation;
-}
-}  // namespace webrtc
diff --git a/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc b/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc
deleted file mode 100644
index 4fc7e83b5b..0000000000
--- a/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc
+++ /dev/null
@@ -1,212 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-
-namespace webrtc {
-
-class TestBitrateObserver : public BitrateObserver {
- public:
-  TestBitrateObserver()
-      : last_bitrate_(0), last_fraction_loss_(0), last_rtt_(0) {}
-
-  virtual void OnNetworkChanged(uint32_t bitrate,
-                                uint8_t fraction_loss,
-                                int64_t rtt) {
-    last_bitrate_ = bitrate;
-    last_fraction_loss_ = fraction_loss;
-    last_rtt_ = rtt;
-  }
-  uint32_t last_bitrate_;
-  uint8_t last_fraction_loss_;
-  int64_t last_rtt_;
-};
-
-class BitrateAllocatorTest : public ::testing::Test {
- protected:
-  BitrateAllocatorTest() : allocator_(new BitrateAllocator()) {
-    allocator_->OnNetworkChanged(300000u, 0, 0);
-  }
-  ~BitrateAllocatorTest() {}
-
-  rtc::scoped_ptr<BitrateAllocator> allocator_;
-};
-
-TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
-  TestBitrateObserver bitrate_observer;
-  int start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer, 100000, 1500000);
-  EXPECT_EQ(300000, start_bitrate);
-  allocator_->OnNetworkChanged(200000, 0, 0);
-  EXPECT_EQ(200000u, bitrate_observer.last_bitrate_);
-
-  // TODO(pbos): Expect capping to 1.5M instead of 3M when not boosting the max
-  // bitrate for FEC/retransmissions (see TODO in BitrateAllocator).
-  allocator_->OnNetworkChanged(4000000, 0, 0);
-  EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
-  start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer, 100000, 4000000);
-  EXPECT_EQ(4000000, start_bitrate);
-
-  start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer, 100000, 1500000);
-  EXPECT_EQ(3000000, start_bitrate);
-  EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
-  allocator_->OnNetworkChanged(1500000, 0, 0);
-  EXPECT_EQ(1500000u, bitrate_observer.last_bitrate_);
-}
-
-TEST_F(BitrateAllocatorTest, TwoBitrateObserversOneRtcpObserver) {
-  TestBitrateObserver bitrate_observer_1;
-  TestBitrateObserver bitrate_observer_2;
-  int start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 300000);
-  EXPECT_EQ(300000, start_bitrate);
-  start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 300000);
-  EXPECT_EQ(200000, start_bitrate);
-
-  // Test too low start bitrate, hence lower than sum of min. Min bitrates will
-  // be allocated to all observers.
-  allocator_->OnNetworkChanged(200000, 0, 50);
-  EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
-  EXPECT_EQ(0, bitrate_observer_1.last_fraction_loss_);
-  EXPECT_EQ(50, bitrate_observer_1.last_rtt_);
-  EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);
-  EXPECT_EQ(0, bitrate_observer_2.last_fraction_loss_);
-  EXPECT_EQ(50, bitrate_observer_2.last_rtt_);
-
-  // Test a bitrate which should be distributed equally.
-  allocator_->OnNetworkChanged(500000, 0, 50);
-  const uint32_t kBitrateToShare = 500000 - 200000 - 100000;
-  EXPECT_EQ(100000u + kBitrateToShare / 2, bitrate_observer_1.last_bitrate_);
-  EXPECT_EQ(200000u + kBitrateToShare / 2, bitrate_observer_2.last_bitrate_);
-
-  // Limited by 2x max bitrates since we leave room for FEC and retransmissions.
-  allocator_->OnNetworkChanged(1500000, 0, 50);
-  EXPECT_EQ(600000u, bitrate_observer_1.last_bitrate_);
-  EXPECT_EQ(600000u, bitrate_observer_2.last_bitrate_);
-}
-
-class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
- protected:
-  BitrateAllocatorTestNoEnforceMin() : allocator_(new BitrateAllocator()) {
-    allocator_->EnforceMinBitrate(false);
-    allocator_->OnNetworkChanged(300000u, 0, 0);
-  }
-  ~BitrateAllocatorTestNoEnforceMin() {}
-
-  rtc::scoped_ptr<BitrateAllocator> allocator_;
-};
-
-// The following three tests verify that the EnforceMinBitrate() method works
-// as intended.
-TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserver) {
-  TestBitrateObserver bitrate_observer_1;
-  int start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
-  EXPECT_EQ(300000, start_bitrate);
-
-  // High REMB.
-  allocator_->OnNetworkChanged(150000, 0, 0);
-  EXPECT_EQ(150000u, bitrate_observer_1.last_bitrate_);
-
-  // Low REMB.
-  allocator_->OnNetworkChanged(10000, 0, 0);
-  EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_);
-
-  allocator_->RemoveBitrateObserver(&bitrate_observer_1);
-}
-
-TEST_F(BitrateAllocatorTestNoEnforceMin, ThreeBitrateObservers) {
-  TestBitrateObserver bitrate_observer_1;
-  TestBitrateObserver bitrate_observer_2;
-  TestBitrateObserver bitrate_observer_3;
-  // Set up the observers with min bitrates at 100000, 200000, and 300000.
-  int start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
-  EXPECT_EQ(300000, start_bitrate);
-
-  start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 400000);
-  EXPECT_EQ(200000, start_bitrate);
-  EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
-
-  start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_3, 300000, 400000);
-  EXPECT_EQ(0, start_bitrate);
-  EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
-  EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);
-
-  // High REMB. Make sure the controllers get a fair share of the surplus
-  // (i.e., what is left after each controller gets its min rate).
-  allocator_->OnNetworkChanged(690000, 0, 0);
-  // Verify that each observer gets its min rate (sum of min rates is 600000),
-  // and that the remaining 90000 is divided equally among the three.
-  uint32_t bitrate_to_share = 690000u - 100000u - 200000u - 300000u;
-  EXPECT_EQ(100000u + bitrate_to_share / 3, bitrate_observer_1.last_bitrate_);
-  EXPECT_EQ(200000u + bitrate_to_share / 3, bitrate_observer_2.last_bitrate_);
-  EXPECT_EQ(300000u + bitrate_to_share / 3, bitrate_observer_3.last_bitrate_);
-
-  // High REMB, but below the sum of min bitrates.
-  allocator_->OnNetworkChanged(500000, 0, 0);
-  // Verify that the first and second observers get their min bitrates, and the
-  // third gets the remainder.
-  EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);  // Min bitrate.
-  EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);  // Min bitrate.
-  EXPECT_EQ(200000u, bitrate_observer_3.last_bitrate_);  // Remainder.
-
-  // Low REMB.
-  allocator_->OnNetworkChanged(10000, 0, 0);
-  // Verify that the first observer gets all the rate, and the rest get zero.
-  EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_);
-  EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_);
-  EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_);
-
-  allocator_->RemoveBitrateObserver(&bitrate_observer_1);
-  allocator_->RemoveBitrateObserver(&bitrate_observer_2);
-  allocator_->RemoveBitrateObserver(&bitrate_observer_3);
-}
-
-TEST_F(BitrateAllocatorTest, ThreeBitrateObserversLowRembEnforceMin) {
-  TestBitrateObserver bitrate_observer_1;
-  TestBitrateObserver bitrate_observer_2;
-  TestBitrateObserver bitrate_observer_3;
-  int start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
-  EXPECT_EQ(300000, start_bitrate);
-
-  start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 400000);
-  EXPECT_EQ(200000, start_bitrate);
-  EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
-
-  start_bitrate =
-      allocator_->AddBitrateObserver(&bitrate_observer_3, 300000, 400000);
-  EXPECT_EQ(300000, start_bitrate);
-  EXPECT_EQ(100000, static_cast<int>(bitrate_observer_1.last_bitrate_));
-  EXPECT_EQ(200000, static_cast<int>(bitrate_observer_2.last_bitrate_));
-
-  // Low REMB. Verify that all observers still get their respective min bitrate.
-  allocator_->OnNetworkChanged(1000, 0, 0);
-  EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);  // Min cap.
-  EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);  // Min cap.
-  EXPECT_EQ(300000u, bitrate_observer_3.last_bitrate_);  // Min cap.
-
-  allocator_->RemoveBitrateObserver(&bitrate_observer_1);
-  allocator_->RemoveBitrateObserver(&bitrate_observer_2);
-  allocator_->RemoveBitrateObserver(&bitrate_observer_3);
-}
-}  // namespace webrtc
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller.gypi b/webrtc/modules/bitrate_controller/bitrate_controller.gypi
index 44c1b89ef2..3d86f2e32a 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller.gypi
+++ b/webrtc/modules/bitrate_controller/bitrate_controller.gypi
@@ -15,11 +15,9 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'bitrate_allocator.cc',
         'bitrate_controller_impl.cc',
         'bitrate_controller_impl.h',
         'include/bitrate_controller.h',
-        'include/bitrate_allocator.h',
         'send_side_bandwidth_estimation.cc',
         'send_side_bandwidth_estimation.h',
       ],
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
index 8857ee4b4a..f8fd2bb987 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
@@ -14,7 +14,7 @@
 #include <algorithm>
 #include <utility>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
@@ -129,6 +129,11 @@ void BitrateControllerImpl::SetReservedBitrate(uint32_t reserved_bitrate_bps) {
   MaybeTriggerOnNetworkChanged();
 }
 
+void BitrateControllerImpl::SetEventLog(RtcEventLog* event_log) {
+  rtc::CritScope cs(&critsect_);
+  bandwidth_estimation_.SetEventLog(event_log);
+}
+
 void BitrateControllerImpl::OnReceivedEstimatedBitrate(uint32_t bitrate) {
   {
     rtc::CritScope cs(&critsect_);
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.h b/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
index a33a0e6f04..b601899631 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
@@ -41,6 +41,8 @@ class BitrateControllerImpl : public BitrateController {
 
   void SetReservedBitrate(uint32_t reserved_bitrate_bps) override;
 
+  void SetEventLog(RtcEventLog* event_log) override;
+
   int64_t TimeUntilNextProcess() override;
   int32_t Process() override;
 
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
index 72831c78d6..2b9e589fbd 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 using webrtc::RtcpBandwidthObserver;
 using webrtc::BitrateObserver;
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_allocator.h b/webrtc/modules/bitrate_controller/include/bitrate_allocator.h
deleted file mode 100644
index 34b9ed5328..0000000000
--- a/webrtc/modules/bitrate_controller/include/bitrate_allocator.h
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- *
- *  Usage: this class will register multiple RtcpBitrateObserver's one at each
- *  RTCP module. It will aggregate the results and run one bandwidth estimation
- *  and push the result to the encoders via BitrateObserver(s).
- */
-
-#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_ALLOCATOR_H_
-#define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_ALLOCATOR_H_
-
-#include <list>
-#include <map>
-#include <utility>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-
-namespace webrtc {
-
-class BitrateObserver;
-
-class BitrateAllocator {
- public:
-  BitrateAllocator();
-
-  void OnNetworkChanged(uint32_t target_bitrate,
-                        uint8_t fraction_loss,
-                        int64_t rtt);
-
-  // Set the start and max send bitrate used by the bandwidth management.
-  //
-  // |observer| updates bitrates if already in use.
-  // |min_bitrate_bps| = 0 equals no min bitrate.
-  // |max_bitrate_bps| = 0 equals no max bitrate.
-  // Returns bitrate allocated for the bitrate observer.
-  int AddBitrateObserver(BitrateObserver* observer,
-                         uint32_t min_bitrate_bps,
-                         uint32_t max_bitrate_bps);
-
-  void RemoveBitrateObserver(BitrateObserver* observer);
-
-  void GetMinMaxBitrateSumBps(int* min_bitrate_sum_bps,
-                              int* max_bitrate_sum_bps) const;
-
-  // This method controls the behavior when the available bitrate is lower than
-  // the minimum bitrate, or the sum of minimum bitrates.
-  // When true, the bitrate will never be set lower than the minimum bitrate(s).
-  // When false, the bitrate observers will be allocated rates up to their
-  // respective minimum bitrate, satisfying one observer after the other.
-  void EnforceMinBitrate(bool enforce_min_bitrate);
-
- private:
-  struct BitrateConfiguration {
-    BitrateConfiguration(uint32_t min_bitrate, uint32_t max_bitrate)
-        : min_bitrate(min_bitrate), max_bitrate(max_bitrate) {}
-    uint32_t min_bitrate;
-    uint32_t max_bitrate;
-  };
-  struct ObserverConfiguration {
-    ObserverConfiguration(BitrateObserver* observer, uint32_t bitrate)
-        : observer(observer), min_bitrate(bitrate) {}
-    BitrateObserver* const observer;
-    uint32_t min_bitrate;
-  };
-  typedef std::pair<BitrateObserver*, BitrateConfiguration>
-      BitrateObserverConfiguration;
-  typedef std::list<BitrateObserverConfiguration> BitrateObserverConfList;
-  typedef std::multimap<uint32_t, ObserverConfiguration> ObserverSortingMap;
-  typedef std::map<BitrateObserver*, int> ObserverBitrateMap;
-
-  BitrateObserverConfList::iterator FindObserverConfigurationPair(
-      const BitrateObserver* observer) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-  ObserverBitrateMap AllocateBitrates() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-  ObserverBitrateMap NormalRateAllocation(uint32_t bitrate,
-                                          uint32_t sum_min_bitrates)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
-  ObserverBitrateMap LowRateAllocation(uint32_t bitrate)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  // Stored in a list to keep track of the insertion order.
-  BitrateObserverConfList bitrate_observers_ GUARDED_BY(crit_sect_);
-  bool bitrate_observers_modified_ GUARDED_BY(crit_sect_);
-  bool enforce_min_bitrate_ GUARDED_BY(crit_sect_);
-  uint32_t last_bitrate_bps_ GUARDED_BY(crit_sect_);
-  uint8_t last_fraction_loss_ GUARDED_BY(crit_sect_);
-  int64_t last_rtt_ GUARDED_BY(crit_sect_);
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_ALLOCATOR_H_
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index bb532886c7..d1eca8e0fe 100644
--- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -17,12 +17,13 @@
 
 #include <map>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
+class RtcEventLog;
 struct PacketInfo;
 
 class BitrateObserver {
@@ -56,6 +57,8 @@ class BitrateController : public Module {
   virtual void SetStartBitrate(int start_bitrate_bps) = 0;
   virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
 
+  virtual void SetEventLog(RtcEventLog* event_log) = 0;
+
   // Gets the available payload bandwidth in bits per second. Note that
   // this bandwidth excludes packet headers.
   virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
diff --git a/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h b/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h
new file mode 100644
index 0000000000..7a7d2e406b
--- /dev/null
+++ b/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_MOCK_MOCK_BITRATE_CONTROLLER_H_
+#define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_MOCK_MOCK_BITRATE_CONTROLLER_H_
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+
+namespace webrtc {
+namespace test {
+
+class MockBitrateObserver : public BitrateObserver {
+ public:
+  MOCK_METHOD3(OnNetworkChanged,
+               void(uint32_t bitrate_bps,
+                    uint8_t fraction_loss,
+                    int64_t rtt_ms));
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_MOCK_MOCK_BITRATE_CONTROLLER_H_
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index 9fa7b0dfc0..258c4d94de 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -13,9 +13,10 @@
 #include <cmath>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/call/rtc_event_log.h"
 
 namespace webrtc {
 namespace {
@@ -59,7 +60,8 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation()
       initially_lost_packets_(0),
       bitrate_at_2_seconds_kbps_(0),
       uma_update_state_(kNoUpdate),
-      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false) {}
+      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
+      event_log_(nullptr) {}
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
@@ -144,8 +146,8 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
   for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
     if (!rampup_uma_stats_updated_[i] &&
         bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
-      RTC_HISTOGRAM_COUNTS_100000(kUmaRampupMetrics[i].metric_name,
-                                  now_ms - first_report_time_ms_);
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(kUmaRampupMetrics[i].metric_name,
+                                         now_ms - first_report_time_ms_);
       rampup_uma_stats_updated_[i] = true;
     }
   }
@@ -154,22 +156,19 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
   } else if (uma_update_state_ == kNoUpdate) {
     uma_update_state_ = kFirstDone;
     bitrate_at_2_seconds_kbps_ = bitrate_kbps;
-    RTC_HISTOGRAM_COUNTS(
-        "WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
-    RTC_HISTOGRAM_COUNTS(
-        "WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0, 2000, 50);
-    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
-                         bitrate_at_2_seconds_kbps_,
-                         0,
-                         2000,
-                         50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitiallyLostPackets",
+                                initially_lost_packets_, 0, 100, 50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialRtt", static_cast<int>(rtt),
+                                0, 2000, 50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialBandwidthEstimate",
+                                bitrate_at_2_seconds_kbps_, 0, 2000, 50);
   } else if (uma_update_state_ == kFirstDone &&
              now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
     uma_update_state_ = kDone;
     int bitrate_diff_kbps =
         std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
-    RTC_HISTOGRAM_COUNTS(
-        "WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialVsConvergedDiff",
+                                bitrate_diff_kbps, 0, 2000, 50);
   }
 }
 
@@ -206,6 +205,11 @@ void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
       // rates).
       bitrate_ += 1000;
 
+      if (event_log_) {
+        event_log_->LogBwePacketLossEvent(
+            bitrate_, last_fraction_loss_,
+            expected_packets_since_last_loss_update_);
+      }
     } else if (last_fraction_loss_ <= 26) {
       // Loss between 2% - 10%: Do nothing.
     } else {
@@ -224,6 +228,11 @@ void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
             512.0);
         has_decreased_since_last_fraction_loss_ = true;
       }
+      if (event_log_) {
+        event_log_->LogBwePacketLossEvent(
+            bitrate_, last_fraction_loss_,
+            expected_packets_since_last_loss_update_);
+      }
     }
   }
   bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
@@ -274,4 +283,9 @@ uint32_t SendSideBandwidthEstimation::CapBitrateToThresholds(
   }
   return bitrate;
 }
+
+void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
+  event_log_ = event_log;
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 40061d3ee7..7ffb42cb54 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -15,10 +15,13 @@
 
 #include <deque>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
+
+class RtcEventLog;
+
 class SendSideBandwidthEstimation {
  public:
   SendSideBandwidthEstimation();
@@ -42,6 +45,8 @@ class SendSideBandwidthEstimation {
   void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
   int GetMinBitrate() const;
 
+  void SetEventLog(RtcEventLog* event_log);
+
  private:
   enum UmaState { kNoUpdate, kFirstDone, kDone };
 
@@ -81,6 +86,7 @@ class SendSideBandwidthEstimation {
   int bitrate_at_2_seconds_kbps_;
   UmaState uma_update_state_;
   std::vector<bool> rampup_uma_stats_updated_;
+  RtcEventLog* event_log_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
diff --git a/webrtc/modules/desktop_capture/differ_block.cc b/webrtc/modules/desktop_capture/differ_block.cc
index 0dcb377411..d4cbda3601 100644
--- a/webrtc/modules/desktop_capture/differ_block.cc
+++ b/webrtc/modules/desktop_capture/differ_block.cc
@@ -12,7 +12,7 @@
 
 #include <string.h>
 
-#include "build/build_config.h"
+#include "webrtc/typedefs.h"
 #include "webrtc/modules/desktop_capture/differ_block_sse2.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
@@ -38,7 +38,7 @@ bool BlockDifference(const uint8_t* image1,
   static bool (*diff_proc)(const uint8_t*, const uint8_t*, int) = NULL;
 
   if (!diff_proc) {
-#if defined(ARCH_CPU_ARM_FAMILY) || defined(ARCH_CPU_MIPS_FAMILY)
+#if defined(WEBRTC_ARCH_ARM_FAMILY) || defined(WEBRTC_ARCH_MIPS_FAMILY)
     // For ARM and MIPS processors, always use C version.
     // TODO(hclam): Implement a NEON version.
     diff_proc = &BlockDifference_C;
diff --git a/webrtc/modules/desktop_capture/screen_capturer_win.cc b/webrtc/modules/desktop_capture/screen_capturer_win.cc
index 1f33155656..18be4eb30b 100644
--- a/webrtc/modules/desktop_capture/screen_capturer_win.cc
+++ b/webrtc/modules/desktop_capture/screen_capturer_win.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 
+#include <utility>
+
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h"
 #include "webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h"
@@ -22,7 +24,7 @@ ScreenCapturer* ScreenCapturer::Create(const DesktopCaptureOptions& options) {
       new ScreenCapturerWinGdi(options));
 
   if (options.allow_use_magnification_api())
-    return new ScreenCapturerWinMagnifier(gdi_capturer.Pass());
+    return new ScreenCapturerWinMagnifier(std::move(gdi_capturer));
 
   return gdi_capturer.release();
 }
diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
index db40478023..066943d294 100644
--- a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
+++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
@@ -12,6 +12,8 @@
 
 #include <assert.h>
 
+#include <utility>
+
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
@@ -37,7 +39,7 @@ Atomic32 ScreenCapturerWinMagnifier::tls_index_(TLS_OUT_OF_INDEXES);
 
 ScreenCapturerWinMagnifier::ScreenCapturerWinMagnifier(
     rtc::scoped_ptr<ScreenCapturer> fallback_capturer)
-    : fallback_capturer_(fallback_capturer.Pass()),
+    : fallback_capturer_(std::move(fallback_capturer)),
       fallback_capturer_started_(false),
       callback_(NULL),
       current_screen_id_(kFullDesktopScreenId),
@@ -53,8 +55,7 @@ ScreenCapturerWinMagnifier::ScreenCapturerWinMagnifier(
       host_window_(NULL),
       magnifier_window_(NULL),
       magnifier_initialized_(false),
-      magnifier_capture_succeeded_(true) {
-}
+      magnifier_capture_succeeded_(true) {}
 
 ScreenCapturerWinMagnifier::~ScreenCapturerWinMagnifier() {
   // DestroyWindow must be called before MagUninitialize. magnifier_window_ is
@@ -236,7 +237,7 @@ BOOL ScreenCapturerWinMagnifier::OnMagImageScalingCallback(
     RECT unclipped,
     RECT clipped,
     HRGN dirty) {
-  assert(tls_index_.Value() != TLS_OUT_OF_INDEXES);
+  assert(tls_index_.Value() != static_cast<int32_t>(TLS_OUT_OF_INDEXES));
 
   ScreenCapturerWinMagnifier* owner =
       reinterpret_cast<ScreenCapturerWinMagnifier*>(
@@ -369,7 +370,7 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
     }
   }
 
-  if (tls_index_.Value() == TLS_OUT_OF_INDEXES) {
+  if (tls_index_.Value() == static_cast<int32_t>(TLS_OUT_OF_INDEXES)) {
     // More than one threads may get here at the same time, but only one will
     // write to tls_index_ using CompareExchange.
     DWORD new_tls_index = TlsAlloc();
@@ -377,7 +378,7 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
       TlsFree(new_tls_index);
   }
 
-  assert(tls_index_.Value() != TLS_OUT_OF_INDEXES);
+  assert(tls_index_.Value() != static_cast<int32_t>(TLS_OUT_OF_INDEXES));
   TlsSetValue(tls_index_.Value(), this);
 
   magnifier_initialized_ = true;
diff --git a/webrtc/modules/desktop_capture/window_capturer_win.cc b/webrtc/modules/desktop_capture/window_capturer_win.cc
index c0d71167a5..54b2768aa8 100644
--- a/webrtc/modules/desktop_capture/window_capturer_win.cc
+++ b/webrtc/modules/desktop_capture/window_capturer_win.cc
@@ -156,15 +156,16 @@ void WindowCapturerWin::Capture(const DesktopRegion& region) {
     return;
   }
 
-  // Stop capturing if the window has been closed or hidden.
-  if (!IsWindow(window_) || !IsWindowVisible(window_)) {
+  // Stop capturing if the window has been closed.
+  if (!IsWindow(window_)) {
     callback_->OnCaptureCompleted(NULL);
     return;
   }
 
-  // Return a 1x1 black frame if the window is minimized, to match the behavior
-  // on Mac.
-  if (IsIconic(window_)) {
+  // Return a 1x1 black frame if the window is minimized or invisible, to match
+  // behavior on mace. Window can be temporarily invisible during the
+  // transition of full screen mode on/off.
+  if (IsIconic(window_) || !IsWindowVisible(window_)) {
     BasicDesktopFrame* frame = new BasicDesktopFrame(DesktopSize(1, 1));
     memset(frame->data(), 0, frame->stride() * frame->size().height());
 
diff --git a/webrtc/modules/interface/module.h b/webrtc/modules/include/module.h
index ffd3065a5c..d02aa95dc8 100644
--- a/webrtc/modules/interface/module.h
+++ b/webrtc/modules/include/module.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_INTERFACE_MODULE_H_
-#define MODULES_INTERFACE_MODULE_H_
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -78,4 +78,4 @@ class RefCountedModule : public Module {
 
 }  // namespace webrtc
 
-#endif  // MODULES_INTERFACE_MODULE_H_
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_H_
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/include/module_common_types.h
index 45e93d8fad..89c5f1b49b 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/include/module_common_types.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULE_COMMON_TYPES_H
-#define MODULE_COMMON_TYPES_H
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
 
 #include <assert.h>
 #include <string.h>  // memcpy
@@ -28,7 +28,7 @@ struct RTPAudioHeader {
   uint8_t numEnergy;                  // number of valid entries in arrOfEnergy
   uint8_t arrOfEnergy[kRtpCsrcSize];  // one energy byte (0-9) per channel
   bool isCNG;                         // is this CNG
-  uint8_t channel;                    // number of channels 2 = stereo
+  size_t channel;                     // number of channels 2 = stereo
 };
 
 const int16_t kNoPictureId = -1;
@@ -38,6 +38,7 @@ const int16_t kNoTl0PicIdx = -1;
 const uint8_t kNoTemporalIdx = 0xFF;
 const uint8_t kNoSpatialIdx = 0xFF;
 const uint8_t kNoGofIdx = 0xFF;
+const uint8_t kNumVp9Buffers = 8;
 const size_t kMaxVp9RefPics = 3;
 const size_t kMaxVp9FramesInGof = 0xFF;  // 8 bits
 const size_t kMaxVp9NumberOfSpatialLayers = 8;
@@ -131,7 +132,7 @@ struct GofInfoVP9 {
       temporal_idx[i] = src.temporal_idx[i];
       temporal_up_switch[i] = src.temporal_up_switch[i];
       num_ref_pics[i] = src.num_ref_pics[i];
-      for (size_t r = 0; r < num_ref_pics[i]; ++r) {
+      for (uint8_t r = 0; r < num_ref_pics[i]; ++r) {
         pid_diff[i][r] = src.pid_diff[i][r];
       }
     }
@@ -140,8 +141,8 @@ struct GofInfoVP9 {
   size_t num_frames_in_gof;
   uint8_t temporal_idx[kMaxVp9FramesInGof];
   bool temporal_up_switch[kMaxVp9FramesInGof];
-  size_t num_ref_pics[kMaxVp9FramesInGof];
-  int16_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
+  uint8_t num_ref_pics[kMaxVp9FramesInGof];
+  uint8_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
 };
 
 struct RTPVideoHeaderVP9 {
@@ -185,9 +186,9 @@ struct RTPVideoHeaderVP9 {
 
   uint8_t gof_idx;  // Index to predefined temporal frame info in SS data.
 
-  size_t num_ref_pics;  // Number of reference pictures used by this layer
-                        // frame.
-  int16_t pid_diff[kMaxVp9RefPics];  // P_DIFF signaled to derive the PictureID
+  uint8_t num_ref_pics;  // Number of reference pictures used by this layer
+                         // frame.
+  uint8_t pid_diff[kMaxVp9RefPics];  // P_DIFF signaled to derive the PictureID
                                      // of the reference pictures.
   int16_t ref_picture_id[kMaxVp9RefPics];  // PictureID of reference pictures.
 
@@ -507,7 +508,7 @@ class AudioFrame {
   void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
                    size_t samples_per_channel, int sample_rate_hz,
                    SpeechType speech_type, VADActivity vad_activity,
-                   int num_channels = 1, uint32_t energy = -1);
+                   size_t num_channels = 1, uint32_t energy = -1);
 
   AudioFrame& Append(const AudioFrame& rhs);
 
@@ -531,7 +532,7 @@ class AudioFrame {
   int16_t data_[kMaxDataSizeSamples];
   size_t samples_per_channel_;
   int sample_rate_hz_;
-  int num_channels_;
+  size_t num_channels_;
   SpeechType speech_type_;
   VADActivity vad_activity_;
   // Note that there is no guarantee that |energy_| is correct. Any user of this
@@ -573,7 +574,7 @@ inline void AudioFrame::UpdateFrame(int id,
                                     int sample_rate_hz,
                                     SpeechType speech_type,
                                     VADActivity vad_activity,
-                                    int num_channels,
+                                    size_t num_channels,
                                     uint32_t energy) {
   id_ = id;
   timestamp_ = timestamp;
@@ -584,7 +585,6 @@ inline void AudioFrame::UpdateFrame(int id,
   num_channels_ = num_channels;
   energy_ = energy;
 
-  assert(num_channels >= 0);
   const size_t length = samples_per_channel * num_channels;
   assert(length <= kMaxDataSizeSamples);
   if (data != NULL) {
@@ -609,7 +609,6 @@ inline void AudioFrame::CopyFrom(const AudioFrame& src) {
   energy_ = src.energy_;
   interleaved_ = src.interleaved_;
 
-  assert(num_channels_ >= 0);
   const size_t length = samples_per_channel_ * num_channels_;
   assert(length <= kMaxDataSizeSamples);
   memcpy(data_, src.data_, sizeof(int16_t) * length);
@@ -807,4 +806,4 @@ class SequenceNumberUnwrapper {
 
 }  // namespace webrtc
 
-#endif  // MODULE_COMMON_TYPES_H
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn
index 05cfb4e555..2a4be728f3 100644
--- a/webrtc/modules/media_file/BUILD.gn
+++ b/webrtc/modules/media_file/BUILD.gn
@@ -14,12 +14,12 @@ config("media_file_config") {
 
 source_set("media_file") {
   sources = [
-    "interface/media_file.h",
-    "interface/media_file_defines.h",
-    "source/media_file_impl.cc",
-    "source/media_file_impl.h",
-    "source/media_file_utility.cc",
-    "source/media_file_utility.h",
+    "media_file.h",
+    "media_file_defines.h",
+    "media_file_impl.cc",
+    "media_file_impl.h",
+    "media_file_utility.cc",
+    "media_file_utility.h",
   ]
 
   if (is_win) {
diff --git a/webrtc/modules/media_file/OWNERS b/webrtc/modules/media_file/OWNERS
index beb9729e04..f6467a4161 100644
--- a/webrtc/modules/media_file/OWNERS
+++ b/webrtc/modules/media_file/OWNERS
@@ -1,5 +1,10 @@
-mflodman@webrtc.org

-perkj@webrtc.org

-niklas.enbom@webrtc.org

-

-per-file BUILD.gn=kjellander@webrtc.org

+mflodman@webrtc.org
+perkj@webrtc.org
+niklas.enbom@webrtc.org
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/media_file/media_file.gypi b/webrtc/modules/media_file/media_file.gypi
index 4ec80c3c52..94a99a22f1 100644
--- a/webrtc/modules/media_file/media_file.gypi
+++ b/webrtc/modules/media_file/media_file.gypi
@@ -17,12 +17,12 @@
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       ],
       'sources': [
-        'interface/media_file.h',
-        'interface/media_file_defines.h',
-        'source/media_file_impl.cc',
-        'source/media_file_impl.h',
-        'source/media_file_utility.cc',
-        'source/media_file_utility.h',
+        'media_file.h',
+        'media_file_defines.h',
+        'media_file_impl.cc',
+        'media_file_impl.h',
+        'media_file_utility.cc',
+        'media_file_utility.h',
       ], # source
       # TODO(jschuh): Bug 1348: fix size_t to int truncations.
       'msvs_disabled_warnings': [ 4267, ],
diff --git a/webrtc/modules/media_file/interface/media_file.h b/webrtc/modules/media_file/media_file.h
index 5b09ad4383..f6924d6bb0 100644
--- a/webrtc/modules/media_file/interface/media_file.h
+++ b/webrtc/modules/media_file/media_file.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -177,4 +177,4 @@ protected:
     virtual ~MediaFile() {}
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
diff --git a/webrtc/modules/media_file/interface/media_file_defines.h b/webrtc/modules/media_file/media_file_defines.h
index ded71a8ca7..a021a148a5 100644
--- a/webrtc/modules/media_file/interface/media_file_defines.h
+++ b/webrtc/modules/media_file/media_file_defines.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -48,4 +48,4 @@ protected:
     FileCallback() {}
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
diff --git a/webrtc/modules/media_file/source/media_file_impl.cc b/webrtc/modules/media_file/media_file_impl.cc
index 50175b86d5..abc7b9d9e0 100644
--- a/webrtc/modules/media_file/source/media_file_impl.cc
+++ b/webrtc/modules/media_file/media_file_impl.cc
@@ -11,7 +11,7 @@
 #include <assert.h>
 
 #include "webrtc/base/format_macros.h"
-#include "webrtc/modules/media_file/source/media_file_impl.h"
+#include "webrtc/modules/media_file/media_file_impl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/media_file/source/media_file_impl.h b/webrtc/modules/media_file/media_file_impl.h
index cdb54d880d..c23f514c75 100644
--- a/webrtc/modules/media_file/source/media_file_impl.h
+++ b/webrtc/modules/media_file/media_file_impl.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/media_file/source/media_file_utility.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/media_file/media_file_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
@@ -145,4 +145,4 @@ private:
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
diff --git a/webrtc/modules/media_file/source/media_file_unittest.cc b/webrtc/modules/media_file/media_file_unittest.cc
index 370d13228a..6541a8fb7c 100644
--- a/webrtc/modules/media_file/source/media_file_unittest.cc
+++ b/webrtc/modules/media_file/media_file_unittest.cc
@@ -9,10 +9,9 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 class MediaFileTest : public testing::Test {
  protected:
@@ -28,8 +27,14 @@ class MediaFileTest : public testing::Test {
   webrtc::MediaFile* media_file_;
 };
 
-TEST_F(MediaFileTest, DISABLED_ON_IOS(
-       DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError))) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  DISABLED_StartPlayingAudioFileWithoutError
+#else
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  StartPlayingAudioFileWithoutError
+#endif
+TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
   // TODO(leozwang): Use hard coded filename here, we want to
   // loop through all audio files in future
   const std::string audio_file = webrtc::test::ProjectRootPath() +
@@ -47,7 +52,12 @@ TEST_F(MediaFileTest, DISABLED_ON_IOS(
   ASSERT_EQ(0, media_file_->StopPlaying());
 }
 
-TEST_F(MediaFileTest, DISABLED_ON_IOS(WriteWavFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteWavFile DISABLED_WriteWavFile
+#else
+#define MAYBE_WriteWavFile WriteWavFile
+#endif
+TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
   // Write file.
   static const size_t kHeaderSize = 44;
   static const size_t kPayloadSize = 320;
diff --git a/webrtc/modules/media_file/source/media_file_utility.cc b/webrtc/modules/media_file/media_file_utility.cc
index 61ae442d0e..8a815cc25d 100644
--- a/webrtc/modules/media_file/source/media_file_utility.cc
+++ b/webrtc/modules/media_file/media_file_utility.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/media_file/source/media_file_utility.h"
+#include "webrtc/modules/media_file/media_file_utility.h"
 
 #include <assert.h>
 #include <sys/stat.h>
@@ -19,7 +19,7 @@
 #include "webrtc/common_audio/wav_header.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
@@ -38,8 +38,8 @@ struct WAVE_RIFF_header
 // the chunk size (16, 18 or 40 byte)
 struct WAVE_CHUNK_header
 {
-   int8_t  fmt_ckID[4];
-   int32_t fmt_ckSize;
+   int8_t   fmt_ckID[4];
+   uint32_t fmt_ckSize;
 };
 }  // unnamed namespace
 
@@ -79,15 +79,15 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
     // TODO (hellner): tmpStr and tmpStr2 seems unnecessary here.
     char tmpStr[6] = "FOUR";
     unsigned char tmpStr2[4];
-    int32_t i, len;
+    size_t i;
     bool dataFound = false;
     bool fmtFound = false;
     int8_t dummyRead;
 
 
     _dataSize = 0;
-    len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
-    if(len != sizeof(WAVE_RIFF_header))
+    int len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
+    if (len != static_cast<int>(sizeof(WAVE_RIFF_header)))
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
                      "Not a wave file (too short)");
@@ -123,14 +123,13 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
     //                 in a subroutine.
     memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
     CHUNKheaderObj.fmt_ckSize =
-        (int32_t) ((uint32_t) tmpStr2[0] +
-                         (((uint32_t)tmpStr2[1])<<8) +
-                         (((uint32_t)tmpStr2[2])<<16) +
-                         (((uint32_t)tmpStr2[3])<<24));
+        (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
+        (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
 
     memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
 
-    while ((len == sizeof(WAVE_CHUNK_header)) && (!fmtFound || !dataFound))
+    while ((len == static_cast<int>(sizeof(WAVE_CHUNK_header))) &&
+           (!fmtFound || !dataFound))
     {
         if(strcmp(tmpStr, "fmt ") == 0)
         {
@@ -164,9 +163,14 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
                 (int16_t) ((uint32_t)tmpStr2[0] +
                                  (((uint32_t)tmpStr2[1])<<8));
 
+            if (CHUNKheaderObj.fmt_ckSize < sizeof(WAVE_FMTINFO_header))
+            {
+                WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+                             "Chunk size is too small");
+                return -1;
+            }
             for (i = 0;
-                 i < (CHUNKheaderObj.fmt_ckSize -
-                      (int32_t)sizeof(WAVE_FMTINFO_header));
+                 i < CHUNKheaderObj.fmt_ckSize - sizeof(WAVE_FMTINFO_header);
                  i++)
             {
                 len = wav.Read(&dummyRead, 1);
@@ -187,7 +191,7 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
         }
         else
         {
-            for (i = 0; i < (CHUNKheaderObj.fmt_ckSize); i++)
+            for (i = 0; i < CHUNKheaderObj.fmt_ckSize; i++)
             {
                 len = wav.Read(&dummyRead, 1);
                 if(len != 1)
@@ -203,10 +207,8 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
 
         memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
         CHUNKheaderObj.fmt_ckSize =
-            (int32_t) ((uint32_t)tmpStr2[0] +
-                             (((uint32_t)tmpStr2[1])<<8) +
-                             (((uint32_t)tmpStr2[2])<<16) +
-                             (((uint32_t)tmpStr2[3])<<24));
+            (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
+            (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
 
         memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
     }
@@ -241,35 +243,17 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
     }
 
     // Calculate the number of bytes that 10 ms of audio data correspond to.
-    if(_wavFormatObj.formatTag == kWavFormatPcm)
-    {
-        // TODO (hellner): integer division for 22050 and 11025 would yield
-        //                 the same result as the else statement. Remove those
-        //                 special cases?
-        if(_wavFormatObj.nSamplesPerSec == 44100)
-        {
-            _readSizeBytes = 440 * _wavFormatObj.nChannels *
-                (_wavFormatObj.nBitsPerSample / 8);
-        } else if(_wavFormatObj.nSamplesPerSec == 22050) {
-            _readSizeBytes = 220 * _wavFormatObj.nChannels *
-                (_wavFormatObj.nBitsPerSample / 8);
-        } else if(_wavFormatObj.nSamplesPerSec == 11025) {
-            _readSizeBytes = 110 * _wavFormatObj.nChannels *
-                (_wavFormatObj.nBitsPerSample / 8);
-        } else {
-            _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
-              _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
-        }
-
-    } else {
-        _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
-            _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
-    }
+    size_t samples_per_10ms =
+        ((_wavFormatObj.formatTag == kWavFormatPcm) &&
+         (_wavFormatObj.nSamplesPerSec == 44100)) ?
+        440 : static_cast<size_t>(_wavFormatObj.nSamplesPerSec / 100);
+    _readSizeBytes = samples_per_10ms * _wavFormatObj.nChannels *
+        (_wavFormatObj.nBitsPerSample / 8);
     return 0;
 }
 
 int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec,
-                                        uint32_t channels,
+                                        size_t channels,
                                         uint32_t bitsPerSample,
                                         uint32_t formatTag)
 {
@@ -376,15 +360,15 @@ int32_t ModuleFileUtility::InitWavReading(InStream& wav,
     if(start > 0)
     {
         uint8_t dummy[WAV_MAX_BUFFER_SIZE];
-        int32_t readLength;
+        int readLength;
         if(_readSizeBytes <= WAV_MAX_BUFFER_SIZE)
         {
             while (_playoutPositionMs < start)
             {
                 readLength = wav.Read(dummy, _readSizeBytes);
-                if(readLength == _readSizeBytes)
+                if(readLength == static_cast<int>(_readSizeBytes))
                 {
-                    _readPos += readLength;
+                    _readPos += _readSizeBytes;
                     _playoutPositionMs += 10;
                 }
                 else // Must have reached EOF before start position!
@@ -406,7 +390,7 @@ int32_t ModuleFileUtility::InitWavReading(InStream& wav,
     {
         return -1;
     }
-    _bytesPerSample = _wavFormatObj.nBitsPerSample / 8;
+    _bytesPerSample = static_cast<size_t>(_wavFormatObj.nBitsPerSample / 8);
 
 
     _startPointInMs = start;
@@ -431,9 +415,9 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
         bufferSize);
 
     // The number of bytes that should be read from file.
-    const uint32_t totalBytesNeeded = _readSizeBytes;
+    const size_t totalBytesNeeded = _readSizeBytes;
     // The number of bytes that will be written to outData.
-    const uint32_t bytesRequested = (codec_info_.channels == 2) ?
+    const size_t bytesRequested = (codec_info_.channels == 2) ?
         totalBytesNeeded >> 1 : totalBytesNeeded;
     if(bufferSize < bytesRequested)
     {
@@ -472,7 +456,7 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
     // Output data is should be mono.
     if(codec_info_.channels == 2)
     {
-        for (uint32_t i = 0; i < bytesRequested / _bytesPerSample; i++)
+        for (size_t i = 0; i < bytesRequested / _bytesPerSample; i++)
         {
             // Sample value is the average of left and right buffer rounded to
             // closest integer value. Note samples can be either 1 or 2 byte.
@@ -490,7 +474,7 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
         }
         memcpy(outData, _tempData, bytesRequested);
     }
-    return bytesRequested;
+    return static_cast<int32_t>(bytesRequested);
 }
 
 int32_t ModuleFileUtility::ReadWavDataAsStereo(
@@ -534,10 +518,10 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
     }
 
     // The number of bytes that should be read from file.
-    const uint32_t totalBytesNeeded = _readSizeBytes;
+    const size_t totalBytesNeeded = _readSizeBytes;
     // The number of bytes that will be written to the left and the right
     // buffers.
-    const uint32_t bytesRequested = totalBytesNeeded >> 1;
+    const size_t bytesRequested = totalBytesNeeded >> 1;
     if(bufferSize < bytesRequested)
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
@@ -558,7 +542,7 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
     // either 1 or 2 bytes
     if(_bytesPerSample == 1)
     {
-        for (uint32_t i = 0; i < bytesRequested; i++)
+        for (size_t i = 0; i < bytesRequested; i++)
         {
             outDataLeft[i]  = _tempData[2 * i];
             outDataRight[i] = _tempData[(2 * i) + 1];
@@ -572,35 +556,29 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
             outDataRight);
 
         // Bytes requested to samples requested.
-        uint32_t sampleCount = bytesRequested >> 1;
-        for (uint32_t i = 0; i < sampleCount; i++)
+        size_t sampleCount = bytesRequested >> 1;
+        for (size_t i = 0; i < sampleCount; i++)
         {
             outLeft[i] = sampleData[2 * i];
             outRight[i] = sampleData[(2 * i) + 1];
         }
     } else {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
-                   "ReadWavStereoData: unsupported sample size %d!",
+                   "ReadWavStereoData: unsupported sample size %" PRIuS "!",
                    _bytesPerSample);
         assert(false);
         return -1;
     }
-    return bytesRequested;
+    return static_cast<int32_t>(bytesRequested);
 }
 
-int32_t ModuleFileUtility::ReadWavData(
-    InStream& wav,
-    uint8_t* buffer,
-    const uint32_t dataLengthInBytes)
+int32_t ModuleFileUtility::ReadWavData(InStream& wav,
+                                       uint8_t* buffer,
+                                       size_t dataLengthInBytes)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, dataLen= %ld)",
-        &wav,
-        buffer,
-        dataLengthInBytes);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, "
+                 "dataLen= %" PRIuS ")", &wav, buffer, dataLengthInBytes);
 
 
     if(buffer == NULL)
@@ -613,7 +591,7 @@ int32_t ModuleFileUtility::ReadWavData(
     // Make sure that a read won't return too few samples.
     // TODO (hellner): why not read the remaining bytes needed from the start
     //                 of the file?
-    if((_dataSize - _readPos) < (int32_t)dataLengthInBytes)
+    if(_dataSize < (_readPos + dataLengthInBytes))
     {
         // Rewind() being -1 may be due to the file not supposed to be looped.
         if(wav.Rewind() == -1)
@@ -685,8 +663,7 @@ int32_t ModuleFileUtility::InitWavWriting(OutStream& wav,
         return -1;
     }
     _writing = false;
-    uint32_t channels = (codecInst.channels == 0) ?
-        1 : codecInst.channels;
+    size_t channels = (codecInst.channels == 0) ? 1 : codecInst.channels;
 
     if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
     {
@@ -696,7 +673,8 @@ int32_t ModuleFileUtility::InitWavWriting(OutStream& wav,
         {
             return -1;
         }
-    }else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
+    }
+    else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
     {
         _bytesPerSample = 1;
         if(WriteWavHeader(wav, 8000, _bytesPerSample, channels, kWavFormatALaw,
@@ -729,15 +707,9 @@ int32_t ModuleFileUtility::WriteWavData(OutStream& out,
                                         const int8_t*  buffer,
                                         const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, dataLen= %" PRIuS
-        ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, "
+                 "dataLen= %" PRIuS ")", &out, buffer, dataLength);
 
     if(buffer == NULL)
     {
@@ -757,19 +729,19 @@ int32_t ModuleFileUtility::WriteWavData(OutStream& out,
 
 int32_t ModuleFileUtility::WriteWavHeader(
     OutStream& wav,
-    const uint32_t freq,
-    const uint32_t bytesPerSample,
-    const uint32_t channels,
-    const uint32_t format,
-    const uint32_t lengthInBytes)
+    uint32_t freq,
+    size_t bytesPerSample,
+    size_t channels,
+    uint32_t format,
+    size_t lengthInBytes)
 {
     // Frame size in bytes for 10 ms of audio.
     // TODO (hellner): 44.1 kHz has 440 samples frame size. Doesn't seem to
     //                 be taken into consideration here!
-    const int32_t frameSize = (freq / 100) * channels;
+    const size_t frameSize = (freq / 100) * channels;
 
     // Calculate the number of full frames that the wave file contain.
-    const int32_t dataLengthInBytes = frameSize * (lengthInBytes / frameSize);
+    const size_t dataLengthInBytes = frameSize * (lengthInBytes / frameSize);
 
     uint8_t buf[kWavHeaderSize];
     webrtc::WriteWavHeader(buf, channels, freq, static_cast<WavFormat>(format),
@@ -785,8 +757,7 @@ int32_t ModuleFileUtility::UpdateWavHeader(OutStream& wav)
     {
         return -1;
     }
-    uint32_t channels = (codec_info_.channels == 0) ?
-        1 : codec_info_.channels;
+    size_t channels = (codec_info_.channels == 0) ? 1 : codec_info_.channels;
 
     if(STR_CASE_CMP(codec_info_.plname, "L16") == 0)
     {
@@ -839,22 +810,17 @@ int32_t ModuleFileUtility::ReadPreEncodedData(
     int8_t* outData,
     const size_t bufferSize)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, outData= 0x%x, "
-        "bufferSize= %" PRIuS ")",
-        &in,
-        outData,
-        bufferSize);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, "
+                 "outData= 0x%x, bufferSize= %" PRIuS ")", &in, outData,
+                 bufferSize);
 
     if(outData == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id, "output buffer NULL");
     }
 
-    uint32_t frameLen;
+    size_t frameLen;
     uint8_t buf[64];
     // Each frame has a two byte header containing the frame length.
     int32_t res = in.Read(buf, 2);
@@ -874,12 +840,9 @@ int32_t ModuleFileUtility::ReadPreEncodedData(
     frameLen = buf[0] + buf[1] * 256;
     if(bufferSize < frameLen)
     {
-        WEBRTC_TRACE(
-            kTraceError,
-            kTraceFile,
-            _id,
-            "buffer not large enough to read %d bytes of pre-encoded data!",
-            frameLen);
+        WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+                     "buffer not large enough to read %" PRIuS " bytes of "
+                     "pre-encoded data!", frameLen);
         return -1;
     }
     return in.Read(outData, frameLen);
@@ -897,24 +860,19 @@ int32_t ModuleFileUtility::InitPreEncodedWriting(
     }
     _writing = true;
     _bytesWritten = 1;
-     out.Write(&_codecId, 1);
-     return 0;
+    out.Write(&_codecId, 1);
+    return 0;
 }
 
 int32_t ModuleFileUtility::WritePreEncodedData(
     OutStream& out,
-    const int8_t*  buffer,
+    const int8_t* buffer,
     const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WritePreEncodedData(out= 0x%x, inData= 0x%x, "
-        "dataLen= %" PRIuS ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WritePreEncodedData(out= 0x%x, "
+                 "inData= 0x%x, dataLen= %" PRIuS ")", &out, buffer,
+                 dataLength);
 
     if(buffer == NULL)
     {
@@ -945,15 +903,9 @@ int32_t ModuleFileUtility::InitCompressedReading(
     const uint32_t start,
     const uint32_t stop)
 {
-    WEBRTC_TRACE(
-        kTraceDebug,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::InitCompressedReading(in= 0x%x, start= %d,\
- stop= %d)",
-        &in,
-        start,
-        stop);
+    WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
+                 "ModuleFileUtility::InitCompressedReading(in= 0x%x, "
+                 "start= %d, stop= %d)", &in, start, stop);
 
 #if defined(WEBRTC_CODEC_ILBC)
     int16_t read_len = 0;
@@ -976,9 +928,8 @@ int32_t ModuleFileUtility::InitCompressedReading(
     if(cnt==64)
     {
         return -1;
-    } else {
-        buf[cnt]=0;
     }
+    buf[cnt]=0;
 
 #ifdef WEBRTC_CODEC_ILBC
     if(!strcmp("#!iLBC20\n", buf))
@@ -996,14 +947,11 @@ int32_t ModuleFileUtility::InitCompressedReading(
             while (_playoutPositionMs <= _startPointInMs)
             {
                 read_len = in.Read(buf, 38);
-                if(read_len == 38)
-                {
-                    _playoutPositionMs += 20;
-                }
-                else
+                if(read_len != 38)
                 {
                     return -1;
                 }
+                _playoutPositionMs += 20;
             }
         }
     }
@@ -1023,14 +971,11 @@ int32_t ModuleFileUtility::InitCompressedReading(
             while (_playoutPositionMs <= _startPointInMs)
             {
                 read_len = in.Read(buf, 50);
-                if(read_len == 50)
-                {
-                    _playoutPositionMs += 20;
-                }
-                else
+                if(read_len != 50)
                 {
                     return -1;
                 }
+                _playoutPositionMs += 20;
             }
         }
     }
@@ -1047,17 +992,11 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
                                               int8_t* outData,
                                               size_t bufferSize)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x, bytes=%"
-        PRIuS ")",
-        &in,
-        outData,
-        bufferSize);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x, "
+                 "bytes=%" PRIuS ")", &in, outData, bufferSize);
 
-    uint32_t bytesRead = 0;
+    int bytesRead = 0;
 
     if(! _reading)
     {
@@ -1069,8 +1008,8 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
     if((_codecId == kCodecIlbc20Ms) ||
         (_codecId == kCodecIlbc30Ms))
     {
-        uint32_t byteSize = 0;
-         if(_codecId == kCodecIlbc30Ms)
+        size_t byteSize = 0;
+        if(_codecId == kCodecIlbc30Ms)
         {
             byteSize = 50;
         }
@@ -1081,20 +1020,20 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
         if(bufferSize < byteSize)
         {
             WEBRTC_TRACE(kTraceError, kTraceFile, _id,
-                           "output buffer is too short to read ILBC compressed\
- data.");
+                         "output buffer is too short to read ILBC compressed "
+                         "data.");
             assert(false);
             return -1;
         }
 
         bytesRead = in.Read(outData, byteSize);
-        if(bytesRead != byteSize)
+        if(bytesRead != static_cast<int>(byteSize))
         {
             if(!in.Rewind())
             {
                 InitCompressedReading(in, _startPointInMs, _stopPointInMs);
                 bytesRead = in.Read(outData, byteSize);
-                if(bytesRead != byteSize)
+                if(bytesRead != static_cast<int>(byteSize))
                 {
                     _reading = false;
                     return -1;
@@ -1136,9 +1075,8 @@ int32_t ModuleFileUtility::InitCompressedWriting(
     const CodecInst& codecInst)
 {
     WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
-               "ModuleFileUtility::InitCompressedWriting(out= 0x%x,\
- codecName= %s)",
-               &out, codecInst.plname);
+                 "ModuleFileUtility::InitCompressedWriting(out= 0x%x, "
+                 "codecName= %s)", &out, codecInst.plname);
 
     _writing = false;
 
@@ -1177,15 +1115,9 @@ int32_t ModuleFileUtility::WriteCompressedData(
     const int8_t* buffer,
     const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x, "
-        "dataLen= %" PRIuS ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x, "
+                 "dataLen= %" PRIuS ")", &out, buffer, dataLength);
 
     if(buffer == NULL)
     {
@@ -1204,19 +1136,12 @@ int32_t ModuleFileUtility::InitPCMReading(InStream& pcm,
                                           const uint32_t stop,
                                           uint32_t freq)
 {
-    WEBRTC_TRACE(
-        kTraceInfo,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, stop=%d,\
- freq=%d)",
-        &pcm,
-        start,
-        stop,
-        freq);
+    WEBRTC_TRACE(kTraceInfo, kTraceFile, _id,
+                 "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, "
+                 "stop=%d, freq=%d)", &pcm, start, stop, freq);
 
     int8_t dummy[320];
-    int32_t read_len;
+    int read_len;
 
     _playoutPositionMs = 0;
     _startPointInMs = start;
@@ -1261,14 +1186,11 @@ int32_t ModuleFileUtility::InitPCMReading(InStream& pcm,
         while (_playoutPositionMs < _startPointInMs)
         {
             read_len = pcm.Read(dummy, _readSizeBytes);
-            if(read_len == _readSizeBytes)
+            if(read_len != static_cast<int>(_readSizeBytes))
             {
-                _playoutPositionMs += 10;
-            }
-            else // Must have reached EOF before start position!
-            {
-                return -1;
+                return -1;  // Must have reached EOF before start position!
             }
+            _playoutPositionMs += 10;
         }
     }
     _reading = true;
@@ -1279,23 +1201,17 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
                                        int8_t* outData,
                                        size_t bufferSize)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, bufSize= %"
-        PRIuS ")",
-        &pcm,
-        outData,
-        bufferSize);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, "
+                 "bufSize= %" PRIuS ")", &pcm, outData, bufferSize);
 
     if(outData == NULL)
     {
-        WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
+        WEBRTC_TRACE(kTraceError, kTraceFile, _id, "buffer NULL");
     }
 
     // Readsize for 10ms of audio data (2 bytes per sample).
-    uint32_t bytesRequested = 2 * codec_info_.plfreq / 100;
+    size_t bytesRequested = static_cast<size_t>(2 * codec_info_.plfreq / 100);
     if(bufferSize <  bytesRequested)
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
@@ -1304,8 +1220,8 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
         return -1;
     }
 
-    uint32_t bytesRead = pcm.Read(outData, bytesRequested);
-    if(bytesRead < bytesRequested)
+    int bytesRead = pcm.Read(outData, bytesRequested);
+    if(bytesRead < static_cast<int>(bytesRequested))
     {
         if(pcm.Rewind() == -1)
         {
@@ -1320,9 +1236,9 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
             }
             else
             {
-                int32_t rest = bytesRequested - bytesRead;
-                int32_t len = pcm.Read(&(outData[bytesRead]), rest);
-                if(len == rest)
+                size_t rest = bytesRequested - bytesRead;
+                int len = pcm.Read(&(outData[bytesRead]), rest);
+                if(len == static_cast<int>(rest))
                 {
                     bytesRead += len;
                 }
@@ -1334,7 +1250,7 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
             if(bytesRead <= 0)
             {
                 WEBRTC_TRACE(kTraceError, kTraceFile, _id,
-                        "ReadPCMData: Failed to rewind audio file.");
+                             "ReadPCMData: Failed to rewind audio file.");
                 return -1;
             }
         }
@@ -1343,7 +1259,7 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
     if(bytesRead <= 0)
     {
         WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
-                   "ReadPCMData: end of file");
+                     "ReadPCMData: end of file");
         return -1;
     }
     _playoutPositionMs += 10;
@@ -1414,15 +1330,9 @@ int32_t ModuleFileUtility::WritePCMData(OutStream& out,
                                         const int8_t*  buffer,
                                         const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, dataLen= %" PRIuS
-        ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, "
+                 "dataLen= %" PRIuS ")", &out, buffer, dataLength);
 
     if(buffer == NULL)
     {
@@ -1585,7 +1495,7 @@ int32_t ModuleFileUtility::FileDurationMs(const char* fileName,
         case kFileFormatCompressedFile:
         {
             int32_t cnt = 0;
-            int32_t read_len = 0;
+            int read_len = 0;
             char buf[64];
             do
             {
@@ -1642,15 +1552,8 @@ int32_t ModuleFileUtility::FileDurationMs(const char* fileName,
 uint32_t ModuleFileUtility::PlayoutPositionMs()
 {
     WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
-               "ModuleFileUtility::PlayoutPosition()");
+                 "ModuleFileUtility::PlayoutPosition()");
 
-    if(_reading)
-    {
-        return _playoutPositionMs;
-    }
-    else
-    {
-        return 0;
-    }
+    return _reading ? _playoutPositionMs : 0;
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/media_file/source/media_file_utility.h b/webrtc/modules/media_file/media_file_utility.h
index 2823ceca8a..bc2fa5a2f0 100644
--- a/webrtc/modules/media_file/source/media_file_utility.h
+++ b/webrtc/modules/media_file/media_file_utility.h
@@ -9,13 +9,13 @@
  */
 
 // Note: the class cannot be used for reading and writing at the same time.
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
 
 #include <stdio.h>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
 
 namespace webrtc {
 class InStream;
@@ -176,11 +176,11 @@ public:
 
 private:
     // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
-    enum{WAV_MAX_BUFFER_SIZE = 480*2*2};
+    static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2;
 
 
     int32_t InitWavCodec(uint32_t samplesPerSec,
-                         uint32_t channels,
+                         size_t channels,
                          uint32_t bitsPerSample,
                          uint32_t formatTag);
 
@@ -194,16 +194,16 @@ private:
     // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
     // lengthInBytes is the number of bytes the audio samples are using up.
     int32_t WriteWavHeader(OutStream& stream,
-                           const uint32_t freqInHz,
-                           const uint32_t bytesPerSample,
-                           const uint32_t channels,
-                           const uint32_t format,
-                           const uint32_t lengthInBytes);
+                           uint32_t freqInHz,
+                           size_t bytesPerSample,
+                           size_t channels,
+                           uint32_t format,
+                           size_t lengthInBytes);
 
     // Put dataLengthInBytes of audio data from stream into the audioBuffer.
     // The return value is the number of bytes written to audioBuffer.
     int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
-                        const uint32_t dataLengthInBytes);
+                        size_t dataLengthInBytes);
 
     // Update the current audio codec being used for reading or writing
     // according to codecInst.
@@ -254,10 +254,10 @@ private:
     // TODO (hellner): why store multiple formats. Just store either codec_info_
     //                 or _wavFormatObj and supply conversion functions.
     WAVE_FMTINFO_header _wavFormatObj;
-    int32_t _dataSize;      // Chunk size if reading a WAV file
+    size_t _dataSize;      // Chunk size if reading a WAV file
     // Number of bytes to read. I.e. frame size in bytes. May be multiple
     // chunks if reading WAV.
-    int32_t _readSizeBytes;
+    size_t _readSizeBytes;
 
     int32_t _id;
 
@@ -270,8 +270,8 @@ private:
     MediaFileUtility_CodecType _codecId;
 
     // The amount of bytes, on average, used for one audio sample.
-    int32_t  _bytesPerSample;
-    int32_t  _readPos;
+    size_t _bytesPerSample;
+    size_t _readPos;
 
     // Only reading or writing can be enabled, not both.
     bool _reading;
@@ -281,4 +281,4 @@ private:
     uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
diff --git a/webrtc/modules/media_file/source/OWNERS b/webrtc/modules/media_file/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/webrtc/modules/media_file/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/webrtc/modules/module_common_types_unittest.cc b/webrtc/modules/module_common_types_unittest.cc
index bc0b7a1a5b..acd58476a1 100644
--- a/webrtc/modules/module_common_types_unittest.cc
+++ b/webrtc/modules/module_common_types_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index f3ac454c19..a7dc0ffe53 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -49,352 +49,6 @@
       },
       'targets': [
         {
-          'target_name': 'modules_unittests',
-          'type': '<(gtest_target_type)',
-          'defines': [
-            '<@(audio_coding_defines)',
-          ],
-          'dependencies': [
-            'acm_receive_test',
-            'acm_send_test',
-            'audio_coding_module',
-            'audio_conference_mixer',
-            'audio_device'  ,
-            'audio_processing',
-            'audioproc_test_utils',
-            'bitrate_controller',
-            'bwe_simulator',
-            'cng',
-            'desktop_capture',
-            'isac_fix',
-            'media_file',
-            'neteq',
-            'neteq_test_support',
-            'neteq_unittest_tools',
-            'paced_sender',
-            'pcm16b',  # Needed by NetEq tests.
-            'red',
-            'remote_bitrate_estimator',
-            'rtp_rtcp',
-            'video_codecs_test_framework',
-            'video_processing',
-            'webrtc_utility',
-            'webrtc_video_coding',
-            '<@(neteq_dependencies)',
-            '<(DEPTH)/testing/gmock.gyp:gmock',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
-            '<(webrtc_root)/modules/modules.gyp:video_capture',
-            '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
-            '<(webrtc_root)/modules/video_coding/codecs/vp9/vp9.gyp:webrtc_vp9',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/test/test.gyp:frame_generator',
-            '<(webrtc_root)/test/test.gyp:rtp_test_utils',
-            '<(webrtc_root)/test/test.gyp:test_support_main',
-            '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
-            '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
-          ],
-          'sources': [
-            'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
-            'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc',
-            'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc',
-            'audio_coding/main/acm2/call_statistics_unittest.cc',
-            'audio_coding/main/acm2/codec_owner_unittest.cc',
-            'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
-            'audio_coding/codecs/cng/cng_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/filters_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/transform_unittest.cc',
-            'audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc',
-            'audio_coding/codecs/isac/main/source/isac_unittest.cc',
-            'audio_coding/codecs/isac/unittest.cc',
-            'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
-            'audio_coding/codecs/opus/opus_unittest.cc',
-            'audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc',
-            'audio_coding/neteq/audio_classifier_unittest.cc',
-            'audio_coding/neteq/audio_multi_vector_unittest.cc',
-            'audio_coding/neteq/audio_vector_unittest.cc',
-            'audio_coding/neteq/background_noise_unittest.cc',
-            'audio_coding/neteq/buffer_level_filter_unittest.cc',
-            'audio_coding/neteq/comfort_noise_unittest.cc',
-            'audio_coding/neteq/decision_logic_unittest.cc',
-            'audio_coding/neteq/decoder_database_unittest.cc',
-            'audio_coding/neteq/delay_manager_unittest.cc',
-            'audio_coding/neteq/delay_peak_detector_unittest.cc',
-            'audio_coding/neteq/dsp_helper_unittest.cc',
-            'audio_coding/neteq/dtmf_buffer_unittest.cc',
-            'audio_coding/neteq/dtmf_tone_generator_unittest.cc',
-            'audio_coding/neteq/expand_unittest.cc',
-            'audio_coding/neteq/merge_unittest.cc',
-            'audio_coding/neteq/nack_unittest.cc',
-            'audio_coding/neteq/neteq_external_decoder_unittest.cc',
-            'audio_coding/neteq/neteq_impl_unittest.cc',
-            'audio_coding/neteq/neteq_network_stats_unittest.cc',
-            'audio_coding/neteq/neteq_stereo_unittest.cc',
-            'audio_coding/neteq/neteq_unittest.cc',
-            'audio_coding/neteq/normal_unittest.cc',
-            'audio_coding/neteq/packet_buffer_unittest.cc',
-            'audio_coding/neteq/payload_splitter_unittest.cc',
-            'audio_coding/neteq/post_decode_vad_unittest.cc',
-            'audio_coding/neteq/random_vector_unittest.cc',
-            'audio_coding/neteq/sync_buffer_unittest.cc',
-            'audio_coding/neteq/timestamp_scaler_unittest.cc',
-            'audio_coding/neteq/time_stretch_unittest.cc',
-            'audio_coding/neteq/mock/mock_audio_decoder.h',
-            'audio_coding/neteq/mock/mock_audio_vector.h',
-            'audio_coding/neteq/mock/mock_buffer_level_filter.h',
-            'audio_coding/neteq/mock/mock_decoder_database.h',
-            'audio_coding/neteq/mock/mock_delay_manager.h',
-            'audio_coding/neteq/mock/mock_delay_peak_detector.h',
-            'audio_coding/neteq/mock/mock_dtmf_buffer.h',
-            'audio_coding/neteq/mock/mock_dtmf_tone_generator.h',
-            'audio_coding/neteq/mock/mock_expand.h',
-            'audio_coding/neteq/mock/mock_external_decoder_pcm16b.h',
-            'audio_coding/neteq/mock/mock_packet_buffer.h',
-            'audio_coding/neteq/mock/mock_payload_splitter.h',
-            'audio_coding/neteq/tools/input_audio_file_unittest.cc',
-            'audio_coding/neteq/tools/packet_unittest.cc',
-            'audio_conference_mixer/test/audio_conference_mixer_unittest.cc',
-            'audio_device/fine_audio_buffer_unittest.cc',
-            'audio_processing/aec/echo_cancellation_unittest.cc',
-            'audio_processing/aec/system_delay_unittest.cc',
-            'audio_processing/agc/agc_manager_direct_unittest.cc',
-            # TODO(ajm): Fix to match new interface.
-            # 'audio_processing/agc/agc_unittest.cc',
-            'audio_processing/agc/histogram_unittest.cc',
-            'audio_processing/agc/mock_agc.h',
-            'audio_processing/beamformer/array_util_unittest.cc',
-            'audio_processing/beamformer/complex_matrix_unittest.cc',
-            'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',
-            'audio_processing/beamformer/matrix_unittest.cc',
-            'audio_processing/beamformer/mock_nonlinear_beamformer.h',
-            'audio_processing/beamformer/nonlinear_beamformer_unittest.cc',
-            'audio_processing/echo_cancellation_impl_unittest.cc',
-            'audio_processing/intelligibility/intelligibility_enhancer_unittest.cc',
-            'audio_processing/intelligibility/intelligibility_utils_unittest.cc',
-            'audio_processing/splitting_filter_unittest.cc',
-            'audio_processing/transient/dyadic_decimator_unittest.cc',
-            'audio_processing/transient/file_utils.cc',
-            'audio_processing/transient/file_utils.h',
-            'audio_processing/transient/file_utils_unittest.cc',
-            'audio_processing/transient/moving_moments_unittest.cc',
-            'audio_processing/transient/transient_detector_unittest.cc',
-            'audio_processing/transient/transient_suppressor_unittest.cc',
-            'audio_processing/transient/wpd_node_unittest.cc',
-            'audio_processing/transient/wpd_tree_unittest.cc',
-            'audio_processing/utility/delay_estimator_unittest.cc',
-            'audio_processing/vad/gmm_unittest.cc',
-            'audio_processing/vad/pitch_based_vad_unittest.cc',
-            'audio_processing/vad/pitch_internal_unittest.cc',
-            'audio_processing/vad/pole_zero_filter_unittest.cc',
-            'audio_processing/vad/standalone_vad_unittest.cc',
-            'audio_processing/vad/vad_audio_proc_unittest.cc',
-            'audio_processing/vad/vad_circular_buffer_unittest.cc',
-            'audio_processing/vad/voice_activity_detector_unittest.cc',
-            'bitrate_controller/bitrate_allocator_unittest.cc',
-            'bitrate_controller/bitrate_controller_unittest.cc',
-            'bitrate_controller/send_side_bandwidth_estimation_unittest.cc',
-            'desktop_capture/desktop_and_cursor_composer_unittest.cc',
-            'desktop_capture/desktop_region_unittest.cc',
-            'desktop_capture/differ_block_unittest.cc',
-            'desktop_capture/differ_unittest.cc',
-            'desktop_capture/mouse_cursor_monitor_unittest.cc',
-            'desktop_capture/screen_capturer_helper_unittest.cc',
-            'desktop_capture/screen_capturer_mac_unittest.cc',
-            'desktop_capture/screen_capturer_mock_objects.h',
-            'desktop_capture/screen_capturer_unittest.cc',
-            'desktop_capture/window_capturer_unittest.cc',
-            'desktop_capture/win/cursor_unittest.cc',
-            'desktop_capture/win/cursor_unittest_resources.h',
-            'desktop_capture/win/cursor_unittest_resources.rc',
-            'media_file/source/media_file_unittest.cc',
-            'module_common_types_unittest.cc',
-            'pacing/bitrate_prober_unittest.cc',
-            'pacing/paced_sender_unittest.cc',
-            'pacing/packet_router_unittest.cc',
-            'remote_bitrate_estimator/bwe_simulations.cc',
-            'remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h',
-            'remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h',
-            'remote_bitrate_estimator/inter_arrival_unittest.cc',
-            'remote_bitrate_estimator/overuse_detector_unittest.cc',
-            'remote_bitrate_estimator/rate_statistics_unittest.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
-            'remote_bitrate_estimator/remote_estimator_proxy_unittest.cc',
-            'remote_bitrate_estimator/send_time_history_unittest.cc',
-            'remote_bitrate_estimator/test/bwe_test_framework_unittest.cc',
-            'remote_bitrate_estimator/test/bwe_unittest.cc',
-            'remote_bitrate_estimator/test/metric_recorder_unittest.cc',
-            'remote_bitrate_estimator/test/estimators/nada_unittest.cc',
-            'remote_bitrate_estimator/transport_feedback_adapter_unittest.cc',
-            'rtp_rtcp/source/mock/mock_rtp_payload_strategy.h',
-            'rtp_rtcp/source/byte_io_unittest.cc',
-            'rtp_rtcp/source/fec_receiver_unittest.cc',
-            'rtp_rtcp/source/fec_test_helper.cc',
-            'rtp_rtcp/source/fec_test_helper.h',
-            'rtp_rtcp/source/h264_sps_parser_unittest.cc',
-            'rtp_rtcp/source/h264_bitstream_parser_unittest.cc',
-            'rtp_rtcp/source/nack_rtx_unittest.cc',
-            'rtp_rtcp/source/packet_loss_stats_unittest.cc',
-            'rtp_rtcp/source/producer_fec_unittest.cc',
-            'rtp_rtcp/source/receive_statistics_unittest.cc',
-            'rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc',
-            'rtp_rtcp/source/rtcp_format_remb_unittest.cc',
-            'rtp_rtcp/source/rtcp_packet_unittest.cc',
-            'rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc',
-            'rtp_rtcp/source/rtcp_receiver_unittest.cc',
-            'rtp_rtcp/source/rtcp_sender_unittest.cc',
-            'rtp_rtcp/source/rtcp_utility_unittest.cc',
-            'rtp_rtcp/source/rtp_fec_unittest.cc',
-            'rtp_rtcp/source/rtp_format_h264_unittest.cc',
-            'rtp_rtcp/source/rtp_format_vp8_test_helper.cc',
-            'rtp_rtcp/source/rtp_format_vp8_test_helper.h',
-            'rtp_rtcp/source/rtp_format_vp8_unittest.cc',
-            'rtp_rtcp/source/rtp_format_vp9_unittest.cc',
-            'rtp_rtcp/source/rtp_packet_history_unittest.cc',
-            'rtp_rtcp/source/rtp_payload_registry_unittest.cc',
-            'rtp_rtcp/source/rtp_rtcp_impl_unittest.cc',
-            'rtp_rtcp/source/rtp_header_extension_unittest.cc',
-            'rtp_rtcp/source/rtp_sender_unittest.cc',
-            'rtp_rtcp/source/vp8_partition_aggregator_unittest.cc',
-            'rtp_rtcp/test/testAPI/test_api.cc',
-            'rtp_rtcp/test/testAPI/test_api.h',
-            'rtp_rtcp/test/testAPI/test_api_audio.cc',
-            'rtp_rtcp/test/testAPI/test_api_rtcp.cc',
-            'rtp_rtcp/test/testAPI/test_api_video.cc',
-            'utility/source/audio_frame_operations_unittest.cc',
-            'utility/source/file_player_unittests.cc',
-            'utility/source/process_thread_impl_unittest.cc',
-            'video_coding/codecs/test/packet_manipulator_unittest.cc',
-            'video_coding/codecs/test/stats_unittest.cc',
-            'video_coding/codecs/test/videoprocessor_unittest.cc',
-            'video_coding/codecs/vp8/default_temporal_layers_unittest.cc',
-            'video_coding/codecs/vp8/reference_picture_selection_unittest.cc',
-            'video_coding/codecs/vp8/screenshare_layers_unittest.cc',
-            'video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc',
-            'video_coding/codecs/vp8/simulcast_unittest.cc',
-            'video_coding/codecs/vp8/simulcast_unittest.h',
-            'video_coding/main/interface/mock/mock_vcm_callbacks.h',
-            'video_coding/main/source/decoding_state_unittest.cc',
-            'video_coding/main/source/jitter_buffer_unittest.cc',
-            'video_coding/main/source/jitter_estimator_tests.cc',
-            'video_coding/main/source/media_optimization_unittest.cc',
-            'video_coding/main/source/receiver_unittest.cc',
-            'video_coding/main/source/session_info_unittest.cc',
-            'video_coding/main/source/timing_unittest.cc',
-            'video_coding/main/source/video_coding_robustness_unittest.cc',
-            'video_coding/main/source/video_receiver_unittest.cc',
-            'video_coding/main/source/video_sender_unittest.cc',
-            'video_coding/main/source/qm_select_unittest.cc',
-            'video_coding/main/source/test/stream_generator.cc',
-            'video_coding/main/source/test/stream_generator.h',
-            'video_coding/utility/quality_scaler_unittest.cc',
-            'video_processing/main/test/unit_test/brightness_detection_test.cc',
-            'video_processing/main/test/unit_test/content_metrics_test.cc',
-            'video_processing/main/test/unit_test/deflickering_test.cc',
-            'video_processing/main/test/unit_test/video_processing_unittest.cc',
-            'video_processing/main/test/unit_test/video_processing_unittest.h',
-          ],
-          'conditions': [
-            ['enable_bwe_test_logging==1', {
-              'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
-            }, {
-              'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
-              'sources!': [
-                'remote_bitrate_estimator/test/bwe_test_logging.cc'
-              ],
-            }],
-            # Run screen/window capturer tests only on platforms where they are
-            # supported.
-            ['desktop_capture_supported==0', {
-              'sources!': [
-                'desktop_capture/desktop_and_cursor_composer_unittest.cc',
-                'desktop_capture/mouse_cursor_monitor_unittest.cc',
-                'desktop_capture/screen_capturer_helper_unittest.cc',
-                'desktop_capture/screen_capturer_mac_unittest.cc',
-                'desktop_capture/screen_capturer_mock_objects.h',
-                'desktop_capture/screen_capturer_unittest.cc',
-                'desktop_capture/window_capturer_unittest.cc',
-              ],
-            }],
-            ['prefer_fixed_point==1', {
-              'defines': [ 'WEBRTC_AUDIOPROC_FIXED_PROFILE' ],
-            }, {
-              'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ],
-            }],
-            ['enable_protobuf==1', {
-              'defines': [
-                'WEBRTC_AUDIOPROC_DEBUG_DUMP',
-              ],
-              'dependencies': [
-                'audioproc_protobuf_utils',
-                'audioproc_unittest_proto',
-              ],
-              'sources': [
-                'audio_processing/audio_processing_impl_unittest.cc',
-                'audio_processing/test/audio_processing_unittest.cc',
-                'audio_processing/test/test_utils.h',
-              ],
-            }],
-            ['build_libvpx==1', {
-              'dependencies': [
-                '<(libvpx_dir)/libvpx.gyp:libvpx_new',
-              ],
-            }],
-            ['OS=="android"', {
-              'dependencies': [
-                '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
-              ],
-              # Need to disable error due to the line in
-              # base/android/jni_android.h triggering it:
-              # const BASE_EXPORT jobject GetApplicationContext()
-              # error: type qualifiers ignored on function return type
-              'cflags': [
-                '-Wno-ignored-qualifiers',
-              ],
-              'sources': [
-                'audio_device/android/audio_device_unittest.cc',
-                'audio_device/android/audio_manager_unittest.cc',
-                'audio_device/android/ensure_initialized.cc',
-                'audio_device/android/ensure_initialized.h',
-              ],
-            }],
-            ['OS=="ios"', {
-              'sources': [
-                'video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc',
-                'audio_device/ios/audio_device_unittest_ios.cc',
-              ],
-              'mac_bundle_resources': [
-                '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
-                '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
-                '<(DEPTH)/resources/audio_coding/teststereo32kHz.pcm',
-                '<(DEPTH)/resources/audio_device/audio_short16.pcm',
-                '<(DEPTH)/resources/audio_device/audio_short44.pcm',
-                '<(DEPTH)/resources/audio_device/audio_short48.pcm',
-                '<(DEPTH)/resources/audio_processing/agc/agc_no_circular_buffer.dat',
-                '<(DEPTH)/resources/audio_processing/agc/agc_pitch_gain.dat',
-                '<(DEPTH)/resources/audio_processing/agc/agc_pitch_lag.dat',
-                '<(DEPTH)/resources/audio_processing/agc/agc_spectral_peak.dat',
-                '<(DEPTH)/resources/audio_processing/agc/agc_voicing_prob.dat',
-                '<(DEPTH)/resources/audio_processing/agc/agc_with_circular_buffer.dat',
-                '<(DEPTH)/resources/short_mixed_mono_48.dat',
-                '<(DEPTH)/resources/short_mixed_mono_48.pcm',
-                '<(DEPTH)/resources/short_mixed_stereo_48.dat',
-                '<(DEPTH)/resources/short_mixed_stereo_48.pcm',
-              ],
-            }],
-          ],
-          # Disable warnings to enable Win64 build, issue 1323.
-          'msvs_disabled_warnings': [
-            4267,  # size_t to int truncation.
-          ],
-        },
-        {
           'target_name': 'modules_tests',
           'type': '<(gtest_target_type)',
           'dependencies': [
@@ -417,25 +71,24 @@
             '<@(audio_coding_defines)',
           ],
           'sources': [
-            'audio_coding/main/test/APITest.cc',
-            'audio_coding/main/test/Channel.cc',
-            'audio_coding/main/test/EncodeDecodeTest.cc',
-            'audio_coding/main/test/PCMFile.cc',
-            'audio_coding/main/test/PacketLossTest.cc',
-            'audio_coding/main/test/RTPFile.cc',
-            'audio_coding/main/test/SpatialAudio.cc',
-            'audio_coding/main/test/TestAllCodecs.cc',
-            'audio_coding/main/test/TestRedFec.cc',
-            'audio_coding/main/test/TestStereo.cc',
-            'audio_coding/main/test/TestVADDTX.cc',
-            'audio_coding/main/test/Tester.cc',
-            'audio_coding/main/test/TimedTrace.cc',
-            'audio_coding/main/test/TwoWayCommunication.cc',
-            'audio_coding/main/test/iSACTest.cc',
-            'audio_coding/main/test/initial_delay_unittest.cc',
-            'audio_coding/main/test/opus_test.cc',
-            'audio_coding/main/test/target_delay_unittest.cc',
-            'audio_coding/main/test/utility.cc',
+            'audio_coding/test/APITest.cc',
+            'audio_coding/test/Channel.cc',
+            'audio_coding/test/EncodeDecodeTest.cc',
+            'audio_coding/test/PCMFile.cc',
+            'audio_coding/test/PacketLossTest.cc',
+            'audio_coding/test/RTPFile.cc',
+            'audio_coding/test/SpatialAudio.cc',
+            'audio_coding/test/TestAllCodecs.cc',
+            'audio_coding/test/TestRedFec.cc',
+            'audio_coding/test/TestStereo.cc',
+            'audio_coding/test/TestVADDTX.cc',
+            'audio_coding/test/Tester.cc',
+            'audio_coding/test/TimedTrace.cc',
+            'audio_coding/test/TwoWayCommunication.cc',
+            'audio_coding/test/iSACTest.cc',
+            'audio_coding/test/opus_test.cc',
+            'audio_coding/test/target_delay_unittest.cc',
+            'audio_coding/test/utility.cc',
             'rtp_rtcp/test/testFec/test_fec.cc',
             'video_coding/codecs/test/videoprocessor_integrationtest.cc',
             'video_coding/codecs/vp8/test/vp8_impl_unittest.cc',
@@ -450,6 +103,379 @@
         },
       ],
       'conditions': [
+        # Does not compile on iOS for ia32 or x64: webrtc:4755.
+        ['OS!="ios" or target_arch=="arm" or target_arch=="arm64"', {
+          'targets': [
+            {
+              'target_name': 'modules_unittests',
+              'type': '<(gtest_target_type)',
+              'defines': [
+                '<@(audio_coding_defines)',
+              ],
+              'dependencies': [
+                'acm_receive_test',
+                'acm_send_test',
+                'audio_coding_module',
+                'audio_conference_mixer',
+                'audio_device'  ,
+                'audio_processing',
+                'audioproc_test_utils',
+                'bitrate_controller',
+                'bwe_simulator',
+                'cng',
+                'desktop_capture',
+                'isac_fix',
+                'media_file',
+                'neteq',
+                'neteq_test_support',
+                'neteq_unittest_tools',
+                'paced_sender',
+                'pcm16b',  # Needed by NetEq tests.
+                'red',
+                'remote_bitrate_estimator',
+                'rtp_rtcp',
+                'video_codecs_test_framework',
+                'video_processing',
+                'webrtc_utility',
+                'webrtc_video_coding',
+                '<@(neteq_dependencies)',
+                '<(DEPTH)/testing/gmock.gyp:gmock',
+                '<(DEPTH)/testing/gtest.gyp:gtest',
+                '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+                '<(webrtc_root)/common.gyp:webrtc_common',
+                '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+                '<(webrtc_root)/common_video/common_video.gyp:common_video',
+                '<(webrtc_root)/modules/modules.gyp:video_capture',
+                '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
+                '<(webrtc_root)/modules/video_coding/codecs/vp9/vp9.gyp:webrtc_vp9',
+                '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+                '<(webrtc_root)/test/test.gyp:fake_video_frames',
+                '<(webrtc_root)/test/test.gyp:rtp_test_utils',
+                '<(webrtc_root)/test/test.gyp:test_support_main',
+                '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
+                '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
+              ],
+              'sources': [
+                'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
+                'audio_coding/acm2/acm_receiver_unittest_oldapi.cc',
+                'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc',
+                'audio_coding/acm2/call_statistics_unittest.cc',
+                'audio_coding/acm2/codec_manager_unittest.cc',
+                'audio_coding/acm2/initial_delay_manager_unittest.cc',
+                'audio_coding/acm2/rent_a_codec_unittest.cc',
+                'audio_coding/codecs/cng/cng_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/filters_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/transform_unittest.cc',
+                'audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc',
+                'audio_coding/codecs/isac/main/source/isac_unittest.cc',
+                'audio_coding/codecs/isac/unittest.cc',
+                'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
+                'audio_coding/codecs/opus/opus_unittest.cc',
+                'audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc',
+                'audio_coding/neteq/audio_classifier_unittest.cc',
+                'audio_coding/neteq/audio_multi_vector_unittest.cc',
+                'audio_coding/neteq/audio_vector_unittest.cc',
+                'audio_coding/neteq/background_noise_unittest.cc',
+                'audio_coding/neteq/buffer_level_filter_unittest.cc',
+                'audio_coding/neteq/comfort_noise_unittest.cc',
+                'audio_coding/neteq/decision_logic_unittest.cc',
+                'audio_coding/neteq/decoder_database_unittest.cc',
+                'audio_coding/neteq/delay_manager_unittest.cc',
+                'audio_coding/neteq/delay_peak_detector_unittest.cc',
+                'audio_coding/neteq/dsp_helper_unittest.cc',
+                'audio_coding/neteq/dtmf_buffer_unittest.cc',
+                'audio_coding/neteq/dtmf_tone_generator_unittest.cc',
+                'audio_coding/neteq/expand_unittest.cc',
+                'audio_coding/neteq/merge_unittest.cc',
+                'audio_coding/neteq/nack_unittest.cc',
+                'audio_coding/neteq/neteq_external_decoder_unittest.cc',
+                'audio_coding/neteq/neteq_impl_unittest.cc',
+                'audio_coding/neteq/neteq_network_stats_unittest.cc',
+                'audio_coding/neteq/neteq_stereo_unittest.cc',
+                'audio_coding/neteq/neteq_unittest.cc',
+                'audio_coding/neteq/normal_unittest.cc',
+                'audio_coding/neteq/packet_buffer_unittest.cc',
+                'audio_coding/neteq/payload_splitter_unittest.cc',
+                'audio_coding/neteq/post_decode_vad_unittest.cc',
+                'audio_coding/neteq/random_vector_unittest.cc',
+                'audio_coding/neteq/sync_buffer_unittest.cc',
+                'audio_coding/neteq/timestamp_scaler_unittest.cc',
+                'audio_coding/neteq/time_stretch_unittest.cc',
+                'audio_coding/neteq/mock/mock_audio_decoder.h',
+                'audio_coding/neteq/mock/mock_audio_vector.h',
+                'audio_coding/neteq/mock/mock_buffer_level_filter.h',
+                'audio_coding/neteq/mock/mock_decoder_database.h',
+                'audio_coding/neteq/mock/mock_delay_manager.h',
+                'audio_coding/neteq/mock/mock_delay_peak_detector.h',
+                'audio_coding/neteq/mock/mock_dtmf_buffer.h',
+                'audio_coding/neteq/mock/mock_dtmf_tone_generator.h',
+                'audio_coding/neteq/mock/mock_expand.h',
+                'audio_coding/neteq/mock/mock_external_decoder_pcm16b.h',
+                'audio_coding/neteq/mock/mock_packet_buffer.h',
+                'audio_coding/neteq/mock/mock_payload_splitter.h',
+                'audio_coding/neteq/tools/input_audio_file_unittest.cc',
+                'audio_coding/neteq/tools/packet_unittest.cc',
+                'audio_conference_mixer/test/audio_conference_mixer_unittest.cc',
+                'audio_device/fine_audio_buffer_unittest.cc',
+                'audio_processing/aec/echo_cancellation_unittest.cc',
+                'audio_processing/aec/system_delay_unittest.cc',
+                'audio_processing/agc/agc_manager_direct_unittest.cc',
+                # TODO(ajm): Fix to match new interface.
+                # 'audio_processing/agc/agc_unittest.cc',
+                'audio_processing/agc/histogram_unittest.cc',
+                'audio_processing/agc/mock_agc.h',
+                'audio_processing/beamformer/array_util_unittest.cc',
+                'audio_processing/beamformer/complex_matrix_unittest.cc',
+                'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',
+                'audio_processing/beamformer/matrix_unittest.cc',
+                'audio_processing/beamformer/mock_nonlinear_beamformer.h',
+                'audio_processing/beamformer/nonlinear_beamformer_unittest.cc',
+                'audio_processing/echo_cancellation_impl_unittest.cc',
+                'audio_processing/intelligibility/intelligibility_enhancer_unittest.cc',
+                'audio_processing/intelligibility/intelligibility_utils_unittest.cc',
+                'audio_processing/splitting_filter_unittest.cc',
+                'audio_processing/transient/dyadic_decimator_unittest.cc',
+                'audio_processing/transient/file_utils.cc',
+                'audio_processing/transient/file_utils.h',
+                'audio_processing/transient/file_utils_unittest.cc',
+                'audio_processing/transient/moving_moments_unittest.cc',
+                'audio_processing/transient/transient_detector_unittest.cc',
+                'audio_processing/transient/transient_suppressor_unittest.cc',
+                'audio_processing/transient/wpd_node_unittest.cc',
+                'audio_processing/transient/wpd_tree_unittest.cc',
+                'audio_processing/utility/delay_estimator_unittest.cc',
+                'audio_processing/vad/gmm_unittest.cc',
+                'audio_processing/vad/pitch_based_vad_unittest.cc',
+                'audio_processing/vad/pitch_internal_unittest.cc',
+                'audio_processing/vad/pole_zero_filter_unittest.cc',
+                'audio_processing/vad/standalone_vad_unittest.cc',
+                'audio_processing/vad/vad_audio_proc_unittest.cc',
+                'audio_processing/vad/vad_circular_buffer_unittest.cc',
+                'audio_processing/vad/voice_activity_detector_unittest.cc',
+                'bitrate_controller/bitrate_controller_unittest.cc',
+                'bitrate_controller/send_side_bandwidth_estimation_unittest.cc',
+                'desktop_capture/desktop_and_cursor_composer_unittest.cc',
+                'desktop_capture/desktop_region_unittest.cc',
+                'desktop_capture/differ_block_unittest.cc',
+                'desktop_capture/differ_unittest.cc',
+                'desktop_capture/mouse_cursor_monitor_unittest.cc',
+                'desktop_capture/screen_capturer_helper_unittest.cc',
+                'desktop_capture/screen_capturer_mac_unittest.cc',
+                'desktop_capture/screen_capturer_mock_objects.h',
+                'desktop_capture/screen_capturer_unittest.cc',
+                'desktop_capture/window_capturer_unittest.cc',
+                'desktop_capture/win/cursor_unittest.cc',
+                'desktop_capture/win/cursor_unittest_resources.h',
+                'desktop_capture/win/cursor_unittest_resources.rc',
+                'media_file/media_file_unittest.cc',
+                'module_common_types_unittest.cc',
+                'pacing/bitrate_prober_unittest.cc',
+                'pacing/paced_sender_unittest.cc',
+                'pacing/packet_router_unittest.cc',
+                'remote_bitrate_estimator/bwe_simulations.cc',
+                'remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h',
+                'remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h',
+                'remote_bitrate_estimator/inter_arrival_unittest.cc',
+                'remote_bitrate_estimator/overuse_detector_unittest.cc',
+                'remote_bitrate_estimator/rate_statistics_unittest.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
+                'remote_bitrate_estimator/remote_estimator_proxy_unittest.cc',
+                'remote_bitrate_estimator/send_time_history_unittest.cc',
+                'remote_bitrate_estimator/test/bwe_test_framework_unittest.cc',
+                'remote_bitrate_estimator/test/bwe_unittest.cc',
+                'remote_bitrate_estimator/test/metric_recorder_unittest.cc',
+                'remote_bitrate_estimator/test/estimators/nada_unittest.cc',
+                'remote_bitrate_estimator/transport_feedback_adapter_unittest.cc',
+                'rtp_rtcp/source/mock/mock_rtp_payload_strategy.h',
+                'rtp_rtcp/source/byte_io_unittest.cc',
+                'rtp_rtcp/source/fec_receiver_unittest.cc',
+                'rtp_rtcp/source/fec_test_helper.cc',
+                'rtp_rtcp/source/fec_test_helper.h',
+                'rtp_rtcp/source/h264_sps_parser_unittest.cc',
+                'rtp_rtcp/source/h264_bitstream_parser_unittest.cc',
+                'rtp_rtcp/source/nack_rtx_unittest.cc',
+                'rtp_rtcp/source/packet_loss_stats_unittest.cc',
+                'rtp_rtcp/source/producer_fec_unittest.cc',
+                'rtp_rtcp/source/receive_statistics_unittest.cc',
+                'rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc',
+                'rtp_rtcp/source/rtcp_format_remb_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/app_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/bye_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/nack_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/pli_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/report_block_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/sli_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc',
+                'rtp_rtcp/source/rtcp_receiver_unittest.cc',
+                'rtp_rtcp/source/rtcp_sender_unittest.cc',
+                'rtp_rtcp/source/rtcp_utility_unittest.cc',
+                'rtp_rtcp/source/rtp_fec_unittest.cc',
+                'rtp_rtcp/source/rtp_format_h264_unittest.cc',
+                'rtp_rtcp/source/rtp_format_vp8_test_helper.cc',
+                'rtp_rtcp/source/rtp_format_vp8_test_helper.h',
+                'rtp_rtcp/source/rtp_format_vp8_unittest.cc',
+                'rtp_rtcp/source/rtp_format_vp9_unittest.cc',
+                'rtp_rtcp/source/rtp_packet_history_unittest.cc',
+                'rtp_rtcp/source/rtp_payload_registry_unittest.cc',
+                'rtp_rtcp/source/rtp_rtcp_impl_unittest.cc',
+                'rtp_rtcp/source/rtp_header_extension_unittest.cc',
+                'rtp_rtcp/source/rtp_sender_unittest.cc',
+                'rtp_rtcp/source/time_util_unittest.cc',
+                'rtp_rtcp/source/vp8_partition_aggregator_unittest.cc',
+                'rtp_rtcp/test/testAPI/test_api.cc',
+                'rtp_rtcp/test/testAPI/test_api.h',
+                'rtp_rtcp/test/testAPI/test_api_audio.cc',
+                'rtp_rtcp/test/testAPI/test_api_rtcp.cc',
+                'rtp_rtcp/test/testAPI/test_api_video.cc',
+                'utility/source/audio_frame_operations_unittest.cc',
+                'utility/source/file_player_unittests.cc',
+                'utility/source/process_thread_impl_unittest.cc',
+                'video_coding/codecs/test/packet_manipulator_unittest.cc',
+                'video_coding/codecs/test/stats_unittest.cc',
+                'video_coding/codecs/test/videoprocessor_unittest.cc',
+                'video_coding/codecs/vp8/default_temporal_layers_unittest.cc',
+                'video_coding/codecs/vp8/reference_picture_selection_unittest.cc',
+                'video_coding/codecs/vp8/screenshare_layers_unittest.cc',
+                'video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc',
+                'video_coding/codecs/vp8/simulcast_unittest.cc',
+                'video_coding/codecs/vp8/simulcast_unittest.h',
+                'video_coding/codecs/vp9/screenshare_layers_unittest.cc',
+                'video_coding/include/mock/mock_vcm_callbacks.h',
+                'video_coding/decoding_state_unittest.cc',
+                'video_coding/jitter_buffer_unittest.cc',
+                'video_coding/jitter_estimator_tests.cc',
+                'video_coding/media_optimization_unittest.cc',
+                'video_coding/receiver_unittest.cc',
+                'video_coding/session_info_unittest.cc',
+                'video_coding/timing_unittest.cc',
+                'video_coding/video_coding_robustness_unittest.cc',
+                'video_coding/video_receiver_unittest.cc',
+                'video_coding/video_sender_unittest.cc',
+                'video_coding/qm_select_unittest.cc',
+                'video_coding/test/stream_generator.cc',
+                'video_coding/test/stream_generator.h',
+                'video_coding/utility/quality_scaler_unittest.cc',
+                'video_processing/test/brightness_detection_test.cc',
+                'video_processing/test/content_metrics_test.cc',
+                'video_processing/test/deflickering_test.cc',
+                'video_processing/test/denoiser_test.cc',
+                'video_processing/test/video_processing_unittest.cc',
+                'video_processing/test/video_processing_unittest.h',
+              ],
+              'conditions': [
+                ['enable_bwe_test_logging==1', {
+                  'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
+                }, {
+                  'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
+                  'sources!': [
+                    'remote_bitrate_estimator/test/bwe_test_logging.cc'
+                  ],
+                }],
+                # Run screen/window capturer tests only on platforms where they are
+                # supported.
+                ['desktop_capture_supported==0', {
+                  'sources!': [
+                    'desktop_capture/desktop_and_cursor_composer_unittest.cc',
+                    'desktop_capture/mouse_cursor_monitor_unittest.cc',
+                    'desktop_capture/screen_capturer_helper_unittest.cc',
+                    'desktop_capture/screen_capturer_mac_unittest.cc',
+                    'desktop_capture/screen_capturer_mock_objects.h',
+                    'desktop_capture/screen_capturer_unittest.cc',
+                    'desktop_capture/window_capturer_unittest.cc',
+                  ],
+                }],
+                ['prefer_fixed_point==1', {
+                  'defines': [ 'WEBRTC_AUDIOPROC_FIXED_PROFILE' ],
+                }, {
+                  'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ],
+                }],
+                ['enable_protobuf==1', {
+                  'defines': [
+                    'WEBRTC_AUDIOPROC_DEBUG_DUMP',
+                    'WEBRTC_NETEQ_UNITTEST_BITEXACT',
+                  ],
+                  'dependencies': [
+                    'audioproc_protobuf_utils',
+                    'audioproc_unittest_proto',
+                    'neteq_unittest_proto',
+                  ],
+                  'sources': [
+                    'audio_processing/audio_processing_impl_locking_unittest.cc',
+                    'audio_processing/audio_processing_impl_unittest.cc',
+                    'audio_processing/test/audio_processing_unittest.cc',
+                    'audio_processing/test/debug_dump_test.cc',
+                    'audio_processing/test/test_utils.h',
+                  ],
+                }],
+                ['build_libvpx==1', {
+                  'dependencies': [
+                    '<(libvpx_dir)/libvpx.gyp:libvpx_new',
+                  ],
+                }],
+                ['OS=="android"', {
+                  'dependencies': [
+                    '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
+                  ],
+                  # Need to disable error due to the line in
+                  # base/android/jni_android.h triggering it:
+                  # const BASE_EXPORT jobject GetApplicationContext()
+                  # error: type qualifiers ignored on function return type
+                  'cflags': [
+                    '-Wno-ignored-qualifiers',
+                  ],
+                  'sources': [
+                    'audio_device/android/audio_device_unittest.cc',
+                    'audio_device/android/audio_manager_unittest.cc',
+                    'audio_device/android/ensure_initialized.cc',
+                    'audio_device/android/ensure_initialized.h',
+                  ],
+                }],
+                ['OS=="ios"', {
+                  'sources': [
+                    'video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc',
+                    'audio_device/ios/audio_device_unittest_ios.cc',
+                  ],
+                  'mac_bundle_resources': [
+                    '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
+                    '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
+                    '<(DEPTH)/resources/audio_coding/teststereo32kHz.pcm',
+                    '<(DEPTH)/resources/audio_device/audio_short16.pcm',
+                    '<(DEPTH)/resources/audio_device/audio_short44.pcm',
+                    '<(DEPTH)/resources/audio_device/audio_short48.pcm',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_no_circular_buffer.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_pitch_gain.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_pitch_lag.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_spectral_peak.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_voicing_prob.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_with_circular_buffer.dat',
+                    '<(DEPTH)/resources/short_mixed_mono_48.dat',
+                    '<(DEPTH)/resources/short_mixed_mono_48.pcm',
+                    '<(DEPTH)/resources/short_mixed_stereo_48.dat',
+                    '<(DEPTH)/resources/short_mixed_stereo_48.pcm',
+                  ],
+                }],
+              ],
+              # Disable warnings to enable Win64 build, issue 1323.
+              'msvs_disabled_warnings': [
+                4267,  # size_t to int truncation.
+              ],
+            },
+          ],
+        }],
         ['OS=="android"', {
           'targets': [
             {
diff --git a/webrtc/modules/modules_unittests.isolate b/webrtc/modules/modules_unittests.isolate
index ba444df8c1..d988821af0 100644
--- a/webrtc/modules/modules_unittests.isolate
+++ b/webrtc/modules/modules_unittests.isolate
@@ -27,10 +27,7 @@
           '<(DEPTH)/resources/audio_coding/neteq4_universal_ref.pcm',
           '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_32.pcm',
           '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_64.pcm',
-          '<(DEPTH)/resources/audio_coding/neteq_network_stats.dat',
-          '<(DEPTH)/resources/audio_coding/neteq_rtcp_stats.dat',
           '<(DEPTH)/resources/audio_coding/neteq_universal_new.rtp',
-          '<(DEPTH)/resources/audio_coding/neteq_universal_ref.pcm',
           '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
           '<(DEPTH)/resources/audio_coding/speech_mono_32_48kHz.pcm',
           '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn
index 3e478c1e76..0354c64fcb 100644
--- a/webrtc/modules/pacing/BUILD.gn
+++ b/webrtc/modules/pacing/BUILD.gn
@@ -10,10 +10,10 @@ source_set("pacing") {
   sources = [
     "bitrate_prober.cc",
     "bitrate_prober.h",
-    "include/paced_sender.h",
-    "include/packet_router.h",
     "paced_sender.cc",
+    "paced_sender.h",
     "packet_router.cc",
+    "packet_router.h",
   ]
 
   configs += [ "../..:common_config" ]
diff --git a/webrtc/modules/pacing/bitrate_prober.cc b/webrtc/modules/pacing/bitrate_prober.cc
index bbbe54f54e..41ad5fa11a 100644
--- a/webrtc/modules/pacing/bitrate_prober.cc
+++ b/webrtc/modules/pacing/bitrate_prober.cc
@@ -15,8 +15,8 @@
 #include <limits>
 #include <sstream>
 
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/pacing/include/mock/mock_paced_sender.h b/webrtc/modules/pacing/mock/mock_paced_sender.h
index b2cefdff8b..01d5f6a6e9 100644
--- a/webrtc/modules/pacing/include/mock/mock_paced_sender.h
+++ b/webrtc/modules/pacing/mock/mock_paced_sender.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_PACING_INCLUDE_MOCK_MOCK_PACED_SENDER_H_
-#define WEBRTC_MODULES_PACING_INCLUDE_MOCK_MOCK_PACED_SENDER_H_
+#ifndef WEBRTC_MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
+#define WEBRTC_MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
 
 #include <vector>
 
-#include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -35,4 +35,4 @@ class MockPacedSender : public PacedSender {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_PACING_INCLUDE_MOCK_MOCK_PACED_SENDER_H_
+#endif  // WEBRTC_MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 5d7ae17b23..121f860c7d 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -8,20 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/pacing/include/paced_sender.h"
-
-#include <assert.h>
+#include "webrtc/modules/pacing/paced_sender.h"
 
 #include <map>
 #include <queue>
 #include <set>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/bitrate_prober.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 namespace {
 // Time limit in milliseconds between packet bursts.
@@ -33,6 +32,9 @@ const int64_t kMaxIntervalTimeMs = 30;
 
 }  // namespace
 
+// TODO(sprang): Move at least PacketQueue and MediaBudget out to separate
+// files, so that we can more easily test them.
+
 namespace webrtc {
 namespace paced_sender {
 struct Packet {
@@ -86,13 +88,19 @@ struct Comparator {
 // Class encapsulating a priority queue with some extensions.
 class PacketQueue {
  public:
-  PacketQueue() : bytes_(0) {}
+  explicit PacketQueue(Clock* clock)
+      : bytes_(0),
+        clock_(clock),
+        queue_time_sum_(0),
+        time_last_updated_(clock_->TimeInMilliseconds()) {}
   virtual ~PacketQueue() {}
 
   void Push(const Packet& packet) {
-    if (!AddToDupeSet(packet)) {
+    if (!AddToDupeSet(packet))
       return;
-    }
+
+    UpdateQueueTime(packet.enqueue_time_ms);
+
     // Store packet in list, use pointers in priority queue for cheaper moves.
     // Packets have a handle to its own iterator in the list, for easy removal
     // when popping from queue.
@@ -114,7 +122,11 @@ class PacketQueue {
   void FinalizePop(const Packet& packet) {
     RemoveFromDupeSet(packet);
     bytes_ -= packet.bytes;
+    queue_time_sum_ -= (time_last_updated_ - packet.enqueue_time_ms);
     packet_list_.erase(packet.this_it);
+    RTC_DCHECK_EQ(packet_list_.size(), prio_queue_.size());
+    if (packet_list_.empty())
+      RTC_DCHECK_EQ(0u, queue_time_sum_);
   }
 
   bool Empty() const { return prio_queue_.empty(); }
@@ -123,13 +135,29 @@ class PacketQueue {
 
   uint64_t SizeInBytes() const { return bytes_; }
 
-  int64_t OldestEnqueueTime() const {
-    std::list<Packet>::const_reverse_iterator it = packet_list_.rbegin();
+  int64_t OldestEnqueueTimeMs() const {
+    auto it = packet_list_.rbegin();
     if (it == packet_list_.rend())
       return 0;
     return it->enqueue_time_ms;
   }
 
+  void UpdateQueueTime(int64_t timestamp_ms) {
+    RTC_DCHECK_GE(timestamp_ms, time_last_updated_);
+    int64_t delta = timestamp_ms - time_last_updated_;
+    // Use packet packet_list_.size() not prio_queue_.size() here, as there
+    // might be an outstanding element popped from prio_queue_ currently in the
+    // SendPacket() call, while packet_list_ will always be correct.
+    queue_time_sum_ += delta * packet_list_.size();
+    time_last_updated_ = timestamp_ms;
+  }
+
+  int64_t AverageQueueTimeMs() const {
+    if (prio_queue_.empty())
+      return 0;
+    return queue_time_sum_ / packet_list_.size();
+  }
+
  private:
   // Try to add a packet to the set of ssrc/seqno identifiers currently in the
   // queue. Return true if inserted, false if this is a duplicate.
@@ -147,7 +175,7 @@ class PacketQueue {
 
   void RemoveFromDupeSet(const Packet& packet) {
     SsrcSeqNoMap::iterator it = dupe_map_.find(packet.ssrc);
-    assert(it != dupe_map_.end());
+    RTC_DCHECK(it != dupe_map_.end());
     it->second.erase(packet.sequence_number);
     if (it->second.empty()) {
       dupe_map_.erase(it);
@@ -165,6 +193,9 @@ class PacketQueue {
   // Map<ssrc, set<seq_no> >, for checking duplicates.
   typedef std::map<uint32_t, std::set<uint16_t> > SsrcSeqNoMap;
   SsrcSeqNoMap dupe_map_;
+  Clock* const clock_;
+  int64_t queue_time_sum_;
+  int64_t time_last_updated_;
 };
 
 class IntervalBudget {
@@ -209,6 +240,7 @@ class IntervalBudget {
 };
 }  // namespace paced_sender
 
+const int64_t PacedSender::kMaxQueueLengthMs = 2000;
 const float PacedSender::kDefaultPaceMultiplier = 2.5f;
 
 PacedSender::PacedSender(Clock* clock,
@@ -225,8 +257,9 @@ PacedSender::PacedSender(Clock* clock,
       padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)),
       prober_(new BitrateProber()),
       bitrate_bps_(1000 * bitrate_kbps),
+      max_bitrate_kbps_(max_bitrate_kbps),
       time_last_update_us_(clock->TimeInMicroseconds()),
-      packets_(new paced_sender::PacketQueue()),
+      packets_(new paced_sender::PacketQueue(clock)),
       packet_counter_(0) {
   UpdateBytesPerInterval(kMinPacketLimitMs);
 }
@@ -244,7 +277,7 @@ void PacedSender::Resume() {
 }
 
 void PacedSender::SetProbingEnabled(bool enabled) {
-  assert(packet_counter_ == 0);
+  RTC_CHECK_EQ(0u, packet_counter_);
   probing_enabled_ = enabled;
 }
 
@@ -252,9 +285,12 @@ void PacedSender::UpdateBitrate(int bitrate_kbps,
                                 int max_bitrate_kbps,
                                 int min_bitrate_kbps) {
   CriticalSectionScoped cs(critsect_.get());
-  media_budget_->set_target_rate_kbps(max_bitrate_kbps);
+  // Don't set media bitrate here as it may be boosted in order to meet max
+  // queue time constraint. Just update max_bitrate_kbps_ and let media_budget_
+  // be updated in Process().
   padding_budget_->set_target_rate_kbps(min_bitrate_kbps);
   bitrate_bps_ = 1000 * bitrate_kbps;
+  max_bitrate_kbps_ = max_bitrate_kbps;
 }
 
 void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
@@ -265,25 +301,23 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
                                bool retransmission) {
   CriticalSectionScoped cs(critsect_.get());
 
-  if (probing_enabled_ && !prober_->IsProbing()) {
+  if (probing_enabled_ && !prober_->IsProbing())
     prober_->SetEnabled(true);
-  }
   prober_->MaybeInitializeProbe(bitrate_bps_);
 
-  if (capture_time_ms < 0) {
-    capture_time_ms = clock_->TimeInMilliseconds();
-  }
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (capture_time_ms < 0)
+    capture_time_ms = now_ms;
 
-  packets_->Push(paced_sender::Packet(
-      priority, ssrc, sequence_number, capture_time_ms,
-      clock_->TimeInMilliseconds(), bytes, retransmission, packet_counter_++));
+  packets_->Push(paced_sender::Packet(priority, ssrc, sequence_number,
+                                      capture_time_ms, now_ms, bytes,
+                                      retransmission, packet_counter_++));
 }
 
 int64_t PacedSender::ExpectedQueueTimeMs() const {
   CriticalSectionScoped cs(critsect_.get());
-  int target_rate = media_budget_->target_rate_kbps();
-  assert(target_rate > 0);
-  return static_cast<int64_t>(packets_->SizeInBytes() * 8 / target_rate);
+  RTC_DCHECK_GT(max_bitrate_kbps_, 0);
+  return static_cast<int64_t>(packets_->SizeInBytes() * 8 / max_bitrate_kbps_);
 }
 
 size_t PacedSender::QueueSizePackets() const {
@@ -294,20 +328,25 @@ size_t PacedSender::QueueSizePackets() const {
 int64_t PacedSender::QueueInMs() const {
   CriticalSectionScoped cs(critsect_.get());
 
-  int64_t oldest_packet = packets_->OldestEnqueueTime();
+  int64_t oldest_packet = packets_->OldestEnqueueTimeMs();
   if (oldest_packet == 0)
     return 0;
 
   return clock_->TimeInMilliseconds() - oldest_packet;
 }
 
+int64_t PacedSender::AverageQueueTimeMs() {
+  CriticalSectionScoped cs(critsect_.get());
+  packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
+  return packets_->AverageQueueTimeMs();
+}
+
 int64_t PacedSender::TimeUntilNextProcess() {
   CriticalSectionScoped cs(critsect_.get());
   if (prober_->IsProbing()) {
     int64_t ret = prober_->TimeUntilNextProbe(clock_->TimeInMilliseconds());
-    if (ret >= 0) {
+    if (ret >= 0)
       return ret;
-    }
   }
   int64_t elapsed_time_us = clock_->TimeInMicroseconds() - time_last_update_us_;
   int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
@@ -319,27 +358,42 @@ int32_t PacedSender::Process() {
   CriticalSectionScoped cs(critsect_.get());
   int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
   time_last_update_us_ = now_us;
-  if (paused_)
-    return 0;
-  if (elapsed_time_ms > 0) {
+  int target_bitrate_kbps = max_bitrate_kbps_;
+  // TODO(holmer): Remove the !paused_ check when issue 5307 has been fixed.
+  if (!paused_ && elapsed_time_ms > 0) {
+    size_t queue_size_bytes = packets_->SizeInBytes();
+    if (queue_size_bytes > 0) {
+      // Assuming equal size packets and input/output rate, the average packet
+      // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
+      // time constraint shall be met. Determine bitrate needed for that.
+      packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
+      int64_t avg_time_left_ms = std::max<int64_t>(
+          1, kMaxQueueLengthMs - packets_->AverageQueueTimeMs());
+      int min_bitrate_needed_kbps =
+          static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
+      if (min_bitrate_needed_kbps > target_bitrate_kbps)
+        target_bitrate_kbps = min_bitrate_needed_kbps;
+    }
+
+    media_budget_->set_target_rate_kbps(target_bitrate_kbps);
+
     int64_t delta_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms);
     UpdateBytesPerInterval(delta_time_ms);
   }
   while (!packets_->Empty()) {
-    if (media_budget_->bytes_remaining() == 0 && !prober_->IsProbing()) {
+    if (media_budget_->bytes_remaining() == 0 && !prober_->IsProbing())
       return 0;
-    }
 
     // Since we need to release the lock in order to send, we first pop the
     // element from the priority queue but keep it in storage, so that we can
     // reinsert it if send fails.
     const paced_sender::Packet& packet = packets_->BeginPop();
+
     if (SendPacket(packet)) {
       // Send succeeded, remove it from the queue.
       packets_->FinalizePop(packet);
-      if (prober_->IsProbing()) {
+      if (prober_->IsProbing())
         return 0;
-      }
     } else {
       // Send failed, put it back into the queue.
       packets_->CancelPop(packet);
@@ -347,14 +401,16 @@ int32_t PacedSender::Process() {
     }
   }
 
-  if (!packets_->Empty())
+  // TODO(holmer): Remove the paused_ check when issue 5307 has been fixed.
+  if (paused_ || !packets_->Empty())
     return 0;
 
   size_t padding_needed;
-  if (prober_->IsProbing())
+  if (prober_->IsProbing()) {
     padding_needed = prober_->RecommendedPacketSize();
-  else
+  } else {
     padding_needed = padding_budget_->bytes_remaining();
+  }
 
   if (padding_needed > 0)
     SendPadding(static_cast<size_t>(padding_needed));
@@ -362,6 +418,11 @@ int32_t PacedSender::Process() {
 }
 
 bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
+  // TODO(holmer): Because of this bug issue 5307 we have to send audio
+  // packets even when the pacer is paused. Here we assume audio packets are
+  // always high priority and that they are the only high priority packets.
+  if (paused_ && packet.priority != kHighPriority)
+    return false;
   critsect_->Leave();
   const bool success = callback_->TimeToSendPacket(packet.ssrc,
                                                    packet.sequence_number,
@@ -369,7 +430,9 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
                                                    packet.retransmission);
   critsect_->Enter();
 
-  if (success) {
+  // TODO(holmer): High priority packets should only be accounted for if we are
+  // allocating bandwidth for audio.
+  if (success && packet.priority != kHighPriority) {
     // Update media bytes sent.
     prober_->PacketSent(clock_->TimeInMilliseconds(), packet.bytes);
     media_budget_->UseBudget(packet.bytes);
diff --git a/webrtc/modules/pacing/include/paced_sender.h b/webrtc/modules/pacing/paced_sender.h
index f142f55173..62e794fdbc 100644
--- a/webrtc/modules/pacing/include/paced_sender.h
+++ b/webrtc/modules/pacing/paced_sender.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
-#define WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
+#ifndef WEBRTC_MODULES_PACING_PACED_SENDER_H_
+#define WEBRTC_MODULES_PACING_PACED_SENDER_H_
 
 #include <list>
 #include <set>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -51,7 +51,11 @@ class PacedSender : public Module, public RtpPacketSender {
     virtual ~Callback() {}
   };
 
-  static const int64_t kDefaultMaxQueueLengthMs = 2000;
+  // Expected max pacer delay in ms. If ExpectedQueueTimeMs() is higher than
+  // this value, the packet producers should wait (eg drop frames rather than
+  // encoding them). Bitrate sent may temporarily exceed target set by
+  // UpdateBitrate() so that this limit will be upheld.
+  static const int64_t kMaxQueueLengthMs;
   // Pace in kbits/s until we receive first estimate.
   static const int kDefaultInitialPaceKbps = 2000;
   // Pacing-rate relative to our target send rate.
@@ -109,6 +113,10 @@ class PacedSender : public Module, public RtpPacketSender {
   // packets in the queue, given the current size and bitrate, ignoring prio.
   virtual int64_t ExpectedQueueTimeMs() const;
 
+  // Returns the average time since being enqueued, in milliseconds, for all
+  // packets currently in the pacer queue, or 0 if queue is empty.
+  virtual int64_t AverageQueueTimeMs();
+
   // Returns the number of milliseconds until the module want a worker thread
   // to call Process.
   int64_t TimeUntilNextProcess() override;
@@ -142,7 +150,10 @@ class PacedSender : public Module, public RtpPacketSender {
       GUARDED_BY(critsect_);
 
   rtc::scoped_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
+  // Actual configured bitrates (media_budget_ may temporarily be higher in
+  // order to meet pace time constraint).
   int bitrate_bps_ GUARDED_BY(critsect_);
+  int max_bitrate_kbps_ GUARDED_BY(critsect_);
 
   int64_t time_last_update_us_ GUARDED_BY(critsect_);
 
@@ -150,4 +161,4 @@ class PacedSender : public Module, public RtpPacketSender {
   uint64_t packet_counter_;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
+#endif  // WEBRTC_MODULES_PACING_PACED_SENDER_H_
diff --git a/webrtc/modules/pacing/paced_sender_unittest.cc b/webrtc/modules/pacing/paced_sender_unittest.cc
index c27444c5ac..588bf3b669 100644
--- a/webrtc/modules/pacing/paced_sender_unittest.cc
+++ b/webrtc/modules/pacing/paced_sender_unittest.cc
@@ -12,7 +12,7 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 using testing::_;
@@ -471,13 +471,15 @@ TEST_F(PacedSenderTest, Priority) {
                              sequence_number++, capture_time_ms, 250, false);
   send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                              sequence_number++, capture_time_ms, 250, false);
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
   send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
                              sequence_number++, capture_time_ms, 250, false);
 
   // Expect all high and normal priority to be sent out first.
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
   EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, capture_time_ms, false))
-      .Times(3)
+      .Times(4)
       .WillRepeatedly(Return(true));
 
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
@@ -497,6 +499,37 @@ TEST_F(PacedSenderTest, Priority) {
   EXPECT_EQ(0, send_bucket_->Process());
 }
 
+TEST_F(PacedSenderTest, HighPrioDoesntAffectBudget) {
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  int64_t capture_time_ms = 56789;
+
+  // As high prio packets doesn't affect the budget, we should be able to send
+  // a high number of them at once.
+  for (int i = 0; i < 25; ++i) {
+    SendAndExpectPacket(PacedSender::kHighPriority, ssrc, sequence_number++,
+                        capture_time_ms, 250, false);
+  }
+  send_bucket_->Process();
+  // Low prio packets does affect the budget, so we should only be able to send
+  // 3 at once, the 4th should be queued.
+  for (int i = 0; i < 3; ++i) {
+    SendAndExpectPacket(PacedSender::kLowPriority, ssrc, sequence_number++,
+                        capture_time_ms, 250, false);
+  }
+  send_bucket_->InsertPacket(PacedSender::kLowPriority, ssrc, sequence_number,
+                             capture_time_ms, 250, false);
+  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
+  clock_.AdvanceTimeMilliseconds(5);
+  send_bucket_->Process();
+  EXPECT_CALL(callback_,
+              TimeToSendPacket(ssrc, sequence_number++, capture_time_ms, false))
+      .Times(1);
+  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
+  clock_.AdvanceTimeMilliseconds(5);
+  send_bucket_->Process();
+}
+
 TEST_F(PacedSenderTest, Pause) {
   uint32_t ssrc_low_priority = 12345;
   uint32_t ssrc = 12346;
@@ -560,20 +593,16 @@ TEST_F(PacedSenderTest, Pause) {
   EXPECT_CALL(callback_, TimeToSendPacket(_, _, capture_time_ms, false))
       .Times(3)
       .WillRepeatedly(Return(true));
-  send_bucket_->Resume();
-
-  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
-  clock_.AdvanceTimeMilliseconds(5);
-  EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
-  EXPECT_EQ(0, send_bucket_->Process());
-
   EXPECT_CALL(callback_, TimeToSendPacket(_, _, second_capture_time_ms, false))
       .Times(1)
       .WillRepeatedly(Return(true));
+  send_bucket_->Resume();
+
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   clock_.AdvanceTimeMilliseconds(5);
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
   EXPECT_EQ(0, send_bucket_->Process());
+
   EXPECT_EQ(0, send_bucket_->QueueInMs());
 }
 
@@ -664,10 +693,9 @@ TEST_F(PacedSenderTest, ExpectedQueueTimeMs) {
 
   EXPECT_EQ(0, send_bucket_->ExpectedQueueTimeMs());
 
-  // Allow for aliasing, duration should be in [expected(n - 1), expected(n)].
-  EXPECT_LE(duration, queue_in_ms);
-  EXPECT_GE(duration,
-            queue_in_ms - static_cast<int64_t>(kPacketSize * 8 / kMaxBitrate));
+  // Allow for aliasing, duration should be within one pack of max time limit.
+  EXPECT_NEAR(duration, PacedSender::kMaxQueueLengthMs,
+              static_cast<int64_t>(kPacketSize * 8 / kMaxBitrate));
 }
 
 TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) {
@@ -830,5 +858,51 @@ TEST_F(PacedSenderTest, PaddingOveruse) {
   send_bucket_->Process();
 }
 
+TEST_F(PacedSenderTest, AverageQueueTime) {
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 1200;
+  const int kBitrateBps = 10 * kPacketSize * 8;  // 10 packets per second.
+  const int kBitrateKbps = (kBitrateBps + 500) / 1000;
+
+  send_bucket_->UpdateBitrate(kBitrateKbps, kBitrateKbps, kBitrateKbps);
+
+  EXPECT_EQ(0, send_bucket_->AverageQueueTimeMs());
+
+  int64_t first_capture_time = clock_.TimeInMilliseconds();
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number, first_capture_time, kPacketSize,
+                             false);
+  clock_.AdvanceTimeMilliseconds(10);
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number + 1, clock_.TimeInMilliseconds(),
+                             kPacketSize, false);
+  clock_.AdvanceTimeMilliseconds(10);
+
+  EXPECT_EQ((20 + 10) / 2, send_bucket_->AverageQueueTimeMs());
+
+  // Only first packet (queued for 20ms) should be removed, leave the second
+  // packet (queued for 10ms) alone in the queue.
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number,
+                                          first_capture_time, false))
+      .Times(1)
+      .WillRepeatedly(Return(true));
+  send_bucket_->Process();
+
+  EXPECT_EQ(10, send_bucket_->AverageQueueTimeMs());
+
+  clock_.AdvanceTimeMilliseconds(10);
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 1,
+                                          first_capture_time + 10, false))
+      .Times(1)
+      .WillRepeatedly(Return(true));
+  for (int i = 0; i < 3; ++i) {
+    clock_.AdvanceTimeMilliseconds(30);  // Max delta.
+    send_bucket_->Process();
+  }
+
+  EXPECT_EQ(0, send_bucket_->AverageQueueTimeMs());
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/webrtc/modules/pacing/pacing.gypi b/webrtc/modules/pacing/pacing.gypi
index faa97841c1..90f663c1b0 100644
--- a/webrtc/modules/pacing/pacing.gypi
+++ b/webrtc/modules/pacing/pacing.gypi
@@ -17,12 +17,12 @@
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       ],
       'sources': [
-        'include/paced_sender.h',
-        'include/packet_router.h',
         'bitrate_prober.cc',
         'bitrate_prober.h',
         'paced_sender.cc',
+        'paced_sender.h',
         'packet_router.cc',
+        'packet_router.h',
       ],
     },
   ], # targets
diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
index 563773b41f..5fd350834a 100644
--- a/webrtc/modules/pacing/packet_router.cc
+++ b/webrtc/modules/pacing/packet_router.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/pacing/include/packet_router.h"
+#include "webrtc/modules/pacing/packet_router.h"
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/pacing/include/packet_router.h b/webrtc/modules/pacing/packet_router.h
index 9d461d13a9..edef1aa9b3 100644
--- a/webrtc/modules/pacing/include/packet_router.h
+++ b/webrtc/modules/pacing/packet_router.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
-#define WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
+#ifndef WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
+#define WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
 
 #include <list>
 
@@ -18,8 +18,8 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
@@ -63,4 +63,4 @@ class PacketRouter : public PacedSender::Callback,
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
+#endif  // WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc
index eecb13757c..31acf44b9b 100644
--- a/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/webrtc/modules/pacing/packet_router_unittest.cc
@@ -13,8 +13,8 @@
 #include "webrtc/base/checks.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/base/scoped_ptr.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
index 2d5573228d..4820e6295f 100644
--- a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -88,8 +88,7 @@ uint32_t AimdRateControl::LatestEstimate() const {
 
 uint32_t AimdRateControl::UpdateBandwidthEstimate(int64_t now_ms) {
   current_bitrate_bps_ = ChangeBitrate(current_bitrate_bps_,
-                                       current_input_._incomingBitRate,
-                                       now_ms);
+                                       current_input_.incoming_bitrate, now_ms);
   if (now_ms - time_of_last_log_ > kLogIntervalMs) {
     time_of_last_log_ = now_ms;
   }
@@ -109,21 +108,21 @@ void AimdRateControl::Update(const RateControlInput* input, int64_t now_ms) {
     const int64_t kInitializationTimeMs = 5000;
     RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs);
     if (time_first_incoming_estimate_ < 0) {
-      if (input->_incomingBitRate > 0) {
+      if (input->incoming_bitrate > 0) {
         time_first_incoming_estimate_ = now_ms;
       }
     } else if (now_ms - time_first_incoming_estimate_ > kInitializationTimeMs &&
-               input->_incomingBitRate > 0) {
-      current_bitrate_bps_ = input->_incomingBitRate;
+               input->incoming_bitrate > 0) {
+      current_bitrate_bps_ = input->incoming_bitrate;
       bitrate_is_initialized_ = true;
     }
   }
 
-  if (updated_ && current_input_._bwState == kBwOverusing) {
+  if (updated_ && current_input_.bw_state == kBwOverusing) {
     // Only update delay factor and incoming bit rate. We always want to react
     // on an over-use.
-    current_input_._noiseVar = input->_noiseVar;
-    current_input_._incomingBitRate = input->_incomingBitRate;
+    current_input_.noise_var = input->noise_var;
+    current_input_.incoming_bitrate = input->incoming_bitrate;
   } else {
     updated_ = true;
     current_input_ = *input;
@@ -145,7 +144,7 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t current_bitrate_bps,
   // An over-use should always trigger us to reduce the bitrate, even though
   // we have not yet established our first estimate. By acting on the over-use,
   // we will end up with a valid estimate.
-  if (!bitrate_is_initialized_ && current_input_._bwState != kBwOverusing)
+  if (!bitrate_is_initialized_ && current_input_.bw_state != kBwOverusing)
     return current_bitrate_bps_;
   updated_ = false;
   ChangeState(current_input_, now_ms);
@@ -284,7 +283,7 @@ void AimdRateControl::UpdateMaxBitRateEstimate(float incoming_bitrate_kbps) {
 
 void AimdRateControl::ChangeState(const RateControlInput& input,
                                   int64_t now_ms) {
-  switch (current_input_._bwState) {
+  switch (current_input_.bw_state) {
     case kBwNormal:
       if (rate_control_state_ == kRcHold) {
         time_last_bitrate_change_ = now_ms;
diff --git a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
index bc5ca41dff..93ae2190d6 100644
--- a/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
+++ b/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
@@ -84,4 +84,4 @@ class AimdRateControl {
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
diff --git a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
index cb8d0db5c7..11fd64f84e 100644
--- a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
+++ b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
@@ -36,7 +36,7 @@ class BweSimulation : public BweTest,
     VerboseLogging(true);
   }
 
-  test::Random random_;
+  Random random_;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(BweSimulation);
@@ -74,7 +74,6 @@ TEST_P(BweSimulation, Verizon4gDownlinkTest) {
 }
 
 TEST_P(BweSimulation, Choke1000kbps500kbps1000kbpsBiDirectional) {
-
   const int kFlowIds[] = {0, 1};
   const size_t kNumFlows = sizeof(kFlowIds) / sizeof(kFlowIds[0]);
 
@@ -106,7 +105,6 @@ TEST_P(BweSimulation, Choke1000kbps500kbps1000kbpsBiDirectional) {
 }
 
 TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) {
-
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
   VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, 0);
@@ -243,7 +241,7 @@ TEST_P(BweSimulation, PacerGoogleWifiTrace3Mbps) {
 }
 
 TEST_P(BweSimulation, SelfFairnessTest) {
-  srand(Clock::GetRealTimeClock()->TimeInMicroseconds());
+  Random prng(Clock::GetRealTimeClock()->TimeInMicroseconds());
   const int kAllFlowIds[] = {0, 1, 2, 3};
   const size_t kNumFlows = sizeof(kAllFlowIds) / sizeof(kAllFlowIds[0]);
   rtc::scoped_ptr<VideoSource> sources[kNumFlows];
@@ -252,7 +250,7 @@ TEST_P(BweSimulation, SelfFairnessTest) {
     // Streams started 20 seconds apart to give them different advantage when
     // competing for the bandwidth.
     sources[i].reset(new AdaptiveVideoSource(kAllFlowIds[i], 30, 300, 0,
-                                             i * (rand() % 40000)));
+                                             i * prng.Rand(39999)));
     senders[i].reset(new VideoSender(&uplink_, sources[i].get(), GetParam()));
   }
 
@@ -283,9 +281,9 @@ TEST_P(BweSimulation, PacedSelfFairness50msTest) {
   const int64_t kAverageOffsetMs = 20 * 1000;
   const int kNumRmcatFlows = 4;
   int64_t offsets_ms[kNumRmcatFlows];
-  offsets_ms[0] = random_.Rand(0, 2 * kAverageOffsetMs);
+  offsets_ms[0] = random_.Rand(2 * kAverageOffsetMs);
   for (int i = 1; i < kNumRmcatFlows; ++i) {
-    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(0, 2 * kAverageOffsetMs);
+    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(2 * kAverageOffsetMs);
   }
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 1000, 3000, 50, 50, 0,
                   offsets_ms);
@@ -295,9 +293,9 @@ TEST_P(BweSimulation, PacedSelfFairness500msTest) {
   const int64_t kAverageOffsetMs = 20 * 1000;
   const int kNumRmcatFlows = 4;
   int64_t offsets_ms[kNumRmcatFlows];
-  offsets_ms[0] = random_.Rand(0, 2 * kAverageOffsetMs);
+  offsets_ms[0] = random_.Rand(2 * kAverageOffsetMs);
   for (int i = 1; i < kNumRmcatFlows; ++i) {
-    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(0, 2 * kAverageOffsetMs);
+    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(2 * kAverageOffsetMs);
   }
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 1000, 3000, 500, 50, 0,
                   offsets_ms);
@@ -307,28 +305,28 @@ TEST_P(BweSimulation, PacedSelfFairness1000msTest) {
   const int64_t kAverageOffsetMs = 20 * 1000;
   const int kNumRmcatFlows = 4;
   int64_t offsets_ms[kNumRmcatFlows];
-  offsets_ms[0] = random_.Rand(0, 2 * kAverageOffsetMs);
+  offsets_ms[0] = random_.Rand(2 * kAverageOffsetMs);
   for (int i = 1; i < kNumRmcatFlows; ++i) {
-    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(0, 2 * kAverageOffsetMs);
+    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(2 * kAverageOffsetMs);
   }
   RunFairnessTest(GetParam(), 4, 0, 1000, 3000, 1000, 50, 0, offsets_ms);
 }
 
 TEST_P(BweSimulation, TcpFairness50msTest) {
   const int64_t kAverageOffsetMs = 20 * 1000;
-  int64_t offset_ms[] = {random_.Rand(0, 2 * kAverageOffsetMs), 0};
+  int64_t offset_ms[] = {random_.Rand(2 * kAverageOffsetMs), 0};
   RunFairnessTest(GetParam(), 1, 1, 1000, 2000, 50, 50, 0, offset_ms);
 }
 
 TEST_P(BweSimulation, TcpFairness500msTest) {
   const int64_t kAverageOffsetMs = 20 * 1000;
-  int64_t offset_ms[] = {random_.Rand(0, 2 * kAverageOffsetMs), 0};
+  int64_t offset_ms[] = {random_.Rand(2 * kAverageOffsetMs), 0};
   RunFairnessTest(GetParam(), 1, 1, 1000, 2000, 500, 50, 0, offset_ms);
 }
 
 TEST_P(BweSimulation, TcpFairness1000msTest) {
   const int kAverageOffsetMs = 20 * 1000;
-  int64_t offset_ms[] = {random_.Rand(0, 2 * kAverageOffsetMs), 0};
+  int64_t offset_ms[] = {random_.Rand(2 * kAverageOffsetMs), 0};
   RunFairnessTest(GetParam(), 1, 1, 1000, 2000, 1000, 50, 0, offset_ms);
 }
 
diff --git a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
index 844fde5b71..3fb7e29e5b 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
@@ -8,53 +8,40 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
 
 #include "webrtc/typedefs.h"
 
-#define BWE_MAX(a,b) ((a)>(b)?(a):(b))
-#define BWE_MIN(a,b) ((a)<(b)?(a):(b))
+#define BWE_MAX(a, b) ((a) > (b) ? (a) : (b))
+#define BWE_MIN(a, b) ((a) < (b) ? (a) : (b))
 
 namespace webrtc {
 
 static const int64_t kBitrateWindowMs = 1000;
 
-enum BandwidthUsage
-{
-    kBwNormal = 0,
-    kBwUnderusing = 1,
-    kBwOverusing = 2,
+enum BandwidthUsage {
+  kBwNormal = 0,
+  kBwUnderusing = 1,
+  kBwOverusing = 2,
 };
 
-enum RateControlState
-{
-    kRcHold,
-    kRcIncrease,
-    kRcDecrease
-};
+enum RateControlState { kRcHold, kRcIncrease, kRcDecrease };
 
-enum RateControlRegion
-{
-    kRcNearMax,
-    kRcAboveMax,
-    kRcMaxUnknown
-};
+enum RateControlRegion { kRcNearMax, kRcAboveMax, kRcMaxUnknown };
+
+struct RateControlInput {
+  RateControlInput(BandwidthUsage bw_state,
+                   uint32_t incoming_bitrate,
+                   double noise_var)
+      : bw_state(bw_state),
+        incoming_bitrate(incoming_bitrate),
+        noise_var(noise_var) {}
 
-class RateControlInput
-{
-public:
-    RateControlInput(BandwidthUsage bwState,
-                     uint32_t incomingBitRate,
-                     double noiseVar)
-        : _bwState(bwState),
-          _incomingBitRate(incomingBitRate),
-          _noiseVar(noiseVar) {}
-
-    BandwidthUsage  _bwState;
-    uint32_t      _incomingBitRate;
-    double              _noiseVar;
+  BandwidthUsage bw_state;
+  uint32_t incoming_bitrate;
+  double noise_var;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
diff --git a/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h b/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h
index edfac977a2..ae05912b5f 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
-#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
 
 #include <vector>
 
@@ -26,4 +26,4 @@ class MockRemoteBitrateObserver : public RemoteBitrateObserver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
diff --git a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
index 4bd9d8c7bc..0734cbf255 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
@@ -17,9 +17,9 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h b/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
index 92d2e28132..a643c1f103 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
-#define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
 
 #include <map>
 
@@ -45,4 +45,4 @@ class SendTimeHistory {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
diff --git a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
index 3dee305bad..f75bc2b03e 100644
--- a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
+++ b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
@@ -14,7 +14,7 @@
 #include <cassert>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -71,8 +71,7 @@ bool InterArrival::ComputeDeltas(uint32_t timestamp,
     current_timestamp_group_.first_timestamp = timestamp;
     current_timestamp_group_.timestamp = timestamp;
     current_timestamp_group_.size = 0;
-  }
-  else {
+  } else {
     current_timestamp_group_.timestamp = LatestTimestamp(
         current_timestamp_group_.timestamp, timestamp);
   }
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc b/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
index c9340892f2..0acd7c29c5 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
@@ -10,11 +10,13 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
 
-#include <algorithm>
-#include <sstream>
 #include <math.h>
 #include <stdlib.h>
 
+#include <algorithm>
+#include <sstream>
+#include <string>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
index bb69a8a0a1..56e9c14206 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
@@ -13,7 +13,7 @@
 #include <list>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc b/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
index dcad04b5f6..50909ebd01 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
@@ -9,11 +9,14 @@
  */
 
 #include <math.h>
+
+#include <algorithm>
 #include <cmath>
 #include <cstdlib>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/random.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
@@ -21,8 +24,6 @@
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
 #include "webrtc/test/field_trial.h"
-#include "webrtc/test/random.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 namespace testing {
@@ -38,7 +39,7 @@ class OveruseDetectorTest : public ::testing::Test {
         overuse_detector_(),
         overuse_estimator_(new OveruseEstimator(options_)),
         inter_arrival_(new InterArrival(5 * 90, kRtpTimestampToMs, true)),
-        random_(1234) {}
+        random_(123456789) {}
 
  protected:
   void SetUp() override {
@@ -55,9 +56,10 @@ class OveruseDetectorTest : public ::testing::Test {
       }
       rtp_timestamp_ += mean_ms * 90;
       now_ms_ += mean_ms;
-      receive_time_ms_ =
-          std::max(receive_time_ms_,
-                   now_ms_ + random_.Gaussian(0, standard_deviation_ms));
+      receive_time_ms_ = std::max<int64_t>(
+          receive_time_ms_,
+          now_ms_ + static_cast<int64_t>(
+                        random_.Gaussian(0, standard_deviation_ms) + 0.5));
       if (kBwOverusing == overuse_detector_->State()) {
         if (last_overuse + 1 != i) {
           unique_overuse++;
@@ -77,9 +79,10 @@ class OveruseDetectorTest : public ::testing::Test {
       }
       rtp_timestamp_ += mean_ms * 90;
       now_ms_ += mean_ms + drift_per_frame_ms;
-      receive_time_ms_ =
-          std::max(receive_time_ms_,
-                   now_ms_ + random_.Gaussian(0, standard_deviation_ms));
+      receive_time_ms_ = std::max<int64_t>(
+          receive_time_ms_,
+          now_ms_ + static_cast<int64_t>(
+                        random_.Gaussian(0, standard_deviation_ms) + 0.5));
       if (kBwOverusing == overuse_detector_->State()) {
         return i + 1;
       }
@@ -114,7 +117,7 @@ class OveruseDetectorTest : public ::testing::Test {
   rtc::scoped_ptr<OveruseDetector> overuse_detector_;
   rtc::scoped_ptr<OveruseEstimator> overuse_estimator_;
   rtc::scoped_ptr<InterArrival> inter_arrival_;
-  test::Random random_;
+  Random random_;
 };
 
 TEST_F(OveruseDetectorTest, GaussianRandom) {
@@ -222,7 +225,7 @@ TEST_F(OveruseDetectorTest, DISABLED_OveruseWithHighVariance100Kbit10fps) {
     UpdateDetector(rtp_timestamp, now_ms_, packet_size);
     rtp_timestamp += frame_duration_ms * 90;
     if (i % 2) {
-      offset = rand() % 50;
+      offset = random_.Rand(0, 49);
       now_ms_ += frame_duration_ms - offset;
     } else {
       now_ms_ += frame_duration_ms + offset;
@@ -254,7 +257,7 @@ TEST_F(OveruseDetectorTest, DISABLED_OveruseWithLowVariance100Kbit10fps) {
     UpdateDetector(rtp_timestamp, now_ms_, packet_size);
     rtp_timestamp += frame_duration_ms * 90;
     if (i % 2) {
-      offset = rand() % 2;
+      offset = random_.Rand(0, 1);
       now_ms_ += frame_duration_ms - offset;
     } else {
       now_ms_ += frame_duration_ms + offset;
@@ -290,7 +293,7 @@ TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
     UpdateDetector(rtp_timestamp, now_ms_, packet_size);
     rtp_timestamp += frame_duration_ms * 90;
     if (i % 2) {
-      offset = rand() % 2;
+      offset = random_.Rand(0, 1);
       now_ms_ += frame_duration_ms - offset;
     } else {
       now_ms_ += frame_duration_ms + offset;
@@ -314,8 +317,13 @@ TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
   EXPECT_EQ(kBwOverusing, overuse_detector_->State());
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance30Kbit3fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance30Kbit3fps \
+  DISABLED_LowGaussianVariance30Kbit3fps
+#else
+#define MAYBE_LowGaussianVariance30Kbit3fps LowGaussianVariance30Kbit3fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance30Kbit3fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 333;
@@ -323,10 +331,10 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 3;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(13, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(14, frames_until_overuse);
+  EXPECT_EQ(13, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift30Kbit3fps) {
@@ -337,7 +345,7 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift30Kbit3fps) {
   int sigma_ms = 3;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(13, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
   EXPECT_EQ(4, frames_until_overuse);
@@ -351,10 +359,10 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance30Kbit3fps) {
   int sigma_ms = 10;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(46, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(42, frames_until_overuse);
+  EXPECT_EQ(32, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
@@ -365,14 +373,19 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
   int sigma_ms = 10;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(46, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
   EXPECT_EQ(4, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance100Kbit5fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance100Kbit5fps \
+  DISABLED_LowGaussianVariance100Kbit5fps
+#else
+#define MAYBE_LowGaussianVariance100Kbit5fps LowGaussianVariance100Kbit5fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit5fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 2;
   int frame_duration_ms = 200;
@@ -380,14 +393,19 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 3;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(12, unique_overuse);
+  EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(12, frames_until_overuse);
+  EXPECT_EQ(13, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(HighGaussianVariance100Kbit5fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_HighGaussianVariance100Kbit5fps \
+  DISABLED_HighGaussianVariance100Kbit5fps
+#else
+#define MAYBE_HighGaussianVariance100Kbit5fps HighGaussianVariance100Kbit5fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit5fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 2;
   int frame_duration_ms = 200;
@@ -395,14 +413,19 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 10;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(16, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(37, frames_until_overuse);
+  EXPECT_EQ(32, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance100Kbit10fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance100Kbit10fps \
+  DISABLED_LowGaussianVariance100Kbit10fps
+#else
+#define MAYBE_LowGaussianVariance100Kbit10fps LowGaussianVariance100Kbit10fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit10fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 100;
@@ -410,14 +433,19 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 3;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(12, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(12, frames_until_overuse);
+  EXPECT_EQ(13, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(HighGaussianVariance100Kbit10fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_HighGaussianVariance100Kbit10fps \
+  DISABLED_HighGaussianVariance100Kbit10fps
+#else
+#define MAYBE_HighGaussianVariance100Kbit10fps HighGaussianVariance100Kbit10fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit10fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 100;
@@ -425,14 +453,19 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 10;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(12, unique_overuse);
+  EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(37, frames_until_overuse);
+  EXPECT_EQ(32, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance300Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance300Kbit30fps \
+  DISABLED_LowGaussianVariance300Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance300Kbit30fps LowGaussianVariance300Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance300Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 33;
@@ -443,7 +476,7 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(14, frames_until_overuse);
+  EXPECT_EQ(15, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift300Kbit30fps) {
@@ -471,7 +504,7 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance300Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(49, frames_until_overuse);
+  EXPECT_EQ(41, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
@@ -485,11 +518,16 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(8, frames_until_overuse);
+  EXPECT_EQ(10, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance1000Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance1000Kbit30fps \
+  DISABLED_LowGaussianVariance1000Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance1000Kbit30fps LowGaussianVariance1000Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance1000Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 3;
   int frame_duration_ms = 33;
@@ -500,7 +538,7 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(14, frames_until_overuse);
+  EXPECT_EQ(15, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift1000Kbit30fps) {
@@ -528,7 +566,7 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance1000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(49, frames_until_overuse);
+  EXPECT_EQ(41, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
@@ -542,11 +580,16 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(8, frames_until_overuse);
+  EXPECT_EQ(10, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance2000Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance2000Kbit30fps \
+  DISABLED_LowGaussianVariance2000Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance2000Kbit30fps LowGaussianVariance2000Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance2000Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 6;
   int frame_duration_ms = 33;
@@ -557,7 +600,7 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(14, frames_until_overuse);
+  EXPECT_EQ(15, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift2000Kbit30fps) {
@@ -585,7 +628,7 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance2000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(49, frames_until_overuse);
+  EXPECT_EQ(41, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift2000Kbit30fps) {
@@ -599,7 +642,7 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift2000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(8, frames_until_overuse);
+  EXPECT_EQ(10, frames_until_overuse);
 }
 
 class OveruseDetectorExperimentTest : public OveruseDetectorTest {
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc b/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
index 4be7b7493b..83917912e8 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
@@ -10,14 +10,15 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
 
-#include <algorithm>
 #include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include <algorithm>
+
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 2dc32a7ee7..97e5cd32e5 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -12,14 +12,16 @@
 
 #include <math.h>
 
+#include <algorithm>
+
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -241,6 +243,7 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(int64_t arrival_time_ms,
   if (!header.extension.hasAbsoluteSendTime) {
     LOG(LS_WARNING) << "RemoteBitrateEstimatorAbsSendTimeImpl: Incoming packet "
                        "is missing absolute send time extension!";
+    return;
   }
   IncomingPacketInfo(arrival_time_ms, header.extension.absoluteSendTime,
                      payload_size, header.ssrc, was_paced);
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
index 195c95aacb..908daf6c31 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
@@ -17,7 +17,6 @@ namespace webrtc {
 class RemoteBitrateEstimatorAbsSendTimeTest :
     public RemoteBitrateEstimatorTest {
  public:
-
   RemoteBitrateEstimatorAbsSendTimeTest() {}
   virtual void SetUp() {
     bitrate_estimator_.reset(new RemoteBitrateEstimatorAbsSendTime(
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index f1a1cb6602..4b7732c80f 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -10,16 +10,18 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 
+#include <utility>
+
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
 #include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
 #include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
-#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -28,19 +30,20 @@ enum { kTimestampGroupLengthMs = 5 };
 static const double kTimestampToMs = 1.0 / 90.0;
 
 struct RemoteBitrateEstimatorSingleStream::Detector {
-    explicit Detector(int64_t last_packet_time_ms,
-                      const OverUseDetectorOptions& options,
-                      bool enable_burst_grouping)
-        : last_packet_time_ms(last_packet_time_ms),
-          inter_arrival(90 * kTimestampGroupLengthMs, kTimestampToMs,
-                        enable_burst_grouping),
-          estimator(options),
-          detector(options) {}
-    int64_t last_packet_time_ms;
-    InterArrival inter_arrival;
-    OveruseEstimator estimator;
-    OveruseDetector detector;
-  };
+  explicit Detector(int64_t last_packet_time_ms,
+                    const OverUseDetectorOptions& options,
+                    bool enable_burst_grouping)
+      : last_packet_time_ms(last_packet_time_ms),
+        inter_arrival(90 * kTimestampGroupLengthMs,
+                      kTimestampToMs,
+                      enable_burst_grouping),
+        estimator(options),
+        detector(options) {}
+  int64_t last_packet_time_ms;
+  InterArrival inter_arrival;
+  OveruseEstimator estimator;
+  OveruseDetector detector;
+};
 
   RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream(
       RemoteBitrateObserver* observer,
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
index a6c182a7bc..7a26a7e63b 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
@@ -17,7 +17,6 @@ namespace webrtc {
 class RemoteBitrateEstimatorSingleTest :
     public RemoteBitrateEstimatorTest {
  public:
-
   RemoteBitrateEstimatorSingleTest() {}
   virtual void SetUp() {
     bitrate_estimator_.reset(new RemoteBitrateEstimatorSingleStream(
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
index 315f5422d9..8b9c0b9a1d 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
@@ -10,6 +10,7 @@
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
 
 #include <algorithm>
+#include <limits>
 #include <utility>
 
 namespace webrtc {
@@ -383,11 +384,11 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
                                         2 * kFrameIntervalAbsSendTime);
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
                    absolute_send_time, true);
-    IncomingPacket(
-        kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
-        timestamp - 90 * kFrameIntervalMs,
-        AddAbsSendTime(absolute_send_time, -int(kFrameIntervalAbsSendTime)),
-        true);
+    IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
+                   timestamp - 90 * kFrameIntervalMs,
+                   AddAbsSendTime(absolute_send_time,
+                                  -static_cast<int>(kFrameIntervalAbsSendTime)),
+                   true);
   }
   bitrate_estimator_->Process();
   EXPECT_TRUE(bitrate_observer_->updated());
@@ -520,8 +521,8 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
   uint32_t timestamp = 0;
   // Initialize absolute_send_time (24 bits) so that it will definitely wrap
   // during the test.
-  uint32_t absolute_send_time =
-      AddAbsSendTime((1 << 24), -int(50 * kFrameIntervalAbsSendTime));
+  uint32_t absolute_send_time = AddAbsSendTime(
+      (1 << 24), -static_cast<int>(50 * kFrameIntervalAbsSendTime));
   // Initial set of frames to increase the bitrate. 6 seconds to have enough
   // time for the first estimate to be generated and for Process() to be called.
   for (int i = 0; i <= 6 * kFramerate; ++i) {
@@ -556,8 +557,10 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
     // Increase time until next batch to simulate over-use.
     clock_.AdvanceTimeMilliseconds(10);
     timestamp += 90 * kFrameIntervalMs - kTimestampGroupLength;
-    absolute_send_time = AddAbsSendTime(absolute_send_time, AddAbsSendTime(
-        kFrameIntervalAbsSendTime, -int(kTimestampGroupLengthAbsSendTime)));
+    absolute_send_time = AddAbsSendTime(
+        absolute_send_time,
+        AddAbsSendTime(kFrameIntervalAbsSendTime,
+                       -static_cast<int>(kTimestampGroupLengthAbsSendTime)));
     bitrate_estimator_->Process();
   }
   EXPECT_TRUE(bitrate_observer_->updated());
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
index 606bb6c4e6..8343d7d57b 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
@@ -14,6 +14,7 @@
 #include <list>
 #include <map>
 #include <utility>
+#include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/constructormagic.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
index d6f049f6ac..2ce144129b 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
@@ -13,8 +13,10 @@
 #include <unistd.h>
 #endif
 
+#include <algorithm>
 #include <sstream>
 
+#include "webrtc/base/random.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
@@ -242,18 +244,20 @@ class BweFeedbackTest
     : public BweTest,
       public ::testing::TestWithParam<BandwidthEstimatorType> {
  public:
-  BweFeedbackTest() : BweTest() {}
+#ifdef WEBRTC_WIN
+  BweFeedbackTest()
+      : BweTest(), random_(Clock::GetRealTimeClock()->TimeInMicroseconds()) {}
+#else
+  BweFeedbackTest()
+      : BweTest(),
+        // Multiply the time by a random-ish odd number derived from the PID.
+        random_((getpid() | 1) *
+                Clock::GetRealTimeClock()->TimeInMicroseconds()) {}
+#endif
   virtual ~BweFeedbackTest() {}
 
  protected:
-  void SetUp() override {
-    unsigned int seed = Clock::GetRealTimeClock()->TimeInMicroseconds();
-#ifndef WEBRTC_WIN
-    seed *= getpid();
-#endif
-    srand(seed);
-    BweTest::SetUp();
-  }
+  Random random_;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(BweFeedbackTest);
@@ -356,7 +360,7 @@ TEST_P(BweFeedbackTest, PacedSelfFairness50msTest) {
   const int kNumRmcatFlows = 4;
   int64_t offset_ms[kNumRmcatFlows];
   for (int i = 0; i < kNumRmcatFlows; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 50, kRttMs,
@@ -370,7 +374,7 @@ TEST_P(BweFeedbackTest, PacedSelfFairness500msTest) {
   const int kNumRmcatFlows = 4;
   int64_t offset_ms[kNumRmcatFlows];
   for (int i = 0; i < kNumRmcatFlows; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 500, kRttMs,
@@ -384,7 +388,7 @@ TEST_P(BweFeedbackTest, PacedSelfFairness1000msTest) {
   const int kNumRmcatFlows = 4;
   int64_t offset_ms[kNumRmcatFlows];
   for (int i = 0; i < kNumRmcatFlows; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 1000, kRttMs,
@@ -397,7 +401,7 @@ TEST_P(BweFeedbackTest, TcpFairness50msTest) {
 
   int64_t offset_ms[2];  // One TCP, one RMCAT flow.
   for (int i = 0; i < 2; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), 1, 1, 300, 2000, 50, kRttMs, kMaxJitterMs,
@@ -410,7 +414,7 @@ TEST_P(BweFeedbackTest, TcpFairness500msTest) {
 
   int64_t offset_ms[2];  // One TCP, one RMCAT flow.
   for (int i = 0; i < 2; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), 1, 1, 300, 2000, 500, kRttMs, kMaxJitterMs,
@@ -423,7 +427,7 @@ TEST_P(BweFeedbackTest, TcpFairness1000msTest) {
 
   int64_t offset_ms[2];  // One TCP, one RMCAT flow.
   for (int i = 0; i < 2; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), 1, 1, 300, 2000, 1000, kRttMs, kMaxJitterMs,
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
index b7f9f65dbc..15ca42dda9 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
@@ -13,14 +13,14 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
+#include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 
 namespace webrtc {
 
 // TODO(sprang): Tune these!
-const int RemoteEstimatorProxy::kDefaultProcessIntervalMs = 200;
+const int RemoteEstimatorProxy::kDefaultProcessIntervalMs = 50;
 const int RemoteEstimatorProxy::kBackWindowMs = 500;
 
 RemoteEstimatorProxy::RemoteEstimatorProxy(Clock* clock,
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
index e867ff77a4..98a68b3dcf 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
+++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
index 826a724e33..7ddd31467b 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
@@ -11,7 +11,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/pacing/include/packet_router.h"
+#include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/system_wrappers/include/clock.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe.h b/webrtc/modules/remote_bitrate_estimator/test/bwe.h
index ef9b3149d7..8d29de2619 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe.h
@@ -11,7 +11,10 @@
 #ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_H_
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_H_
 
+#include <list>
+#include <map>
 #include <sstream>
+#include <string>
 
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
index f837638474..9da21c1aaa 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
@@ -12,9 +12,10 @@
 
 #include <sstream>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
@@ -662,7 +663,7 @@ void BweTest::RunSelfFairness(BandwidthEstimatorType bwe_type) {
 void BweTest::RunRoundTripTimeFairness(BandwidthEstimatorType bwe_type) {
   const int kAllFlowIds[] = {0, 1, 2, 3, 4};  // Five RMCAT flows.
   const int64_t kAllOneWayDelayMs[] = {10, 25, 50, 100, 150};
-  const size_t kNumFlows = ARRAY_SIZE(kAllFlowIds);
+  const size_t kNumFlows = arraysize(kAllFlowIds);
   rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumFlows];
   rtc::scoped_ptr<VideoSender> senders[kNumFlows];
   rtc::scoped_ptr<MetricRecorder> metric_recorders[kNumFlows];
@@ -774,10 +775,10 @@ void BweTest::RunMultipleShortTcpFairness(
   const int kAllTcpFlowIds[] = {2, 3, 4, 5, 6, 7, 8, 9, 10, 11};
 
   assert(tcp_starting_times_ms.size() == tcp_file_sizes_bytes.size() &&
-         tcp_starting_times_ms.size() == ARRAY_SIZE(kAllTcpFlowIds));
+         tcp_starting_times_ms.size() == arraysize(kAllTcpFlowIds));
 
-  const size_t kNumRmcatFlows = ARRAY_SIZE(kAllRmcatFlowIds);
-  const size_t kNumTotalFlows = kNumRmcatFlows + ARRAY_SIZE(kAllTcpFlowIds);
+  const size_t kNumRmcatFlows = arraysize(kAllRmcatFlowIds);
+  const size_t kNumTotalFlows = kNumRmcatFlows + arraysize(kAllTcpFlowIds);
 
   rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumRmcatFlows];
   rtc::scoped_ptr<PacketSender> senders[kNumTotalFlows];
@@ -869,7 +870,7 @@ void BweTest::RunMultipleShortTcpFairness(
 // During the test, one of the flows is paused and later resumed.
 void BweTest::RunPauseResumeFlows(BandwidthEstimatorType bwe_type) {
   const int kAllFlowIds[] = {0, 1, 2};  // Three RMCAT flows.
-  const size_t kNumFlows = ARRAY_SIZE(kAllFlowIds);
+  const size_t kNumFlows = arraysize(kAllFlowIds);
 
   rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumFlows];
   rtc::scoped_ptr<VideoSender> senders[kNumFlows];
@@ -947,7 +948,7 @@ std::vector<int> BweTest::GetFileSizesBytes(int num_files) {
   const int kMinKbytes = 100;
   const int kMaxKbytes = 1000;
 
-  test::Random random(0x12345678);
+  Random random(0x12345678);
   std::vector<int> tcp_file_sizes_bytes;
 
   while (num_files-- > 0) {
@@ -960,7 +961,7 @@ std::vector<int> BweTest::GetFileSizesBytes(int num_files) {
 std::vector<int64_t> BweTest::GetStartingTimesMs(int num_files) {
   // OFF state behaves as an exp. distribution with mean = 10 seconds.
   const float kMeanMs = 10000.0f;
-  test::Random random(0x12345678);
+  Random random(0x12345678);
 
   std::vector<int64_t> tcp_starting_times_ms;
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
index 64dfa85535..b3df7124e3 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_BASELINEFILE_H_
 
 #include <string>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
index 2881eba424..d470324ac3 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
@@ -16,7 +16,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
index 4574d3d8a1..41bf836c9e 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
@@ -323,7 +323,7 @@ void LossFilter::SetLoss(float loss_percent) {
 void LossFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
-    if (random_.Rand() < loss_fraction_) {
+    if (random_.Rand<float>() < loss_fraction_) {
       delete *it;
       it = in_out->erase(it);
     } else {
@@ -391,7 +391,7 @@ void JitterFilter::SetMaxJitter(int64_t max_jitter_ms) {
 }
 
 namespace {
-inline int64_t TruncatedNSigmaGaussian(test::Random* const random,
+inline int64_t TruncatedNSigmaGaussian(Random* const random,
                                        int64_t mean,
                                        int64_t std_dev) {
   int64_t gaussian_random = random->Gaussian(mean, std_dev);
@@ -459,7 +459,7 @@ void ReorderFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
     PacketsIt last_it = in_out->begin();
     PacketsIt it = last_it;
     while (++it != in_out->end()) {
-      if (random_.Rand() < reorder_fraction_) {
+      if (random_.Rand<float>() < reorder_fraction_) {
         int64_t t1 = (*last_it)->send_time_us();
         int64_t t2 = (*it)->send_time_us();
         std::swap(*last_it, *it);
@@ -586,7 +586,7 @@ bool TraceBasedDeliveryFilter::Init(const std::string& filename) {
     return false;
   }
   int64_t first_timestamp = -1;
-  while(!feof(trace_file)) {
+  while (!feof(trace_file)) {
     const size_t kMaxLineLength = 100;
     char line[kMaxLineLength];
     if (fgets(line, kMaxLineLength, trace_file)) {
@@ -680,6 +680,7 @@ VideoSource::VideoSource(int flow_id,
       frame_period_ms_(1000.0 / fps),
       bits_per_second_(1000 * kbps),
       frame_size_bytes_(bits_per_second_ / 8 / fps),
+      random_(0x12345678),
       flow_id_(flow_id),
       next_frame_ms_(first_frame_offset_ms),
       next_frame_rand_ms_(0),
@@ -713,9 +714,7 @@ void VideoSource::RunFor(int64_t time_ms, Packets* in_out) {
     const int64_t kRandAmplitude = 2;
     // A variance picked uniformly from {-1, 0, 1} ms is added to the frame
     // timestamp.
-    next_frame_rand_ms_ =
-        kRandAmplitude * static_cast<float>(rand()) / RAND_MAX -
-        kRandAmplitude / 2;
+    next_frame_rand_ms_ = kRandAmplitude * (random_.Rand<float>() - 0.5);
 
     // Ensure frame will not have a negative timestamp.
     int64_t next_frame_ms =
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
index 6b24cf30a6..3bb9b95f4b 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
@@ -17,21 +17,23 @@
 #include <algorithm>
 #include <list>
 #include <numeric>
+#include <set>
 #include <sstream>
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "webrtc/base/common.h"
+#include "webrtc/base/random.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/test/random.h"
 
 namespace webrtc {
 
@@ -44,7 +46,7 @@ class DelayCapHelper;
 
 class RateCounter {
  public:
-  RateCounter(int64_t window_size_ms)
+  explicit RateCounter(int64_t window_size_ms)
       : window_size_us_(1000 * window_size_ms),
         recently_received_packets_(0),
         recently_received_bytes_(0),
@@ -265,7 +267,7 @@ class LossFilter : public PacketProcessor {
   virtual void RunFor(int64_t time_ms, Packets* in_out);
 
  private:
-  test::Random random_;
+  Random random_;
   float loss_fraction_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LossFilter);
@@ -299,7 +301,7 @@ class JitterFilter : public PacketProcessor {
   int64_t MeanUs();
 
  private:
-  test::Random random_;
+  Random random_;
   int64_t stddev_jitter_us_;
   int64_t last_send_time_us_;
   bool reordering_;  // False by default.
@@ -318,7 +320,7 @@ class ReorderFilter : public PacketProcessor {
   virtual void RunFor(int64_t time_ms, Packets* in_out);
 
  private:
-  test::Random random_;
+  Random random_;
   float reorder_fraction_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ReorderFilter);
@@ -415,6 +417,7 @@ class VideoSource {
   uint32_t frame_size_bytes_;
 
  private:
+  Random random_;
   const int flow_id_;
   int64_t next_frame_ms_;
   int64_t next_frame_rand_ms_;
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
index 627260678b..6bdfa847df 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
@@ -22,39 +22,6 @@ namespace webrtc {
 namespace testing {
 namespace bwe {
 
-TEST(BweTestFramework_RandomTest, Gaussian) {
-  enum {
-    kN = 100000,
-    kBuckets = 100,
-    kMean = 49,
-    kStddev = 10
-  };
-
-  test::Random random(0x12345678);
-
-  int buckets[kBuckets] = {0};
-  for (int i = 0; i < kN; ++i) {
-    int index = random.Gaussian(kMean, kStddev);
-    if (index >= 0 && index < kBuckets) {
-      buckets[index]++;
-    }
-  }
-
-  const double kPi = 3.14159265358979323846;
-  const double kScale = kN / (kStddev * sqrt(2.0 * kPi));
-  const double kDiv = -2.0 * kStddev * kStddev;
-  double self_corr = 0.0;
-  double bucket_corr = 0.0;
-  for (int n = 0; n < kBuckets; ++n) {
-    double normal_dist = kScale * exp((n - kMean) * (n - kMean) / kDiv);
-    self_corr += normal_dist * normal_dist;
-    bucket_corr += normal_dist * buckets[n];
-  }
-  printf("Correlation: %f (random sample), %f (self), %f (quotient)\n",
-         bucket_corr, self_corr, bucket_corr / self_corr);
-  EXPECT_NEAR(1.0, bucket_corr / self_corr, 0.0004);
-}
-
 static bool IsSequenceNumberSorted(const Packets& packets) {
   PacketsConstIt last_it = packets.begin();
   for (PacketsConstIt it = last_it; it != packets.end(); ++it) {
@@ -533,7 +500,7 @@ TEST(BweTestFramework_JitterFilterTest, Jitter1031) {
   TestJitterFilter(1031);
 }
 
-static void TestReorderFilter(uint16_t reorder_percent, uint16_t near_value) {
+static void TestReorderFilter(uint16_t reorder_percent) {
   const uint16_t kPacketCount = 10000;
 
   // Generate packets with 10 ms interval.
@@ -559,16 +526,23 @@ static void TestReorderFilter(uint16_t reorder_percent, uint16_t near_value) {
   for (auto* packet : packets) {
     const MediaPacket* media_packet = static_cast<const MediaPacket*>(packet);
     uint16_t sequence_number = media_packet->header().sequenceNumber;
+    // The expected position for sequence number s is in position s-1.
     if (sequence_number < last_sequence_number) {
       distance += last_sequence_number - sequence_number;
     }
     last_sequence_number = sequence_number;
   }
 
-  // Because reordering is random, we allow a threshold when comparing. The
-  // maximum distance a packet can be moved is PacketCount - 1.
-  EXPECT_NEAR(
-    ((kPacketCount - 1) * reorder_percent) / 100, distance, near_value);
+  // The probability that two elements are swapped is p = reorder_percent / 100.
+  double p = static_cast<double>(reorder_percent) / 100;
+  // The expected number of swaps we perform is p * (PacketCount - 1),
+  // and each swap increases the distance by one.
+  double mean = p * (kPacketCount - 1);
+  // If pair i is chosen to be swapped with probability p, the variance for that
+  // pair is p * (1 - p). Since there are (kPacketCount - 1) independent pairs,
+  // the variance for the number of swaps is (kPacketCount - 1) * p * (1 - p).
+  double std_deviation = sqrt((kPacketCount - 1) * p * (1 - p));
+  EXPECT_NEAR(mean, distance, 3 * std_deviation);
 
   for (auto* packet : packets)
     delete packet;
@@ -576,23 +550,23 @@ static void TestReorderFilter(uint16_t reorder_percent, uint16_t near_value) {
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder0) {
   // For 0% reordering, no packets should have been moved, so result is exact.
-  TestReorderFilter(0, 0);
+  TestReorderFilter(0);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder10) {
-  TestReorderFilter(10, 30);
+  TestReorderFilter(10);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder20) {
-  TestReorderFilter(20, 20);
+  TestReorderFilter(20);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder50) {
-  TestReorderFilter(50, 20);
+  TestReorderFilter(50);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder70) {
-  TestReorderFilter(70, 20);
+  TestReorderFilter(70);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder100) {
@@ -600,7 +574,7 @@ TEST(BweTestFramework_ReorderFilterTest, Reorder100) {
   // adjacent packets, when the likelihood of a swap is 1.0, a swap will always
   // occur, so the stream will be in order except for the first packet, which
   // has been moved to the end. Therefore we expect the result to be exact here.
-  TestReorderFilter(100.0, 0);
+  TestReorderFilter(100.0);
 }
 
 class BweTestFramework_ChokeFilterTest : public ::testing::Test {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
index dcc08d8dde..3a84e81a0b 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
@@ -18,8 +18,8 @@
 #include <algorithm>
 #include <sstream>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 namespace webrtc {
 namespace testing {
@@ -57,27 +57,27 @@ Logging* Logging::GetInstance() {
 
 void Logging::SetGlobalContext(uint32_t name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.tag = ToString(name);
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = ToString(name);
 }
 
 void Logging::SetGlobalContext(const std::string& name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.tag = name;
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = name;
 }
 
 void Logging::SetGlobalContext(const char* name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.tag = name;
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = name;
 }
 
 void Logging::SetGlobalEnable(bool enabled) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.enabled = enabled;
+  thread_map_[rtc::CurrentThreadId()].global_state.enabled = enabled;
 }
 
 void Logging::Log(const char format[], ...) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
@@ -96,7 +96,7 @@ void Logging::Plot(int figure, double value) {
 
 void Logging::Plot(int figure, double value, const std::string& alg_name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   std::string label = state.tag + '@' + alg_name;
@@ -119,7 +119,7 @@ void Logging::PlotBar(int figure,
                       double value,
                       int flow_id) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
@@ -132,7 +132,7 @@ void Logging::PlotBaselineBar(int figure,
                               double value,
                               int flow_id) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
@@ -148,7 +148,7 @@ void Logging::PlotErrorBar(int figure,
                            const std::string& error_title,
                            int flow_id) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
@@ -167,7 +167,7 @@ void Logging::PlotLimitErrorBar(int figure,
                                 const std::string& limit_title,
                                 int flow_id) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
@@ -182,7 +182,7 @@ void Logging::PlotLabel(int figure,
                         const std::string& y_label,
                         int num_flows) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
@@ -219,7 +219,7 @@ void Logging::PushState(const std::string& append_to_tag, int64_t timestamp_ms,
                         bool enabled) {
   CriticalSectionScoped cs(crit_sect_.get());
   State new_state(append_to_tag, timestamp_ms, enabled);
-  ThreadState* thread_state = &thread_map_[ThreadWrapper::GetThreadId()];
+  ThreadState* thread_state = &thread_map_[rtc::CurrentThreadId()];
   std::stack<State>* stack = &thread_state->stack;
   if (stack->empty()) {
     new_state.MergePrevious(thread_state->global_state);
@@ -231,7 +231,7 @@ void Logging::PushState(const std::string& append_to_tag, int64_t timestamp_ms,
 
 void Logging::PopState() {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   std::stack<State>* stack = &it->second.stack;
   int64_t newest_timestamp_ms = stack->top().timestamp_ms;
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
index 4115d30c2a..cc7807ba8a 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
@@ -130,12 +130,12 @@
 #define BWE_TEST_LOGGING_GLOBAL_CONTEXT(name) \
     do { \
       webrtc::testing::bwe::Logging::GetInstance()->SetGlobalContext(name); \
-    } while (0);
+    } while (0)
 
 #define BWE_TEST_LOGGING_GLOBAL_ENABLE(enabled) \
     do { \
       webrtc::testing::bwe::Logging::GetInstance()->SetGlobalEnable(enabled); \
-    } while (0);
+    } while (0)
 
 #define __BWE_TEST_LOGGING_CONTEXT_NAME(ctx, line) ctx ## line
 #define __BWE_TEST_LOGGING_CONTEXT_DECLARE(ctx, line, name, time, enabled) \
@@ -155,36 +155,36 @@
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG2(name, format, _1, _2) \
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG3(name, format, _1, _2, _3) \
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG4(name, format, _1, _2, _3, _4) \
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, \
                                                         _4); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG5(name, format, _1, _2, _3, _4, _5) \
     do {\
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, \
                                                         _4, _5); \
-    } while (0);
+    } while (0)
 
 #define BWE_TEST_LOGGING_PLOT(figure, name, time, value)                  \
   do {                                                                    \
     __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __PLOT__, name,        \
                                        static_cast<int64_t>(time), true); \
     webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, value);    \
-  } while (0);
+  } while (0)
 
 #define BWE_TEST_LOGGING_PLOT_WITH_NAME(figure, name, time, value, alg_name) \
   do {                                                                       \
@@ -192,21 +192,21 @@
                                        static_cast<int64_t>(time), true);    \
     webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, value,        \
                                                        alg_name);            \
-  } while (0);
+  } while (0)
 
 #define BWE_TEST_LOGGING_BAR(figure, name, value, flow_id)                     \
   do {                                                                         \
     BWE_TEST_LOGGING_CONTEXT(name);                                            \
     webrtc::testing::bwe::Logging::GetInstance()->PlotBar(figure, name, value, \
                                                           flow_id);            \
-  } while (0);
+  } while (0)
 
 #define BWE_TEST_LOGGING_BASELINEBAR(figure, name, value, flow_id) \
   do {                                                             \
     BWE_TEST_LOGGING_CONTEXT(name);                                \
     webrtc::testing::bwe::Logging::GetInstance()->PlotBaselineBar( \
         figure, name, value, flow_id);                             \
-  } while (0);
+  } while (0)
 
 #define BWE_TEST_LOGGING_ERRORBAR(figure, name, value, ylow, yhigh, title, \
                                   flow_id)                                 \
@@ -214,7 +214,7 @@
     BWE_TEST_LOGGING_CONTEXT(name);                                        \
     webrtc::testing::bwe::Logging::GetInstance()->PlotErrorBar(            \
         figure, name, value, ylow, yhigh, title, flow_id);                 \
-  } while (0);
+  } while (0)
 
 #define BWE_TEST_LOGGING_LIMITERRORBAR(                                        \
     figure, name, value, ylow, yhigh, error_title, ymax, limit_title, flow_id) \
@@ -223,14 +223,14 @@
     webrtc::testing::bwe::Logging::GetInstance()->PlotLimitErrorBar(           \
         figure, name, value, ylow, yhigh, error_title, ymax, limit_title,      \
         flow_id);                                                              \
-  } while (0);
+  } while (0)
 
 #define BWE_TEST_LOGGING_LABEL(figure, title, y_label, num_flows) \
   do {                                                            \
     BWE_TEST_LOGGING_CONTEXT(title);                              \
     webrtc::testing::bwe::Logging::GetInstance()->PlotLabel(      \
         figure, title, y_label, num_flows);                       \
-  } while (0);
+  } while (0)
 
 namespace webrtc {
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
index 6b3ce4847c..6245ccd25d 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
@@ -13,6 +13,7 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
 
 namespace webrtc {
 namespace testing {
@@ -241,7 +242,7 @@ TEST_F(BweReceiverTest, PacketLossDuplicatedPackets) {
 
   // Missing the element 5.
   const uint16_t kSequenceNumbers[] = {1, 2, 3, 4, 6, 7, 8};
-  const int kNumPackets = ARRAY_SIZE(kSequenceNumbers);
+  const int kNumPackets = arraysize(kSequenceNumbers);
 
   // Insert each sequence number twice.
   for (int i = 0; i < 2; ++i) {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
index d77447f1ea..6166ff8c2d 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
@@ -18,10 +18,11 @@
 #include <algorithm>
 #include <vector>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
@@ -63,7 +64,7 @@ void NadaBweReceiver::ReceivePacket(int64_t arrival_time_ms,
   }
 
   delay_signal_ms_ = delay_ms - baseline_delay_ms_;  // Refered as d_n.
-  const int kMedian = ARRAY_SIZE(last_delays_ms_);
+  const int kMedian = arraysize(last_delays_ms_);
   last_delays_ms_[(last_delays_index_++) % kMedian] = delay_signal_ms_;
   int size = std::min(last_delays_index_, kMedian);
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
index eee90cf463..bf23d09884 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
@@ -20,7 +20,7 @@
 #include <list>
 #include <map>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/voice_engine/channel.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
index a0f56b73b7..51afae1df4 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
@@ -13,6 +13,7 @@
 #include <algorithm>
 #include <numeric>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
@@ -357,7 +358,7 @@ TEST_F(NadaReceiverSideTest, FeedbackIncreasingDelay) {
   // Baseline delay will be 50 ms.
   // Delay signals should be: [0 10 20 30 40 50 60 70] ms.
   const int64_t kMedianFilteredDelaysMs[] = {0, 5, 10, 15, 20, 30, 40, 50};
-  const int kNumPackets = ARRAY_SIZE(kMedianFilteredDelaysMs);
+  const int kNumPackets = arraysize(kMedianFilteredDelaysMs);
   const float kAlpha = 0.1f;  // Used for exponential smoothing.
 
   int64_t exp_smoothed_delays_ms[kNumPackets];
@@ -426,7 +427,7 @@ TEST_F(NadaReceiverSideTest, FeedbackWarpedDelay) {
   // Delay signals should be: [0 200 400 600 800 1000 1200 1400] ms.
   const int64_t kMedianFilteredDelaysMs[] = {
       0, 100, 200, 300, 400, 600, 800, 1000};
-  const int kNumPackets = ARRAY_SIZE(kMedianFilteredDelaysMs);
+  const int kNumPackets = arraysize(kMedianFilteredDelaysMs);
   const float kAlpha = 0.1f;  // Used for exponential smoothing.
 
   int64_t exp_smoothed_delays_ms[kNumPackets];
@@ -480,7 +481,7 @@ TEST_F(FilterTest, ExponentialSmoothingConstantArray) {
 
 TEST_F(FilterTest, ExponentialSmoothingInitialPertubation) {
   const int64_t kSignal[] = {90000, 0, 0, 0, 0, 0};
-  const int kNumElements = ARRAY_SIZE(kSignal);
+  const int kNumElements = arraysize(kSignal);
   int64_t exp_smoothed[kNumElements];
   ExponentialSmoothingFilter(kSignal, kNumElements, exp_smoothed);
   for (int i = 1; i < kNumElements; ++i) {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
index b18b9f06b9..9599b01933 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
@@ -17,7 +17,7 @@
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
index a02abc6ab8..b7e4f971fa 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
@@ -16,7 +16,7 @@
 #include "webrtc/base/common.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc b/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
index 6202b4a6a3..559757c0eb 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
@@ -10,10 +10,10 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 
-#include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
-
 #include <algorithm>
 
+#include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
+
 namespace webrtc {
 namespace testing {
 namespace bwe {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet.h b/webrtc/modules/remote_bitrate_estimator/test/packet.h
index 11885a4544..4a361c4dc2 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet.h
@@ -16,7 +16,7 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
index f70c212af7..793e06421f 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
@@ -14,10 +14,10 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
index f1faa49d7e..3bcbc0a071 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
@@ -15,7 +15,7 @@
 #include <sstream>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
index c42647e2d3..f48ed62f57 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
@@ -13,11 +13,12 @@
 
 #include <list>
 #include <limits>
+#include <set>
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 
@@ -149,7 +150,7 @@ class TcpSender : public PacketSender {
  private:
   struct InFlight {
    public:
-    InFlight(const MediaPacket& packet)
+    explicit InFlight(const MediaPacket& packet)
         : sequence_number(packet.header().sequenceNumber),
           time_ms(packet.send_time_ms()) {}
 
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
index 9493805a1c..f138035de5 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -10,15 +10,17 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
 
-#include <sstream>
 #include <stdio.h>
+
+#include <set>
+#include <sstream>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 namespace flags {
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
index 19e4a07b4d..4574faf8b7 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
@@ -14,8 +14,8 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 class Observer : public webrtc::RemoteBitrateObserver {
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index e277481886..bf698728e8 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -14,8 +14,8 @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 int main(int argc, char** argv) {
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
index f2e073aa53..5904594ac8 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
@@ -8,14 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
+
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
-#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
index 58829b072b..93f30e6cee 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
@@ -15,8 +15,8 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
 
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
index b2bc646e2d..64d0e55397 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
@@ -18,9 +18,9 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 using ::testing::_;
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index c651424611..a3d3403172 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -10,14 +10,14 @@ import("../../build/webrtc.gni")
 
 source_set("rtp_rtcp") {
   sources = [
-    "interface/fec_receiver.h",
-    "interface/receive_statistics.h",
-    "interface/remote_ntp_time_estimator.h",
-    "interface/rtp_header_parser.h",
-    "interface/rtp_payload_registry.h",
-    "interface/rtp_receiver.h",
-    "interface/rtp_rtcp.h",
-    "interface/rtp_rtcp_defines.h",
+    "include/fec_receiver.h",
+    "include/receive_statistics.h",
+    "include/remote_ntp_time_estimator.h",
+    "include/rtp_header_parser.h",
+    "include/rtp_payload_registry.h",
+    "include/rtp_receiver.h",
+    "include/rtp_rtcp.h",
+    "include/rtp_rtcp_defines.h",
     "mocks/mock_rtp_rtcp.h",
     "source/bitrate.cc",
     "source/bitrate.h",
@@ -46,8 +46,40 @@ source_set("rtp_rtcp") {
     "source/remote_ntp_time_estimator.cc",
     "source/rtcp_packet.cc",
     "source/rtcp_packet.h",
+    "source/rtcp_packet/app.cc",
+    "source/rtcp_packet/app.h",
+    "source/rtcp_packet/bye.cc",
+    "source/rtcp_packet/bye.h",
+    "source/rtcp_packet/compound_packet.cc",
+    "source/rtcp_packet/compound_packet.h",
+    "source/rtcp_packet/dlrr.cc",
+    "source/rtcp_packet/dlrr.h",
+    "source/rtcp_packet/extended_jitter_report.cc",
+    "source/rtcp_packet/extended_jitter_report.h",
+    "source/rtcp_packet/nack.cc",
+    "source/rtcp_packet/nack.h",
+    "source/rtcp_packet/pli.cc",
+    "source/rtcp_packet/pli.h",
+    "source/rtcp_packet/psfb.cc",
+    "source/rtcp_packet/psfb.h",
+    "source/rtcp_packet/receiver_report.cc",
+    "source/rtcp_packet/receiver_report.h",
+    "source/rtcp_packet/report_block.cc",
+    "source/rtcp_packet/report_block.h",
+    "source/rtcp_packet/rrtr.cc",
+    "source/rtcp_packet/rrtr.h",
+    "source/rtcp_packet/rtpfb.cc",
+    "source/rtcp_packet/rtpfb.h",
+    "source/rtcp_packet/sli.cc",
+    "source/rtcp_packet/sli.h",
+    "source/rtcp_packet/tmmbn.cc",
+    "source/rtcp_packet/tmmbn.h",
+    "source/rtcp_packet/tmmbr.cc",
+    "source/rtcp_packet/tmmbr.h",
     "source/rtcp_packet/transport_feedback.cc",
     "source/rtcp_packet/transport_feedback.h",
+    "source/rtcp_packet/voip_metric.cc",
+    "source/rtcp_packet/voip_metric.h",
     "source/rtcp_receiver.cc",
     "source/rtcp_receiver.h",
     "source/rtcp_receiver_help.cc",
diff --git a/webrtc/modules/rtp_rtcp/OWNERS b/webrtc/modules/rtp_rtcp/OWNERS
index 2b08b6b2dc..fd12dcea0c 100644
--- a/webrtc/modules/rtp_rtcp/OWNERS
+++ b/webrtc/modules/rtp_rtcp/OWNERS
@@ -1,6 +1,11 @@
-stefan@webrtc.org

-henrik.lundin@webrtc.org

-mflodman@webrtc.org

-asapersson@webrtc.org

-

-per-file BUILD.gn=kjellander@webrtc.org

+stefan@webrtc.org
+henrik.lundin@webrtc.org
+mflodman@webrtc.org
+asapersson@webrtc.org
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h b/webrtc/modules/rtp_rtcp/include/fec_receiver.h
index 3608165dab..65e85ad7a5 100644
--- a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
+++ b/webrtc/modules/rtp_rtcp/include/fec_receiver.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -43,4 +43,4 @@ class FecReceiver {
   virtual FecPacketCounter GetPacketCounter() const = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/include/receive_statistics.h
index 6bd5cd846e..b4a7cd0de2 100644
--- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ b/webrtc/modules/rtp_rtcp/include/receive_statistics.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
 
 #include <map>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -99,4 +99,4 @@ class NullReceiveStatistics : public ReceiveStatistics {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
index 0ffba212a6..56c6e48691 100644
--- a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
+++ b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
@@ -48,4 +48,4 @@ class RemoteNtpTimeEstimator {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
index c7a0268ef0..2e30d898ec 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
 
 #include "webrtc/common_video/rotation.h"
 
@@ -51,4 +51,4 @@ inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) {
 }
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
index 2809996b25..329de32611 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
@@ -7,10 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -41,4 +41,4 @@ class RtpHeaderParser {
   virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index 313bef1112..fae864107f 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+
+#include <map>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -27,7 +29,7 @@ class RTPPayloadStrategy {
 
   virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                                    const uint32_t frequency,
-                                   const uint8_t channels,
+                                   const size_t channels,
                                    const uint32_t rate) const = 0;
 
   virtual void UpdatePayloadRate(RtpUtility::Payload* payload,
@@ -37,7 +39,7 @@ class RTPPayloadStrategy {
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payloadType,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) const = 0;
 
   virtual int GetPayloadTypeFrequency(
@@ -52,14 +54,14 @@ class RTPPayloadStrategy {
 class RTPPayloadRegistry {
  public:
   // The registry takes ownership of the strategy.
-  RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
+  explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
   ~RTPPayloadRegistry();
 
   int32_t RegisterReceivePayload(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payload_type,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate,
       bool* created_new_payload_type);
 
@@ -69,7 +71,7 @@ class RTPPayloadRegistry {
   int32_t ReceivePayloadType(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate,
       int8_t* payload_type) const;
 
@@ -108,8 +110,16 @@ class RTPPayloadRegistry {
 
   int GetPayloadTypeFrequency(uint8_t payload_type) const;
 
+  // DEPRECATED. Use PayloadTypeToPayload below that returns const Payload*
+  // instead of taking output parameter.
+  // TODO(danilchap): Remove this when all callers have been updated.
   bool PayloadTypeToPayload(const uint8_t payload_type,
-                            RtpUtility::Payload*& payload) const;
+                            RtpUtility::Payload*& payload) const {  // NOLINT
+    payload =
+        const_cast<RtpUtility::Payload*>(PayloadTypeToPayload(payload_type));
+    return payload != nullptr;
+  }
+  const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
 
   void ResetLastReceivedPayloadTypes() {
     CriticalSectionScoped cs(crit_sect_.get());
@@ -145,7 +155,7 @@ class RTPPayloadRegistry {
   int8_t last_received_media_payload_type() const {
     CriticalSectionScoped cs(crit_sect_.get());
     return last_received_media_payload_type_;
-  };
+  }
 
   bool use_rtx_payload_mapping_on_restore() const {
     CriticalSectionScoped cs(crit_sect_.get());
@@ -163,7 +173,7 @@ class RTPPayloadRegistry {
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const size_t payload_name_length,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate);
 
   bool IsRtxInternal(const RTPHeader& header) const;
@@ -190,4 +200,4 @@ class RTPPayloadRegistry {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
index 2fb8ac5d61..0640d5cc19 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -61,7 +61,7 @@ class RtpReceiver {
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payload_type,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) = 0;
 
   // De-registers |payload_type| from the payload registry.
@@ -100,4 +100,4 @@ class RtpReceiver {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index f907408573..6a7022a94c 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -8,14 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
 
 #include <set>
+#include <utility>
 #include <vector>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 // Forward declarations.
@@ -580,9 +581,13 @@ class RtpRtcp : public Module {
     *
     *   return -1 on failure else 0
     */
-     virtual int32_t SendREDPayloadType(
-         int8_t& payloadType) const = 0;
-
+    // DEPRECATED. Use SendREDPayloadType below that takes output parameter
+    // by pointer instead of by reference.
+    // TODO(danilchap): Remove this when all callers have been updated.
+    int32_t SendREDPayloadType(int8_t& payloadType) const {  // NOLINT
+      return SendREDPayloadType(&payloadType);
+    }
+    virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 0;
      /*
      * Store the audio level in dBov for header-extension-for-audio-level-
      * indication.
@@ -614,10 +619,17 @@ class RtpRtcp : public Module {
     /*
     *   Get generic FEC setting
     */
-    virtual void GenericFECStatus(bool& enable,
-                                     uint8_t& payloadTypeRED,
-                                     uint8_t& payloadTypeFEC) = 0;
-
+    // DEPRECATED. Use GenericFECStatus below that takes output parameters
+    // by pointers instead of by references.
+    // TODO(danilchap): Remove this when all callers have been updated.
+    void GenericFECStatus(bool& enable,               // NOLINT
+                          uint8_t& payloadTypeRED,    // NOLINT
+                          uint8_t& payloadTypeFEC) {  // NOLINT
+      GenericFECStatus(&enable, &payloadTypeRED, &payloadTypeFEC);
+    }
+    virtual void GenericFECStatus(bool* enable,
+                                  uint8_t* payload_type_red,
+                                  uint8_t* payload_type_fec) = 0;
 
     virtual int32_t SetFecParameters(
         const FecProtectionParams* delta_params,
@@ -638,4 +650,4 @@ class RtpRtcp : public Module {
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 6936352aca..fad97f19cc 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
 
 #include <stddef.h>
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
@@ -33,31 +33,23 @@ const int kVideoPayloadTypeFrequency = 90000;
 // Minimum RTP header size in bytes.
 const uint8_t kRtpHeaderSize = 12;
 
-struct AudioPayload
-{
+struct AudioPayload {
     uint32_t    frequency;
-    uint8_t     channels;
+    size_t      channels;
     uint32_t    rate;
 };
 
-struct VideoPayload
-{
+struct VideoPayload {
     RtpVideoCodecTypes   videoCodecType;
     uint32_t       maxRate;
 };
 
-union PayloadUnion
-{
+union PayloadUnion {
     AudioPayload Audio;
     VideoPayload Video;
 };
 
-enum RTPAliveType
-{
-    kRtpDead   = 0,
-    kRtpNoRtp = 1,
-    kRtpAlive  = 2
-};
+enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
 
 enum ProtectionType {
   kUnprotectedPacket,
@@ -78,10 +70,7 @@ enum RTPExtensionType {
   kRtpExtensionTransportSequenceNumber,
 };
 
-enum RTCPAppSubTypes
-{
-    kAppSubtypeBwe     = 0x00
-};
+enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
 
 // TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
 enum RTCPPacketType : uint32_t {
@@ -109,17 +98,9 @@ enum RTCPPacketType : uint32_t {
 
 enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
 
-enum RtpRtcpPacketType
-{
-    kPacketRtp        = 0,
-    kPacketKeepAlive = 1
-};
+enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 };
 
-enum NACKMethod
-{
-    kNackOff      = 0,
-    kNackRtcp     = 2
-};
+enum NACKMethod { kNackOff = 0, kNackRtcp = 2 };
 
 enum RetransmissionMode : uint8_t {
   kRetransmitOff = 0x0,
@@ -138,8 +119,7 @@ enum RtxMode {
 
 const size_t kRtxHeaderSize = 2;
 
-struct RTCPSenderInfo
-{
+struct RTCPSenderInfo {
     uint32_t NTPseconds;
     uint32_t NTPfraction;
     uint32_t RTPtimeStamp;
@@ -206,40 +186,36 @@ struct RtpState {
   bool media_has_been_sent;
 };
 
-class RtpData
-{
-public:
-    virtual ~RtpData() {}
+class RtpData {
+ public:
+  virtual ~RtpData() {}
 
-    virtual int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const WebRtcRTPHeader* rtpHeader) = 0;
+  virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData,
+                                        const size_t payloadSize,
+                                        const WebRtcRTPHeader* rtpHeader) = 0;
 
-    virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                   size_t packet_length) = 0;
+  virtual bool OnRecoveredPacket(const uint8_t* packet,
+                                 size_t packet_length) = 0;
 };
 
-class RtpFeedback
-{
-public:
-    virtual ~RtpFeedback() {}
-
-    // Receiving payload change or SSRC change. (return success!)
-    /*
-    *   channels    - number of channels in codec (1 = mono, 2 = stereo)
-    */
-    virtual int32_t OnInitializeDecoder(
-        const int8_t payloadType,
-        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-        const int frequency,
-        const uint8_t channels,
-        const uint32_t rate) = 0;
-
-    virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
-
-    virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
-                                       const bool added) = 0;
+class RtpFeedback {
+ public:
+  virtual ~RtpFeedback() {}
+
+  // Receiving payload change or SSRC change. (return success!)
+  /*
+  *   channels    - number of channels in codec (1 = mono, 2 = stereo)
+  */
+  virtual int32_t OnInitializeDecoder(
+      const int8_t payloadType,
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      const int frequency,
+      const size_t channels,
+      const uint32_t rate) = 0;
+
+  virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
+
+  virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0;
 };
 
 class RtpAudioFeedback {
@@ -346,7 +322,7 @@ class RtcpRttStats {
 
   virtual int64_t LastProcessedRtt() const = 0;
 
-  virtual ~RtcpRttStats() {};
+  virtual ~RtcpRttStats() {}
 };
 
 // Null object version of RtpFeedback.
@@ -357,7 +333,7 @@ class NullRtpFeedback : public RtpFeedback {
   int32_t OnInitializeDecoder(const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
-                              const uint8_t channels,
+                              const size_t channels,
                               const uint32_t rate) override {
     return 0;
   }
@@ -437,4 +413,4 @@ class TransportSequenceNumberAllocator {
 };
 
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index bc4aec8967..796be1304c 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -11,11 +11,15 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
 #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
 
+#include <set>
+#include <utility>
+#include <vector>
+
 #include "testing/gmock/include/gmock/gmock.h"
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
@@ -108,7 +112,10 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD0(SendingMedia,
       bool());
   MOCK_CONST_METHOD4(BitrateSent,
-      void(uint32_t* totalRate, uint32_t* videoRate, uint32_t* fecRate, uint32_t* nackRate));
+                     void(uint32_t* totalRate,
+                          uint32_t* videoRate,
+                          uint32_t* fecRate,
+                          uint32_t* nackRate));
   MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver*));
   MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver*(void));
   MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
@@ -172,7 +179,10 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD1(RemoteRTCPStat,
       int32_t(std::vector<RTCPReportBlock>* receiveBlocks));
   MOCK_METHOD4(SetRTCPApplicationSpecificData,
-      int32_t(const uint8_t subType, const uint32_t name, const uint8_t* data, const uint16_t length));
+               int32_t(const uint8_t subType,
+                       const uint32_t name,
+                       const uint8_t* data,
+                       const uint16_t length));
   MOCK_METHOD1(SetRTCPVoIPMetrics,
       int32_t(const RTCPVoIPMetric* VoIPMetric));
   MOCK_METHOD1(SetRtcpXrRrtrStatus,
@@ -214,12 +224,9 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level));
   MOCK_METHOD1(SetSendREDPayloadType,
       int32_t(const int8_t payloadType));
-  MOCK_CONST_METHOD1(SendREDPayloadType,
-      int32_t(int8_t& payloadType));
+  MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t* payloadType));
   MOCK_METHOD2(SetRTPAudioLevelIndicationStatus,
       int32_t(const bool enable, const uint8_t ID));
-  MOCK_CONST_METHOD2(GetRTPAudioLevelIndicationStatus,
-      int32_t(bool& enable, uint8_t& ID));
   MOCK_METHOD1(SetAudioLevel,
       int32_t(const uint8_t level_dBov));
   MOCK_METHOD1(SetTargetSendBitrate,
@@ -229,9 +236,9 @@ class MockRtpRtcp : public RtpRtcp {
                     const uint8_t payload_type_red,
                     const uint8_t payload_type_fec));
   MOCK_METHOD3(GenericFECStatus,
-               void(bool& enable,
-                    uint8_t& payloadTypeRED,
-                    uint8_t& payloadTypeFEC));
+               void(bool* enable,
+                    uint8_t* payloadTypeRED,
+                    uint8_t* payloadTypeFEC));
   MOCK_METHOD2(SetFecParameters,
       int32_t(const FecProtectionParams* delta_params,
               const FecProtectionParams* key_params));
@@ -239,8 +246,6 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const KeyFrameRequestMethod method));
   MOCK_METHOD0(RequestKeyFrame,
       int32_t());
-  MOCK_CONST_METHOD3(Version,
-      int32_t(char* version, uint32_t& remaining_buffer_in_bytes, uint32_t& position));
   MOCK_METHOD0(TimeUntilNextProcess,
         int64_t());
   MOCK_METHOD0(Process,
diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
index e35a75cd0f..d340f746be 100644
--- a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
+++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
@@ -17,14 +17,14 @@
       ],
       'sources': [
         # Common
-        'interface/fec_receiver.h',
-        'interface/receive_statistics.h',
-        'interface/remote_ntp_time_estimator.h',
-        'interface/rtp_header_parser.h',
-        'interface/rtp_payload_registry.h',
-        'interface/rtp_receiver.h',
-        'interface/rtp_rtcp.h',
-        'interface/rtp_rtcp_defines.h',
+        'include/fec_receiver.h',
+        'include/receive_statistics.h',
+        'include/remote_ntp_time_estimator.h',
+        'include/rtp_header_parser.h',
+        'include/rtp_payload_registry.h',
+        'include/rtp_receiver.h',
+        'include/rtp_rtcp.h',
+        'include/rtp_rtcp_defines.h',
         'source/bitrate.cc',
         'source/bitrate.h',
         'source/byte_io.h',
@@ -41,8 +41,40 @@
         'source/rtp_rtcp_impl.h',
         'source/rtcp_packet.cc',
         'source/rtcp_packet.h',
+        'source/rtcp_packet/app.cc',
+        'source/rtcp_packet/app.h',
+        'source/rtcp_packet/bye.cc',
+        'source/rtcp_packet/bye.h',
+        'source/rtcp_packet/compound_packet.cc',
+        'source/rtcp_packet/compound_packet.h',
+        'source/rtcp_packet/dlrr.cc',
+        'source/rtcp_packet/dlrr.h',
+        'source/rtcp_packet/extended_jitter_report.cc',
+        'source/rtcp_packet/extended_jitter_report.h',
+        'source/rtcp_packet/nack.cc',
+        'source/rtcp_packet/nack.h',
+        'source/rtcp_packet/pli.cc',
+        'source/rtcp_packet/pli.h',
+        'source/rtcp_packet/psfb.cc',
+        'source/rtcp_packet/psfb.h',
+        'source/rtcp_packet/receiver_report.cc',
+        'source/rtcp_packet/receiver_report.h',
+        'source/rtcp_packet/report_block.cc',
+        'source/rtcp_packet/report_block.h',
+        'source/rtcp_packet/rrtr.cc',
+        'source/rtcp_packet/rrtr.h',
+        'source/rtcp_packet/rtpfb.cc',
+        'source/rtcp_packet/rtpfb.h',
+        'source/rtcp_packet/sli.cc',
+        'source/rtcp_packet/sli.h',
+        'source/rtcp_packet/tmmbn.cc',
+        'source/rtcp_packet/tmmbn.h',
+        'source/rtcp_packet/tmmbr.cc',
+        'source/rtcp_packet/tmmbr.h',
         'source/rtcp_packet/transport_feedback.cc',
         'source/rtcp_packet/transport_feedback.h',
+        'source/rtcp_packet/voip_metric.cc',
+        'source/rtcp_packet/voip_metric.h',
         'source/rtcp_receiver.cc',
         'source/rtcp_receiver.h',
         'source/rtcp_receiver_help.cc',
diff --git a/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
new file mode 100644
index 0000000000..c318452482
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
@@ -0,0 +1,6 @@
+#tmmbr_help is refactored in CL#1474693002
+exclude_files=tmmbr_help.*
+#rtcp_utility planned to be removed when webrtc:5260 will be finished.
+exclude_files=rtcp_utility.*
+#rtcp_receiver/rtcp_receiver_help will be refactored more deeply as part of webrtc:5260
+exclude_files=rtcp_receiver.*
diff --git a/webrtc/modules/rtp_rtcp/source/byte_io.h b/webrtc/modules/rtp_rtcp/source/byte_io.h
index d8903b8483..c69c178078 100644
--- a/webrtc/modules/rtp_rtcp/source/byte_io.h
+++ b/webrtc/modules/rtp_rtcp/source/byte_io.h
@@ -51,17 +51,14 @@ namespace webrtc {
 // platform that doesn't use two's complement, implement conversion to/from
 // wire format.
 
-namespace {
-inline void AssertTwosComplement() {
-  // Assume the if any one signed integer type is two's complement, then all
-  // other will be too.
-  static_assert(
-      (-1 & 0x03) == 0x03,
-      "Only two's complement representation of signed integers supported.");
-}
+// Assume the if any one signed integer type is two's complement, then all
+// other will be too.
+static_assert(
+    (-1 & 0x03) == 0x03,
+    "Only two's complement representation of signed integers supported.");
+
 // Plain const char* won't work for static_assert, use #define instead.
 #define kSizeErrorMsg "Byte size must be less than or equal to data type size."
-}
 
 // Utility class for getting the unsigned equivalent of a signed type.
 template <typename T>
diff --git a/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc b/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
index becea912ab..ab21b8704a 100644
--- a/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
+++ b/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
 
-#include <string.h>  //memset
+#include <string.h>
 
 namespace webrtc {
 DTMFqueue::DTMFqueue()
@@ -21,7 +21,9 @@ DTMFqueue::DTMFqueue()
   memset(dtmf_level_, 0, sizeof(dtmf_level_));
 }
 
-DTMFqueue::~DTMFqueue() { delete dtmf_critsect_; }
+DTMFqueue::~DTMFqueue() {
+  delete dtmf_critsect_;
+}
 
 int32_t DTMFqueue::AddDTMF(uint8_t key, uint16_t len, uint8_t level) {
   CriticalSectionScoped lock(dtmf_critsect_);
diff --git a/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h b/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
index 6105ae1d24..0b39908bb1 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
@@ -27,7 +27,8 @@
 
 #include "webrtc/typedefs.h"
 
-namespace {
+namespace webrtc {
+namespace fec_private_tables {
 
 const uint8_t kMaskBursty1_1[2] = {
   0x80, 0x00
@@ -756,5 +757,6 @@ const uint8_t** kPacketMaskBurstyTbl[12] = {
   kPacketMaskBursty12
 };
 
-}  // namespace
+}  // namespace fec_private_tables
+}  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
diff --git a/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h b/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
index ff6de43b76..295d749873 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
 
 // This file contains a set of packets masks for the FEC code. The masks in
 // this table are specifically designed to favor recovery to random loss.
@@ -17,7 +17,8 @@
 
 #include "webrtc/typedefs.h"
 
-namespace {
+namespace webrtc {
+namespace fec_private_tables {
 
 const uint8_t kMaskRandom10_1[2] = {
   0xff, 0xc0
@@ -24518,5 +24519,6 @@ const uint8_t** kPacketMaskRandomTbl[48] = {
   kPacketMaskRandom48
 };
 
-}  // namespace
-#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
+}  // namespace fec_private_tables
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index 24db39b902..6a63813f40 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -14,8 +14,8 @@
 // This header is included to get the nested declaration of Packet structure.
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/typedefs.h"
 
@@ -25,7 +25,7 @@ class CriticalSectionWrapper;
 
 class FecReceiverImpl : public FecReceiver {
  public:
-  FecReceiverImpl(RtpData* callback);
+  explicit FecReceiverImpl(RtpData* callback);
   virtual ~FecReceiverImpl();
 
   int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
index f64b537a52..bb22e1d580 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -15,8 +15,8 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
diff --git a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
index e1791adba3..aacc2d1ecc 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
 namespace webrtc {
@@ -54,6 +54,6 @@ class FrameGenerator {
   uint16_t seq_num_;
   uint32_t timestamp_;
 };
-}
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
index aad418f015..b85d813790 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -10,15 +10,15 @@
 
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
-#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
 #include <algorithm>
 #include <iterator>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
 
@@ -36,11 +36,11 @@ const uint8_t kUlpHeaderSizeLBitClear = (2 + kMaskSizeLBitClear);
 // Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
 const uint8_t kTransportOverhead = 28;
 
-enum {
-  kMaxFecPackets = ForwardErrorCorrection::kMaxMediaPackets
-};
+enum { kMaxFecPackets = ForwardErrorCorrection::kMaxMediaPackets };
 
-int32_t ForwardErrorCorrection::Packet::AddRef() { return ++ref_count_; }
+int32_t ForwardErrorCorrection::Packet::AddRef() {
+  return ++ref_count_;
+}
 
 int32_t ForwardErrorCorrection::Packet::Release() {
   int32_t ref_count;
@@ -72,7 +72,8 @@ class FecPacket : public ForwardErrorCorrection::SortablePacket {
 };
 
 bool ForwardErrorCorrection::SortablePacket::LessThan(
-    const SortablePacket* first, const SortablePacket* second) {
+    const SortablePacket* first,
+    const SortablePacket* second) {
   return IsNewerSequenceNumber(second->seq_num, first->seq_num);
 }
 
@@ -83,8 +84,7 @@ ForwardErrorCorrection::RecoveredPacket::RecoveredPacket() {}
 ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() {}
 
 ForwardErrorCorrection::ForwardErrorCorrection()
-    : generated_fec_packets_(kMaxMediaPackets),
-      fec_packet_received_(false) {}
+    : generated_fec_packets_(kMaxMediaPackets), fec_packet_received_(false) {}
 
 ForwardErrorCorrection::~ForwardErrorCorrection() {}
 
@@ -112,7 +112,6 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
                                             FecMaskType fec_mask_type,
                                             PacketList* fec_packet_list) {
   const uint16_t num_media_packets = media_packet_list.size();
-
   // Sanity check arguments.
   assert(num_media_packets > 0);
   assert(num_important_packets >= 0 &&
@@ -126,12 +125,10 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
   }
 
   bool l_bit = (num_media_packets > 8 * kMaskSizeLBitClear);
-  int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+  int num_mask_bytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
 
   // Do some error checking on the media packets.
-  PacketList::const_iterator media_list_it = media_packet_list.begin();
-  while (media_list_it != media_packet_list.end()) {
-    Packet* media_packet = *media_list_it;
+  for (Packet* media_packet : media_packet_list) {
     assert(media_packet);
 
     if (media_packet->length < kRtpHeaderSize) {
@@ -146,7 +143,6 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
       LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
                       << "with overhead is larger than " << IP_PACKET_SIZE;
     }
-    media_list_it++;
   }
 
   int num_fec_packets =
@@ -167,29 +163,29 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
 
   // -- Generate packet masks --
   // Always allocate space for a large mask.
-  uint8_t* packet_mask = new uint8_t[num_fec_packets * kMaskSizeLBitSet];
-  memset(packet_mask, 0, num_fec_packets * num_maskBytes);
+  rtc::scoped_ptr<uint8_t[]> packet_mask(
+      new uint8_t[num_fec_packets * kMaskSizeLBitSet]);
+  memset(packet_mask.get(), 0, num_fec_packets * num_mask_bytes);
   internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
                                 num_important_packets, use_unequal_protection,
-                                mask_table, packet_mask);
-
-  int num_maskBits = InsertZerosInBitMasks(media_packet_list, packet_mask,
-                                           num_maskBytes, num_fec_packets);
+                                mask_table, packet_mask.get());
 
-  l_bit = (num_maskBits > 8 * kMaskSizeLBitClear);
+  int num_mask_bits = InsertZerosInBitMasks(
+      media_packet_list, packet_mask.get(), num_mask_bytes, num_fec_packets);
 
-  if (num_maskBits < 0) {
-    delete[] packet_mask;
+  if (num_mask_bits < 0) {
     return -1;
   }
+  l_bit = (num_mask_bits > 8 * kMaskSizeLBitClear);
   if (l_bit) {
-    num_maskBytes = kMaskSizeLBitSet;
+    num_mask_bytes = kMaskSizeLBitSet;
   }
 
-  GenerateFecBitStrings(media_packet_list, packet_mask, num_fec_packets, l_bit);
-  GenerateFecUlpHeaders(media_packet_list, packet_mask, l_bit, num_fec_packets);
+  GenerateFecBitStrings(media_packet_list, packet_mask.get(), num_fec_packets,
+                        l_bit);
+  GenerateFecUlpHeaders(media_packet_list, packet_mask.get(), l_bit,
+                        num_fec_packets);
 
-  delete[] packet_mask;
   return 0;
 }
 
@@ -206,26 +202,30 @@ int ForwardErrorCorrection::GetNumberOfFecPackets(int num_media_packets,
 }
 
 void ForwardErrorCorrection::GenerateFecBitStrings(
-    const PacketList& media_packet_list, uint8_t* packet_mask,
-    int num_fec_packets, bool l_bit) {
+    const PacketList& media_packet_list,
+    uint8_t* packet_mask,
+    int num_fec_packets,
+    bool l_bit) {
   if (media_packet_list.empty()) {
     return;
   }
   uint8_t media_payload_length[2];
-  const int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+  const int num_mask_bytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
   const uint16_t ulp_header_size =
       l_bit ? kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
   const uint16_t fec_rtp_offset =
       kFecHeaderSize + ulp_header_size - kRtpHeaderSize;
 
   for (int i = 0; i < num_fec_packets; ++i) {
+    Packet* const fec_packet = &generated_fec_packets_[i];
     PacketList::const_iterator media_list_it = media_packet_list.begin();
-    uint32_t pkt_mask_idx = i * num_maskBytes;
+    uint32_t pkt_mask_idx = i * num_mask_bytes;
     uint32_t media_pkt_idx = 0;
     uint16_t fec_packet_length = 0;
     uint16_t prev_seq_num = ParseSequenceNumber((*media_list_it)->data);
     while (media_list_it != media_packet_list.end()) {
-      // Each FEC packet has a multiple byte mask.
+      // Each FEC packet has a multiple byte mask. Determine if this media
+      // packet should be included in FEC packet i.
       if (packet_mask[pkt_mask_idx] & (1 << (7 - media_pkt_idx))) {
         Packet* media_packet = *media_list_it;
 
@@ -235,42 +235,40 @@ void ForwardErrorCorrection::GenerateFecBitStrings(
 
         fec_packet_length = media_packet->length + fec_rtp_offset;
         // On the first protected packet, we don't need to XOR.
-        if (generated_fec_packets_[i].length == 0) {
+        if (fec_packet->length == 0) {
           // Copy the first 2 bytes of the RTP header.
-          memcpy(generated_fec_packets_[i].data, media_packet->data, 2);
+          memcpy(fec_packet->data, media_packet->data, 2);
           // Copy the 5th to 8th bytes of the RTP header.
-          memcpy(&generated_fec_packets_[i].data[4], &media_packet->data[4], 4);
+          memcpy(&fec_packet->data[4], &media_packet->data[4], 4);
           // Copy network-ordered payload size.
-          memcpy(&generated_fec_packets_[i].data[8], media_payload_length, 2);
+          memcpy(&fec_packet->data[8], media_payload_length, 2);
 
           // Copy RTP payload, leaving room for the ULP header.
-          memcpy(
-              &generated_fec_packets_[i].data[kFecHeaderSize + ulp_header_size],
-              &media_packet->data[kRtpHeaderSize],
-              media_packet->length - kRtpHeaderSize);
+          memcpy(&fec_packet->data[kFecHeaderSize + ulp_header_size],
+                 &media_packet->data[kRtpHeaderSize],
+                 media_packet->length - kRtpHeaderSize);
         } else {
           // XOR with the first 2 bytes of the RTP header.
-          generated_fec_packets_[i].data[0] ^= media_packet->data[0];
-          generated_fec_packets_[i].data[1] ^= media_packet->data[1];
+          fec_packet->data[0] ^= media_packet->data[0];
+          fec_packet->data[1] ^= media_packet->data[1];
 
           // XOR with the 5th to 8th bytes of the RTP header.
           for (uint32_t j = 4; j < 8; ++j) {
-            generated_fec_packets_[i].data[j] ^= media_packet->data[j];
+            fec_packet->data[j] ^= media_packet->data[j];
           }
 
           // XOR with the network-ordered payload size.
-          generated_fec_packets_[i].data[8] ^= media_payload_length[0];
-          generated_fec_packets_[i].data[9] ^= media_payload_length[1];
+          fec_packet->data[8] ^= media_payload_length[0];
+          fec_packet->data[9] ^= media_payload_length[1];
 
           // XOR with RTP payload, leaving room for the ULP header.
           for (int32_t j = kFecHeaderSize + ulp_header_size;
                j < fec_packet_length; j++) {
-            generated_fec_packets_[i].data[j] ^=
-                media_packet->data[j - fec_rtp_offset];
+            fec_packet->data[j] ^= media_packet->data[j - fec_rtp_offset];
           }
         }
-        if (fec_packet_length > generated_fec_packets_[i].length) {
-          generated_fec_packets_[i].length = fec_packet_length;
+        if (fec_packet_length > fec_packet->length) {
+          fec_packet->length = fec_packet_length;
         }
       }
       media_list_it++;
@@ -279,19 +277,18 @@ void ForwardErrorCorrection::GenerateFecBitStrings(
         media_pkt_idx += static_cast<uint16_t>(seq_num - prev_seq_num);
         prev_seq_num = seq_num;
       }
-      if (media_pkt_idx == 8) {
-        // Switch to the next mask byte.
-        media_pkt_idx = 0;
-        pkt_mask_idx++;
-      }
+      pkt_mask_idx += media_pkt_idx / 8;
+      media_pkt_idx %= 8;
     }
-    assert(generated_fec_packets_[i].length);
-    //Note: This shouldn't happen: means packet mask is wrong or poorly designed
+    RTC_DCHECK_GT(fec_packet->length, 0u)
+        << "Packet mask is wrong or poorly designed.";
   }
 }
 
 int ForwardErrorCorrection::InsertZerosInBitMasks(
-    const PacketList& media_packets, uint8_t* packet_mask, int num_mask_bytes,
+    const PacketList& media_packets,
+    uint8_t* packet_mask,
+    int num_mask_bytes,
     int num_fec_packets) {
   uint8_t* new_mask = NULL;
   if (media_packets.size() <= 1) {
@@ -307,6 +304,9 @@ int ForwardErrorCorrection::InsertZerosInBitMasks(
     // required.
     return media_packets.size();
   }
+  // We can only protect 8 * kMaskSizeLBitSet packets.
+  if (total_missing_seq_nums + media_packets.size() > 8 * kMaskSizeLBitSet)
+    return -1;
   // Allocate the new mask.
   int new_mask_bytes = kMaskSizeLBitClear;
   if (media_packets.size() + total_missing_seq_nums > 8 * kMaskSizeLBitClear) {
@@ -357,7 +357,8 @@ int ForwardErrorCorrection::InsertZerosInBitMasks(
   return new_bit_index;
 }
 
-void ForwardErrorCorrection::InsertZeroColumns(int num_zeros, uint8_t* new_mask,
+void ForwardErrorCorrection::InsertZeroColumns(int num_zeros,
+                                               uint8_t* new_mask,
                                                int new_mask_bytes,
                                                int num_fec_packets,
                                                int new_bit_index) {
@@ -368,9 +369,12 @@ void ForwardErrorCorrection::InsertZeroColumns(int num_zeros, uint8_t* new_mask,
   }
 }
 
-void ForwardErrorCorrection::CopyColumn(uint8_t* new_mask, int new_mask_bytes,
-                                        uint8_t* old_mask, int old_mask_bytes,
-                                        int num_fec_packets, int new_bit_index,
+void ForwardErrorCorrection::CopyColumn(uint8_t* new_mask,
+                                        int new_mask_bytes,
+                                        uint8_t* old_mask,
+                                        int old_mask_bytes,
+                                        int num_fec_packets,
+                                        int new_bit_index,
                                         int old_bit_index) {
   // Copy column from the old mask to the beginning of the new mask and shift it
   // out from the old mask.
@@ -386,7 +390,9 @@ void ForwardErrorCorrection::CopyColumn(uint8_t* new_mask, int new_mask_bytes,
 }
 
 void ForwardErrorCorrection::GenerateFecUlpHeaders(
-    const PacketList& media_packet_list, uint8_t* packet_mask, bool l_bit,
+    const PacketList& media_packet_list,
+    uint8_t* packet_mask,
+    bool l_bit,
     int num_fec_packets) {
   // -- Generate FEC and ULP headers --
   //
@@ -412,33 +418,34 @@ void ForwardErrorCorrection::GenerateFecUlpHeaders(
   PacketList::const_iterator media_list_it = media_packet_list.begin();
   Packet* media_packet = *media_list_it;
   assert(media_packet != NULL);
-  int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+  int num_mask_bytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
   const uint16_t ulp_header_size =
       l_bit ? kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
 
   for (int i = 0; i < num_fec_packets; ++i) {
+    Packet* const fec_packet = &generated_fec_packets_[i];
     // -- FEC header --
-    generated_fec_packets_[i].data[0] &= 0x7f;  // Set E to zero.
+    fec_packet->data[0] &= 0x7f;  // Set E to zero.
     if (l_bit == 0) {
-      generated_fec_packets_[i].data[0] &= 0xbf;  // Clear the L bit.
+      fec_packet->data[0] &= 0xbf;  // Clear the L bit.
     } else {
-      generated_fec_packets_[i].data[0] |= 0x40;  // Set the L bit.
+      fec_packet->data[0] |= 0x40;  // Set the L bit.
     }
     // Two byte sequence number from first RTP packet to SN base.
     // We use the same sequence number base for every FEC packet,
     // but that's not required in general.
-    memcpy(&generated_fec_packets_[i].data[2], &media_packet->data[2], 2);
+    memcpy(&fec_packet->data[2], &media_packet->data[2], 2);
 
     // -- ULP header --
     // Copy the payload size to the protection length field.
     // (We protect the entire packet.)
     ByteWriter<uint16_t>::WriteBigEndian(
-        &generated_fec_packets_[i].data[10],
-        generated_fec_packets_[i].length - kFecHeaderSize - ulp_header_size);
+        &fec_packet->data[10],
+        fec_packet->length - kFecHeaderSize - ulp_header_size);
 
     // Copy the packet mask.
-    memcpy(&generated_fec_packets_[i].data[12], &packet_mask[i * num_maskBytes],
-           num_maskBytes);
+    memcpy(&fec_packet->data[12], &packet_mask[i * num_mask_bytes],
+           num_mask_bytes);
   }
 }
 
@@ -460,7 +467,7 @@ void ForwardErrorCorrection::ResetState(
     ProtectedPacketList::iterator protected_packet_list_it;
     protected_packet_list_it = fec_packet->protected_pkt_list.begin();
     while (protected_packet_list_it != fec_packet->protected_pkt_list.end()) {
-      delete* protected_packet_list_it;
+      delete *protected_packet_list_it;
       protected_packet_list_it =
           fec_packet->protected_pkt_list.erase(protected_packet_list_it);
     }
@@ -472,7 +479,8 @@ void ForwardErrorCorrection::ResetState(
 }
 
 void ForwardErrorCorrection::InsertMediaPacket(
-    ReceivedPacket* rx_packet, RecoveredPacketList* recovered_packet_list) {
+    ReceivedPacket* rx_packet,
+    RecoveredPacketList* recovered_packet_list) {
   RecoveredPacketList::iterator recovered_packet_list_it =
       recovered_packet_list->begin();
 
@@ -538,9 +546,9 @@ void ForwardErrorCorrection::InsertFECPacket(
 
   const uint16_t seq_num_base =
       ByteReader<uint16_t>::ReadBigEndian(&fec_packet->pkt->data[2]);
-  const uint16_t maskSizeBytes =
-      (fec_packet->pkt->data[0] & 0x40) ? kMaskSizeLBitSet
-                                        : kMaskSizeLBitClear;  // L bit set?
+  const uint16_t maskSizeBytes = (fec_packet->pkt->data[0] & 0x40)
+                                     ? kMaskSizeLBitSet
+                                     : kMaskSizeLBitClear;  // L bit set?
 
   for (uint16_t byte_idx = 0; byte_idx < maskSizeBytes; ++byte_idx) {
     uint8_t packet_mask = fec_packet->pkt->data[12 + byte_idx];
@@ -574,7 +582,8 @@ void ForwardErrorCorrection::InsertFECPacket(
 }
 
 void ForwardErrorCorrection::AssignRecoveredPackets(
-    FecPacket* fec_packet, const RecoveredPacketList* recovered_packets) {
+    FecPacket* fec_packet,
+    const RecoveredPacketList* recovered_packets) {
   // Search for missing packets which have arrived or have been recovered by
   // another FEC packet.
   ProtectedPacketList* not_recovered = &fec_packet->protected_pkt_list;
@@ -599,7 +608,6 @@ void ForwardErrorCorrection::AssignRecoveredPackets(
 void ForwardErrorCorrection::InsertPackets(
     ReceivedPacketList* received_packet_list,
     RecoveredPacketList* recovered_packet_list) {
-
   while (!received_packet_list->empty()) {
     ReceivedPacket* rx_packet = received_packet_list->front();
 
@@ -611,9 +619,9 @@ void ForwardErrorCorrection::InsertPackets(
     // old FEC packets based on timestamp information or better sequence number
     // thresholding (e.g., to distinguish between wrap-around and reordering).
     if (!fec_packet_list_.empty()) {
-      uint16_t seq_num_diff = abs(
-          static_cast<int>(rx_packet->seq_num) -
-          static_cast<int>(fec_packet_list_.front()->seq_num));
+      uint16_t seq_num_diff =
+          abs(static_cast<int>(rx_packet->seq_num) -
+              static_cast<int>(fec_packet_list_.front()->seq_num));
       if (seq_num_diff > 0x3fff) {
         DiscardFECPacket(fec_packet_list_.front());
         fec_packet_list_.pop_front();
@@ -637,9 +645,9 @@ void ForwardErrorCorrection::InsertPackets(
 bool ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet,
                                           RecoveredPacket* recovered) {
   // This is the first packet which we try to recover with.
-  const uint16_t ulp_header_size =
-      fec_packet->pkt->data[0] & 0x40 ? kUlpHeaderSizeLBitSet
-                                      : kUlpHeaderSizeLBitClear;  // L bit set?
+  const uint16_t ulp_header_size = fec_packet->pkt->data[0] & 0x40
+                                       ? kUlpHeaderSizeLBitSet
+                                       : kUlpHeaderSizeLBitClear;  // L bit set?
   if (fec_packet->pkt->length <
       static_cast<size_t>(kFecHeaderSize + ulp_header_size)) {
     LOG(LS_WARNING)
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
index f92f014db3..9ba6ce0438 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/ref_count.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
index 6d9be90de1..fae59078b1 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
@@ -17,6 +17,8 @@
 #include "webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h"
 
 namespace {
+using webrtc::fec_private_tables::kPacketMaskBurstyTbl;
+using webrtc::fec_private_tables::kPacketMaskRandomTbl;
 
 // Allow for different modes of protection for packets in UEP case.
 enum ProtectionMode {
@@ -37,8 +39,11 @@ enum ProtectionMode {
 //                                [0, num_rows * num_sub_mask_bytes]
 // \param[out] packet_mask        A pointer to hold the output mask, of size
 //                                [0, x * num_mask_bytes], where x >= num_rows.
-void FitSubMask(int num_mask_bytes, int num_sub_mask_bytes, int num_rows,
-                const uint8_t* sub_mask, uint8_t* packet_mask) {
+void FitSubMask(int num_mask_bytes,
+                int num_sub_mask_bytes,
+                int num_rows,
+                const uint8_t* sub_mask,
+                uint8_t* packet_mask) {
   if (num_mask_bytes == num_sub_mask_bytes) {
     memcpy(packet_mask, sub_mask, num_rows * num_sub_mask_bytes);
   } else {
@@ -70,13 +75,15 @@ void FitSubMask(int num_mask_bytes, int num_sub_mask_bytes, int num_rows,
 // \param[out] packet_mask        A pointer to hold the output mask, of size
 //                                [0, x * num_mask_bytes],
 //                                where x >= end_row_fec.
-// TODO (marpan): This function is doing three things at the same time:
+// TODO(marpan): This function is doing three things at the same time:
 // shift within a byte, byte shift and resizing.
 // Split up into subroutines.
-void ShiftFitSubMask(int num_mask_bytes, int res_mask_bytes,
-                     int num_column_shift, int end_row, const uint8_t* sub_mask,
+void ShiftFitSubMask(int num_mask_bytes,
+                     int res_mask_bytes,
+                     int num_column_shift,
+                     int end_row,
+                     const uint8_t* sub_mask,
                      uint8_t* packet_mask) {
-
   // Number of bit shifts within a byte
   const int num_bit_shifts = (num_column_shift % 8);
   const int num_byte_shifts = num_column_shift >> 3;
@@ -128,7 +135,6 @@ void ShiftFitSubMask(int num_mask_bytes, int res_mask_bytes,
     // For the first byte in the row (j=0 case).
     shift_right_curr_byte = sub_mask[pkt_mask_idx2] >> num_bit_shifts;
     packet_mask[pkt_mask_idx] = shift_right_curr_byte;
-
   }
 }
 }  // namespace
@@ -151,7 +157,9 @@ FecMaskType PacketMaskTable::InitMaskType(FecMaskType fec_mask_type,
   assert(num_media_packets <= static_cast<int>(sizeof(kPacketMaskRandomTbl) /
                                                sizeof(*kPacketMaskRandomTbl)));
   switch (fec_mask_type) {
-    case kFecMaskRandom: { return kFecMaskRandom; }
+    case kFecMaskRandom: {
+      return kFecMaskRandom;
+    }
     case kFecMaskBursty: {
       int max_media_packets = static_cast<int>(sizeof(kPacketMaskBurstyTbl) /
                                                sizeof(*kPacketMaskBurstyTbl));
@@ -170,17 +178,24 @@ FecMaskType PacketMaskTable::InitMaskType(FecMaskType fec_mask_type,
 // |fec_mask_type|.
 const uint8_t*** PacketMaskTable::InitMaskTable(FecMaskType fec_mask_type) {
   switch (fec_mask_type) {
-    case kFecMaskRandom: { return kPacketMaskRandomTbl; }
-    case kFecMaskBursty: { return kPacketMaskBurstyTbl; }
+    case kFecMaskRandom: {
+      return kPacketMaskRandomTbl;
+    }
+    case kFecMaskBursty: {
+      return kPacketMaskBurstyTbl;
+    }
   }
   assert(false);
   return kPacketMaskRandomTbl;
 }
 
 // Remaining protection after important (first partition) packet protection
-void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
-                               int num_fec_for_imp_packets, int num_mask_bytes,
-                               ProtectionMode mode, uint8_t* packet_mask,
+void RemainingPacketProtection(int num_media_packets,
+                               int num_fec_remaining,
+                               int num_fec_for_imp_packets,
+                               int num_mask_bytes,
+                               ProtectionMode mode,
+                               uint8_t* packet_mask,
                                const PacketMaskTable& mask_table) {
   if (mode == kModeNoOverlap) {
     // sub_mask21
@@ -191,8 +206,10 @@ void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
     const int res_mask_bytes =
         (l_bit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
 
-    const uint8_t* packet_mask_sub_21 = mask_table.fec_packet_mask_table()[
-        num_media_packets - num_fec_for_imp_packets - 1][num_fec_remaining - 1];
+    const uint8_t* packet_mask_sub_21 =
+        mask_table.fec_packet_mask_table()[num_media_packets -
+                                           num_fec_for_imp_packets -
+                                           1][num_fec_remaining - 1];
 
     ShiftFitSubMask(num_mask_bytes, res_mask_bytes, num_fec_for_imp_packets,
                     (num_fec_for_imp_packets + num_fec_remaining),
@@ -201,8 +218,9 @@ void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
   } else if (mode == kModeOverlap || mode == kModeBiasFirstPacket) {
     // sub_mask22
 
-    const uint8_t* packet_mask_sub_22 = mask_table
-        .fec_packet_mask_table()[num_media_packets - 1][num_fec_remaining - 1];
+    const uint8_t* packet_mask_sub_22 =
+        mask_table.fec_packet_mask_table()[num_media_packets -
+                                           1][num_fec_remaining - 1];
 
     FitSubMask(num_mask_bytes, num_mask_bytes, num_fec_remaining,
                packet_mask_sub_22,
@@ -217,41 +235,42 @@ void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
   } else {
     assert(false);
   }
-
 }
 
 // Protection for important (first partition) packets
-void ImportantPacketProtection(int num_fec_for_imp_packets, int num_imp_packets,
-                               int num_mask_bytes, uint8_t* packet_mask,
+void ImportantPacketProtection(int num_fec_for_imp_packets,
+                               int num_imp_packets,
+                               int num_mask_bytes,
+                               uint8_t* packet_mask,
                                const PacketMaskTable& mask_table) {
   const int l_bit = num_imp_packets > 16 ? 1 : 0;
   const int num_imp_mask_bytes =
       (l_bit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
 
   // Get sub_mask1 from table
-  const uint8_t* packet_mask_sub_1 = mask_table.fec_packet_mask_table()[
-      num_imp_packets - 1][num_fec_for_imp_packets - 1];
+  const uint8_t* packet_mask_sub_1 =
+      mask_table.fec_packet_mask_table()[num_imp_packets -
+                                         1][num_fec_for_imp_packets - 1];
 
   FitSubMask(num_mask_bytes, num_imp_mask_bytes, num_fec_for_imp_packets,
              packet_mask_sub_1, packet_mask);
-
 }
 
 // This function sets the protection allocation: i.e., how many FEC packets
 // to use for num_imp (1st partition) packets, given the: number of media
 // packets, number of FEC packets, and number of 1st partition packets.
-int SetProtectionAllocation(int num_media_packets, int num_fec_packets,
+int SetProtectionAllocation(int num_media_packets,
+                            int num_fec_packets,
                             int num_imp_packets) {
-
-  // TODO (marpan): test different cases for protection allocation:
+  // TODO(marpan): test different cases for protection allocation:
 
   // Use at most (alloc_par * num_fec_packets) for important packets.
   float alloc_par = 0.5;
   int max_num_fec_for_imp = alloc_par * num_fec_packets;
 
-  int num_fec_for_imp_packets =
-      (num_imp_packets < max_num_fec_for_imp) ? num_imp_packets
-                                              : max_num_fec_for_imp;
+  int num_fec_for_imp_packets = (num_imp_packets < max_num_fec_for_imp)
+                                    ? num_imp_packets
+                                    : max_num_fec_for_imp;
 
   // Fall back to equal protection in this case
   if (num_fec_packets == 1 && (num_media_packets > 2 * num_imp_packets)) {
@@ -268,7 +287,7 @@ int SetProtectionAllocation(int num_media_packets, int num_fec_packets,
 // Current version has 3 modes (options) to build UEP mask from existing ones.
 // Various other combinations may be added in future versions.
 // Longer-term, we may add another set of tables specifically for UEP cases.
-// TODO (marpan): also consider modification of masks for bursty loss cases.
+// TODO(marpan): also consider modification of masks for bursty loss cases.
 
 // Mask is characterized as (#packets_to_protect, #fec_for_protection).
 // Protection factor defined as: (#fec_for_protection / #packets_to_protect).
@@ -306,13 +325,14 @@ int SetProtectionAllocation(int num_media_packets, int num_fec_packets,
 // Protection Mode 2 may be extended for a sort of sliding protection
 // (i.e., vary the number/density of "1s" across columns) across packets.
 
-void UnequalProtectionMask(int num_media_packets, int num_fec_packets,
-                           int num_imp_packets, int num_mask_bytes,
+void UnequalProtectionMask(int num_media_packets,
+                           int num_fec_packets,
+                           int num_imp_packets,
+                           int num_mask_bytes,
                            uint8_t* packet_mask,
                            const PacketMaskTable& mask_table) {
-
   // Set Protection type and allocation
-  // TODO (marpan): test/update for best mode and some combinations thereof.
+  // TODO(marpan): test/update for best mode and some combinations thereof.
 
   ProtectionMode mode = kModeOverlap;
   int num_fec_for_imp_packets = 0;
@@ -341,11 +361,12 @@ void UnequalProtectionMask(int num_media_packets, int num_fec_packets,
                               num_fec_for_imp_packets, num_mask_bytes, mode,
                               packet_mask, mask_table);
   }
-
 }
 
-void GeneratePacketMasks(int num_media_packets, int num_fec_packets,
-                         int num_imp_packets, bool use_unequal_protection,
+void GeneratePacketMasks(int num_media_packets,
+                         int num_fec_packets,
+                         int num_imp_packets,
+                         bool use_unequal_protection,
                          const PacketMaskTable& mask_table,
                          uint8_t* packet_mask) {
   assert(num_media_packets > 0);
@@ -361,16 +382,15 @@ void GeneratePacketMasks(int num_media_packets, int num_fec_packets,
     // Retrieve corresponding mask table directly:for equal-protection case.
     // Mask = (k,n-k), with protection factor = (n-k)/k,
     // where k = num_media_packets, n=total#packets, (n-k)=num_fec_packets.
-    memcpy(packet_mask, mask_table.fec_packet_mask_table()[
-                            num_media_packets - 1][num_fec_packets - 1],
+    memcpy(packet_mask,
+           mask_table.fec_packet_mask_table()[num_media_packets -
+                                              1][num_fec_packets - 1],
            num_fec_packets * num_mask_bytes);
-  } else  //UEP case
-      {
+  } else {  // UEP case
     UnequalProtectionMask(num_media_packets, num_fec_packets, num_imp_packets,
                           num_mask_bytes, packet_mask, mask_table);
-
   }  // End of UEP modification
-}  //End of GetPacketMasks
+}  // End of GetPacketMasks
 
 }  // namespace internal
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h b/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h
index 53ef2a61f4..28276afb72 100644
--- a/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h
+++ b/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
 
-#include <stdint.h>
 #include <stddef.h>
+#include <stdint.h>
 
 namespace rtc {
 class BitBuffer;
diff --git a/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc b/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc
index ceadf4cb6b..7a7e3ed293 100644
--- a/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc
@@ -12,6 +12,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/bitbuffer.h"
 
 namespace webrtc {
@@ -121,7 +122,7 @@ TEST(H264SpsParserTest, TestSampleSPSHdLandscape) {
   const uint8_t buffer[] = {0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50, 0x05,
                             0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0, 0x00,
                             0x00, 0x2A, 0xE0, 0xF1, 0x83, 0x19, 0x60};
-  H264SpsParser parser = H264SpsParser(buffer, ARRAY_SIZE(buffer));
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
   EXPECT_TRUE(parser.Parse());
   EXPECT_EQ(1280u, parser.width());
   EXPECT_EQ(720u, parser.height());
@@ -133,7 +134,7 @@ TEST(H264SpsParserTest, TestSampleSPSVgaLandscape) {
   const uint8_t buffer[] = {0x7A, 0x00, 0x1E, 0xBC, 0xD9, 0x40, 0xA0, 0x2F,
                             0xF8, 0x98, 0x40, 0x00, 0x00, 0x03, 0x01, 0x80,
                             0x00, 0x00, 0x56, 0x83, 0xC5, 0x8B, 0x65, 0x80};
-  H264SpsParser parser = H264SpsParser(buffer, ARRAY_SIZE(buffer));
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
   EXPECT_TRUE(parser.Parse());
   EXPECT_EQ(640u, parser.width());
   EXPECT_EQ(360u, parser.height());
@@ -145,7 +146,7 @@ TEST(H264SpsParserTest, TestSampleSPSWeirdResolution) {
   const uint8_t buffer[] = {0x7A, 0x00, 0x0D, 0xBC, 0xD9, 0x43, 0x43, 0x3E,
                             0x5E, 0x10, 0x00, 0x00, 0x03, 0x00, 0x60, 0x00,
                             0x00, 0x15, 0xA0, 0xF1, 0x42, 0x99, 0x60};
-  H264SpsParser parser = H264SpsParser(buffer, ARRAY_SIZE(buffer));
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
   EXPECT_TRUE(parser.Parse());
   EXPECT_EQ(200u, parser.width());
   EXPECT_EQ(400u, parser.height());
@@ -154,7 +155,7 @@ TEST(H264SpsParserTest, TestSampleSPSWeirdResolution) {
 TEST(H264SpsParserTest, TestSyntheticSPSQvgaLandscape) {
   uint8_t buffer[kSpsBufferMaxSize] = {0};
   GenerateFakeSps(320u, 180u, buffer);
-  H264SpsParser parser = H264SpsParser(buffer, ARRAY_SIZE(buffer));
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
   EXPECT_TRUE(parser.Parse());
   EXPECT_EQ(320u, parser.width());
   EXPECT_EQ(180u, parser.height());
@@ -163,7 +164,7 @@ TEST(H264SpsParserTest, TestSyntheticSPSQvgaLandscape) {
 TEST(H264SpsParserTest, TestSyntheticSPSWeirdResolution) {
   uint8_t buffer[kSpsBufferMaxSize] = {0};
   GenerateFakeSps(156u, 122u, buffer);
-  H264SpsParser parser = H264SpsParser(buffer, ARRAY_SIZE(buffer));
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
   EXPECT_TRUE(parser.Parse());
   EXPECT_EQ(156u, parser.width());
   EXPECT_EQ(122u, parser.height());
diff --git a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
index f577cbaad1..011829cc6c 100644
--- a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 namespace webrtc {
 
@@ -23,7 +23,7 @@ class MockRTPPayloadStrategy : public RTPPayloadStrategy {
   MOCK_CONST_METHOD4(PayloadIsCompatible,
                      bool(const RtpUtility::Payload& payload,
                           const uint32_t frequency,
-                          const uint8_t channels,
+                          const size_t channels,
                           const uint32_t rate));
   MOCK_CONST_METHOD2(UpdatePayloadRate,
                      void(RtpUtility::Payload* payload, const uint32_t rate));
@@ -34,10 +34,10 @@ class MockRTPPayloadStrategy : public RTPPayloadStrategy {
       RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                            const int8_t payloadType,
                            const uint32_t frequency,
-                           const uint8_t channels,
+                           const size_t channels,
                            const uint32_t rate));
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 07a3693507..e19c31bfec 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -16,15 +16,15 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/transport.h"
 
-using namespace webrtc;
+namespace webrtc {
 
 const int kVideoNackListSize = 30;
 const uint32_t kTestSsrc = 3456;
@@ -35,8 +35,7 @@ const int kNumFrames = 30;
 const int kPayloadType = 123;
 const int kRtxPayloadType = 98;
 
-class VerifyingRtxReceiver : public NullRtpData
-{
+class VerifyingRtxReceiver : public NullRtpData {
  public:
   VerifyingRtxReceiver() {}
 
@@ -54,7 +53,7 @@ class VerifyingRtxReceiver : public NullRtpData
 
 class TestRtpFeedback : public NullRtpFeedback {
  public:
-  TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
+  explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
   void OnIncomingSSRCChanged(const uint32_t ssrc) override {
@@ -86,9 +85,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
     rtp_receiver_ = receiver;
   }
 
-  void DropEveryNthPacket(int n) {
-    packet_loss_ = n;
-  }
+  void DropEveryNthPacket(int n) { packet_loss_ = n; }
 
   void DropConsecutivePackets(int start, int total) {
     consecutive_drop_start_ = start;
@@ -100,15 +97,13 @@ class RtxLoopBackTransport : public webrtc::Transport {
                size_t len,
                const PacketOptions& options) override {
     count_++;
-    const unsigned char* ptr = static_cast<const unsigned  char*>(data);
+    const unsigned char* ptr = static_cast<const unsigned char*>(data);
     uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
-    if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
+    if (ssrc == rtx_ssrc_)
+      count_rtx_ssrc_++;
     uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
     size_t packet_length = len;
-    // TODO(pbos): Figure out why this needs to be initialized. Likely this
-    // is hiding a bug either in test setup or other code.
-    // https://code.google.com/p/webrtc/issues/detail?id=3183
-    uint8_t restored_packet[1500] = {0};
+    uint8_t restored_packet[1500];
     RTPHeader header;
     rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     if (!parser->Parse(ptr, len, &header)) {
@@ -136,21 +131,19 @@ class RtxLoopBackTransport : public webrtc::Transport {
       if (!parser->Parse(restored_packet, packet_length, &header)) {
         return false;
       }
+      ptr = restored_packet;
     } else {
       rtp_payload_registry_->SetIncomingPayloadType(header);
     }
 
-    const uint8_t* restored_packet_payload =
-        restored_packet + header.headerLength;
-    packet_length -= header.headerLength;
     PayloadUnion payload_specific;
     if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                     &payload_specific)) {
       return false;
     }
-    if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_payload,
-                                          packet_length, payload_specific,
-                                          true)) {
+    if (!rtp_receiver_->IncomingRtpPacket(header, ptr + header.headerLength,
+                                          packet_length - header.headerLength,
+                                          payload_specific, true)) {
       return false;
     }
     return true;
@@ -194,8 +187,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
     rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
 
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
-        &fake_clock, &receiver_, rtp_feedback_.get(),
-        &rtp_payload_registry_));
+        &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_));
 
     rtp_rtcp_module_->SetSSRC(kTestSsrc);
     rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
@@ -215,11 +207,9 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
 
     EXPECT_EQ(0, rtp_rtcp_module_->RegisterSendPayload(video_codec));
     rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
-    EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(video_codec.plName,
-                                                       video_codec.plType,
-                                                       90000,
-                                                       0,
-                                                       video_codec.maxBitrate));
+    EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(
+                     video_codec.plName, video_codec.plType, 90000, 0,
+                     video_codec.maxBitrate));
     rtp_payload_registry_.SetRtxPayloadType(kRtxPayloadType, kPayloadType);
 
     for (size_t n = 0; n < payload_data_length; n++) {
@@ -230,8 +220,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
   int BuildNackList(uint16_t* nack_list) {
     receiver_.sequence_numbers_.sort();
     std::list<uint16_t> missing_sequence_numbers;
-    std::list<uint16_t>::iterator it =
-        receiver_.sequence_numbers_.begin();
+    std::list<uint16_t>::iterator it = receiver_.sequence_numbers_.begin();
 
     while (it != receiver_.sequence_numbers_.end()) {
       uint16_t sequence_number_1 = *it;
@@ -239,15 +228,14 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
       if (it != receiver_.sequence_numbers_.end()) {
         uint16_t sequence_number_2 = *it;
         // Add all missing sequence numbers to list
-        for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2;
-            ++i) {
+        for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
           missing_sequence_numbers.push_back(i);
         }
       }
     }
     int n = 0;
     for (it = missing_sequence_numbers.begin();
-        it != missing_sequence_numbers.end(); ++it) {
+         it != missing_sequence_numbers.end(); ++it) {
       nack_list[n++] = (*it);
     }
     return n;
@@ -298,7 +286,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
   rtc::scoped_ptr<TestRtpFeedback> rtp_feedback_;
   RtxLoopBackTransport transport_;
   VerifyingRtxReceiver receiver_;
-  uint8_t  payload_data[65000];
+  uint8_t payload_data[65000];
   size_t payload_data_length;
   SimulatedClock fake_clock;
 };
@@ -345,8 +333,10 @@ TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
-      *(receiver_.sequence_numbers_.rbegin()));
+            *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
+
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
index 683b951f1e..be4b453454 100644
--- a/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
@@ -9,8 +9,10 @@
  */
 
 #include <list>
+#include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
@@ -54,6 +56,53 @@ class ProducerFecTest : public ::testing::Test {
   FrameGenerator* generator_;
 };
 
+// Verifies bug found via fuzzing, where a gap in the packet sequence caused us
+// to move past the end of the current FEC packet mask byte without moving to
+// the next byte. That likely caused us to repeatedly read from the same byte,
+// and if that byte didn't protect packets we would generate empty FEC.
+TEST_F(ProducerFecTest, NoEmptyFecWithSeqNumGaps) {
+  struct Packet {
+    size_t header_size;
+    size_t payload_size;
+    uint16_t seq_num;
+    bool marker_bit;
+  };
+  std::vector<Packet> protected_packets;
+  protected_packets.push_back({15, 3, 41, 0});
+  protected_packets.push_back({14, 1, 43, 0});
+  protected_packets.push_back({19, 0, 48, 0});
+  protected_packets.push_back({19, 0, 50, 0});
+  protected_packets.push_back({14, 3, 51, 0});
+  protected_packets.push_back({13, 8, 52, 0});
+  protected_packets.push_back({19, 2, 53, 0});
+  protected_packets.push_back({12, 3, 54, 0});
+  protected_packets.push_back({21, 0, 55, 0});
+  protected_packets.push_back({13, 3, 57, 1});
+  FecProtectionParams params = {117, 0, 3, kFecMaskBursty};
+  producer_->SetFecParameters(&params, 0);
+  uint8_t packet[28] = {0};
+  for (Packet p : protected_packets) {
+    if (p.marker_bit) {
+      packet[1] |= 0x80;
+    } else {
+      packet[1] &= ~0x80;
+    }
+    ByteWriter<uint16_t>::WriteBigEndian(&packet[2], p.seq_num);
+    producer_->AddRtpPacketAndGenerateFec(packet, p.payload_size,
+                                          p.header_size);
+    uint16_t num_fec_packets = producer_->NumAvailableFecPackets();
+    std::vector<RedPacket*> fec_packets;
+    if (num_fec_packets > 0) {
+      fec_packets =
+          producer_->GetFecPackets(kRedPayloadType, 99, 100, p.header_size);
+      EXPECT_EQ(num_fec_packets, fec_packets.size());
+    }
+    for (RedPacket* fec_packet : fec_packets) {
+      delete fec_packet;
+    }
+  }
+}
+
 TEST_F(ProducerFecTest, OneFrameFec) {
   // The number of media packets (|kNumPackets|), number of frames (one for
   // this test), and the protection factor (|params->fec_rate|) are set to make
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 8ac7e0a383..24f1e2c96e 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -14,7 +14,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
@@ -37,8 +37,6 @@ StreamStatisticianImpl::StreamStatisticianImpl(
       cumulative_loss_(0),
       jitter_q4_transmission_time_offset_(0),
       last_receive_time_ms_(0),
-      last_receive_time_secs_(0),
-      last_receive_time_frac_(0),
       last_received_timestamp_(0),
       last_received_transmission_time_offset_(0),
       received_seq_first_(0),
@@ -79,9 +77,7 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
   // are received, 4 will be ignored.
   if (in_order) {
     // Current time in samples.
-    uint32_t receive_time_secs;
-    uint32_t receive_time_frac;
-    clock_->CurrentNtp(receive_time_secs, receive_time_frac);
+    NtpTime receive_time(*clock_);
 
     // Wrong if we use RetransmitOfOldPacket.
     if (receive_counters_.transmitted.packets > 1 &&
@@ -97,11 +93,10 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
     if (header.timestamp != last_received_timestamp_ &&
         (receive_counters_.transmitted.packets -
          receive_counters_.retransmitted.packets) > 1) {
-      UpdateJitter(header, receive_time_secs, receive_time_frac);
+      UpdateJitter(header, receive_time);
     }
     last_received_timestamp_ = header.timestamp;
-    last_receive_time_secs_ = receive_time_secs;
-    last_receive_time_frac_ = receive_time_frac;
+    last_receive_time_ntp_ = receive_time;
     last_receive_time_ms_ = clock_->TimeInMilliseconds();
   }
 
@@ -113,14 +108,11 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
 }
 
 void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
-                                          uint32_t receive_time_secs,
-                                          uint32_t receive_time_frac) {
-  uint32_t receive_time_rtp = RtpUtility::ConvertNTPTimeToRTP(
-      receive_time_secs, receive_time_frac, header.payload_type_frequency);
+                                          NtpTime receive_time) {
+  uint32_t receive_time_rtp =
+      NtpToRtp(receive_time, header.payload_type_frequency);
   uint32_t last_receive_time_rtp =
-      RtpUtility::ConvertNTPTimeToRTP(last_receive_time_secs_,
-                                      last_receive_time_frac_,
-                                      header.payload_type_frequency);
+      NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
   int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
       (header.timestamp - last_received_timestamp_);
 
@@ -267,6 +259,7 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
   stats.fraction_lost = local_fraction_lost;
 
   // We need a counter for cumulative loss too.
+  // TODO(danilchap): Ensure cumulative loss is below maximum value of 2^24.
   cumulative_loss_ += missing;
   stats.cumulative_lost = cumulative_loss_;
   stats.extended_max_sequence_number =
@@ -319,8 +312,8 @@ void StreamStatisticianImpl::ProcessBitrate() {
 void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
                                                 uint32_t* frac) const {
   CriticalSectionScoped cs(stream_lock_.get());
-  *secs = last_receive_time_secs_;
-  *frac = last_receive_time_frac_;
+  *secs = last_receive_time_ntp_.seconds();
+  *frac = last_receive_time_ntp_.fractions();
 }
 
 bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index fe42990fe9..025dcd42c7 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -11,13 +11,15 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 #include <algorithm>
+#include <map>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
 
 namespace webrtc {
 
@@ -51,9 +53,7 @@ class StreamStatisticianImpl : public StreamStatistician {
  private:
   bool InOrderPacketInternal(uint16_t sequence_number) const;
   RtcpStatistics CalculateRtcpStatistics();
-  void UpdateJitter(const RTPHeader& header,
-                    uint32_t receive_time_secs,
-                    uint32_t receive_time_frac);
+  void UpdateJitter(const RTPHeader& header, NtpTime receive_time);
   void UpdateCounters(const RTPHeader& rtp_header,
                       size_t packet_length,
                       bool retransmitted);
@@ -72,8 +72,7 @@ class StreamStatisticianImpl : public StreamStatistician {
   uint32_t jitter_q4_transmission_time_offset_;
 
   int64_t last_receive_time_ms_;
-  uint32_t last_receive_time_secs_;
-  uint32_t last_receive_time_frac_;
+  NtpTime last_receive_time_ntp_;
   uint32_t last_received_timestamp_;
   int32_t last_received_transmission_time_offset_;
   uint16_t received_seq_first_;
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 5b522281df..c265c17c04 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -11,7 +11,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
index 74fc9cdc56..ccc15ec417 100644
--- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
+++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/system_wrappers/include/clock.h"
diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
index bc9cf2ee39..797c7883a9 100644
--- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
@@ -11,7 +11,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 using ::testing::_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 7a7645fd1b..87c0259b3e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -21,16 +21,13 @@
 #include "webrtc/test/null_transport.h"
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
 namespace {
 
-using namespace webrtc;
-
-
 class TestTransport : public Transport {
  public:
-  TestTransport(RTCPReceiver* rtcp_receiver) :
-    rtcp_receiver_(rtcp_receiver) {
-  }
+  explicit TestTransport(RTCPReceiver* rtcp_receiver)
+      : rtcp_receiver_(rtcp_receiver) {}
 
   bool SendRtp(const uint8_t* /*data*/,
                size_t /*len*/,
@@ -38,9 +35,8 @@ class TestTransport : public Transport {
     return false;
   }
   bool SendRtcp(const uint8_t* packet, size_t packetLength) override {
-    RTCPUtility::RTCPParserV2 rtcpParser((uint8_t*)packet,
-                                         packetLength,
-                                         true); // Allow non-compound RTCP
+    RTCPUtility::RTCPParserV2 rtcpParser(packet, packetLength,
+                                         true);  // Allow non-compound RTCP
 
     EXPECT_TRUE(rtcpParser.IsValid());
     RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
@@ -53,11 +49,11 @@ class TestTransport : public Transport {
               rtcpPacketInformation.receiverEstimatedMaxBitrate);
     return true;
   }
+
  private:
   RTCPReceiver* rtcp_receiver_;
 };
 
-
 class RtcpFormatRembTest : public ::testing::Test {
  protected:
   RtcpFormatRembTest()
@@ -134,3 +130,4 @@ TEST_F(RtcpFormatRembTest, TestCompund) {
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRemb));
 }
 }  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
index 792caa7b8b..eef2978371 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
+#include <algorithm>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
@@ -19,38 +21,25 @@ using webrtc::RTCPUtility::kBtReceiverReferenceTime;
 using webrtc::RTCPUtility::kBtVoipMetric;
 
 using webrtc::RTCPUtility::PT_APP;
-using webrtc::RTCPUtility::PT_BYE;
 using webrtc::RTCPUtility::PT_IJ;
 using webrtc::RTCPUtility::PT_PSFB;
-using webrtc::RTCPUtility::PT_RR;
 using webrtc::RTCPUtility::PT_RTPFB;
 using webrtc::RTCPUtility::PT_SDES;
 using webrtc::RTCPUtility::PT_SR;
 using webrtc::RTCPUtility::PT_XR;
 
 using webrtc::RTCPUtility::RTCPPacketAPP;
-using webrtc::RTCPUtility::RTCPPacketBYE;
 using webrtc::RTCPUtility::RTCPPacketPSFBAPP;
 using webrtc::RTCPUtility::RTCPPacketPSFBFIR;
 using webrtc::RTCPUtility::RTCPPacketPSFBFIRItem;
-using webrtc::RTCPUtility::RTCPPacketPSFBPLI;
 using webrtc::RTCPUtility::RTCPPacketPSFBREMBItem;
 using webrtc::RTCPUtility::RTCPPacketPSFBRPSI;
-using webrtc::RTCPUtility::RTCPPacketPSFBSLI;
-using webrtc::RTCPUtility::RTCPPacketPSFBSLIItem;
 using webrtc::RTCPUtility::RTCPPacketReportBlockItem;
-using webrtc::RTCPUtility::RTCPPacketRR;
 using webrtc::RTCPUtility::RTCPPacketRTPFBNACK;
 using webrtc::RTCPUtility::RTCPPacketRTPFBNACKItem;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBN;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBNItem;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBR;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBRItem;
 using webrtc::RTCPUtility::RTCPPacketSR;
 using webrtc::RTCPUtility::RTCPPacketXRDLRRReportBlockItem;
-using webrtc::RTCPUtility::RTCPPacketXRReceiverReferenceTimeItem;
 using webrtc::RTCPUtility::RTCPPacketXR;
-using webrtc::RTCPUtility::RTCPPacketXRVOIPMetricItem;
 
 namespace webrtc {
 namespace rtcp {
@@ -122,21 +111,6 @@ void CreateSenderReport(const RTCPPacketSR& sr,
   AssignUWord32(buffer, pos, sr.SenderOctetCount);
 }
 
-//  Receiver report (RR), header (RFC 3550).
-//
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P|    RC   |   PT=RR=201   |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                     SSRC of packet sender                     |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-
-void CreateReceiverReport(const RTCPPacketRR& rr,
-                          uint8_t* buffer,
-                          size_t* pos) {
-  AssignUWord32(buffer, pos, rr.SenderSSRC);
-}
-
 //  Report block (RFC 3550).
 //
 //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ -154,43 +128,15 @@ void CreateReceiverReport(const RTCPPacketRR& rr,
 //  |                   delay since last SR (DLSR)                  |
 //  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
 
-void CreateReportBlocks(const std::vector<RTCPPacketReportBlockItem>& blocks,
+void CreateReportBlocks(const std::vector<ReportBlock>& blocks,
                         uint8_t* buffer,
                         size_t* pos) {
-  for (std::vector<RTCPPacketReportBlockItem>::const_iterator
-       it = blocks.begin(); it != blocks.end(); ++it) {
-    AssignUWord32(buffer, pos, (*it).SSRC);
-    AssignUWord8(buffer, pos, (*it).FractionLost);
-    AssignUWord24(buffer, pos, (*it).CumulativeNumOfPacketsLost);
-    AssignUWord32(buffer, pos, (*it).ExtendedHighestSequenceNumber);
-    AssignUWord32(buffer, pos, (*it).Jitter);
-    AssignUWord32(buffer, pos, (*it).LastSR);
-    AssignUWord32(buffer, pos, (*it).DelayLastSR);
+  for (const ReportBlock& block : blocks) {
+    block.Create(buffer + *pos);
+    *pos += ReportBlock::kLength;
   }
 }
 
-// Transmission Time Offsets in RTP Streams (RFC 5450).
-//
-//      0                   1                   2                   3
-//      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     .                                                               .
-//     .                                                               .
-//     .                                                               .
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateIj(const std::vector<uint32_t>& ij_items,
-              uint8_t* buffer,
-              size_t* pos) {
-  for (uint32_t item : ij_items)
-    AssignUWord32(buffer, pos, item);
-}
-
 // Source Description (SDES) (RFC 3550).
 //
 //         0                   1                   2                   3
@@ -233,129 +179,6 @@ void CreateSdes(const std::vector<Sdes::Chunk>& chunks,
   }
 }
 
-// Bye packet (BYE) (RFC 3550).
-//
-//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |                           SSRC/CSRC                           |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       :                              ...                              :
-//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-// (opt) |     length    |               reason for leaving            ...
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateBye(const RTCPPacketBYE& bye,
-               const std::vector<uint32_t>& csrcs,
-               uint8_t* buffer,
-               size_t* pos) {
-  AssignUWord32(buffer, pos, bye.SenderSSRC);
-  for (uint32_t csrc : csrcs)
-    AssignUWord32(buffer, pos, csrc);
-}
-
-// Application-Defined packet (APP) (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P| subtype |   PT=APP=204  |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                           SSRC/CSRC                           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                          name (ASCII)                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   application-dependent data                ...
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateApp(const RTCPPacketAPP& app,
-               uint32_t ssrc,
-               uint8_t* buffer,
-               size_t* pos) {
-  AssignUWord32(buffer, pos, ssrc);
-  AssignUWord32(buffer, pos, app.Name);
-  memcpy(buffer + *pos, app.Data, app.Size);
-  *pos += app.Size;
-}
-
-// RFC 4585: Feedback format.
-//
-// Common packet format:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |V=2|P|   FMT   |       PT      |          length               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of packet sender                        |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of media source                         |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   :            Feedback Control Information (FCI)                 :
-//   :
-//
-
-// Picture loss indication (PLI) (RFC 4585).
-//
-// FCI: no feedback control information.
-
-void CreatePli(const RTCPPacketPSFBPLI& pli,
-               uint8_t* buffer,
-               size_t* pos) {
-  AssignUWord32(buffer, pos, pli.SenderSSRC);
-  AssignUWord32(buffer, pos, pli.MediaSSRC);
-}
-
-// Slice loss indication (SLI) (RFC 4585).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            First        |        Number           | PictureID |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateSli(const RTCPPacketPSFBSLI& sli,
-               const RTCPPacketPSFBSLIItem& sli_item,
-               uint8_t* buffer,
-               size_t* pos) {
-  AssignUWord32(buffer, pos, sli.SenderSSRC);
-  AssignUWord32(buffer, pos, sli.MediaSSRC);
-
-  AssignUWord8(buffer, pos, sli_item.FirstMB >> 5);
-  AssignUWord8(buffer, pos, (sli_item.FirstMB << 3) +
-                            ((sli_item.NumberOfMB >> 10) & 0x07));
-  AssignUWord8(buffer, pos, sli_item.NumberOfMB >> 2);
-  AssignUWord8(buffer, pos, (sli_item.NumberOfMB << 6) + sli_item.PictureId);
-}
-
-// Generic NACK (RFC 4585).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            PID                |             BLP               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateNack(const RTCPPacketRTPFBNACK& nack,
-                const std::vector<RTCPPacketRTPFBNACKItem>& nack_fields,
-                size_t start_index,
-                size_t end_index,
-                uint8_t* buffer,
-                size_t* pos) {
-  AssignUWord32(buffer, pos, nack.SenderSSRC);
-  AssignUWord32(buffer, pos, nack.MediaSSRC);
-  for (size_t i = start_index; i < end_index; ++i) {
-    const RTCPPacketRTPFBNACKItem& nack_item = nack_fields[i];
-    AssignUWord16(buffer, pos, nack_item.PacketID);
-    AssignUWord16(buffer, pos, nack_item.BitMask);
-  }
-}
-
 // Reference picture selection indication (RPSI) (RFC 4585).
 //
 // FCI:
@@ -407,66 +230,6 @@ void CreateFir(const RTCPPacketPSFBFIR& fir,
   AssignUWord24(buffer, pos, 0);
 }
 
-void CreateTmmbrItem(const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
-                     uint8_t* buffer,
-                     size_t* pos) {
-  uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000;
-  uint32_t mantissa = 0;
-  uint8_t exp = 0;
-  ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
-
-  AssignUWord32(buffer, pos, tmmbr_item.SSRC);
-  AssignUWord8(buffer, pos, (exp << 2) + ((mantissa >> 15) & 0x03));
-  AssignUWord8(buffer, pos, mantissa >> 7);
-  AssignUWord8(buffer, pos, (mantissa << 1) +
-                            ((tmmbr_item.MeasuredOverhead >> 8) & 0x01));
-  AssignUWord8(buffer, pos, tmmbr_item.MeasuredOverhead);
-}
-
-// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateTmmbr(const RTCPPacketRTPFBTMMBR& tmmbr,
-                 const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
-                 uint8_t* buffer,
-                 size_t* pos) {
-  AssignUWord32(buffer, pos, tmmbr.SenderSSRC);
-  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
-  CreateTmmbrItem(tmmbr_item, buffer, pos);
-}
-
-// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateTmmbn(const RTCPPacketRTPFBTMMBN& tmmbn,
-                 const std::vector<RTCPPacketRTPFBTMMBRItem>& tmmbn_items,
-                 uint8_t* buffer,
-                 size_t* pos) {
-  AssignUWord32(buffer, pos, tmmbn.SenderSSRC);
-  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
-  for (uint8_t i = 0; i < tmmbn_items.size(); ++i) {
-    CreateTmmbrItem(tmmbn_items[i], buffer, pos);
-  }
-}
-
 // Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
 //
 //    0                   1                   2                   3
@@ -529,130 +292,6 @@ void CreateXrHeader(const RTCPPacketXR& header,
   AssignUWord32(buffer, pos, header.OriginatorSSRC);
 }
 
-void CreateXrBlockHeader(uint8_t block_type,
-                         uint16_t block_length,
-                         uint8_t* buffer,
-                         size_t* pos) {
-  AssignUWord8(buffer, pos, block_type);
-  AssignUWord8(buffer, pos, 0);
-  AssignUWord16(buffer, pos, block_length);
-}
-
-// Receiver Reference Time Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=4      |   reserved    |       block length = 2        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |              NTP timestamp, most significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |             NTP timestamp, least significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateRrtr(const std::vector<RTCPPacketXRReceiverReferenceTimeItem>& rrtrs,
-                uint8_t* buffer,
-                size_t* pos) {
-  const uint16_t kBlockLength = 2;
-  for (std::vector<RTCPPacketXRReceiverReferenceTimeItem>::const_iterator it =
-       rrtrs.begin(); it != rrtrs.end(); ++it) {
-    CreateXrBlockHeader(kBtReceiverReferenceTime, kBlockLength, buffer, pos);
-    AssignUWord32(buffer, pos, (*it).NTPMostSignificant);
-    AssignUWord32(buffer, pos, (*it).NTPLeastSignificant);
-  }
-}
-
-// DLRR Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=5      |   reserved    |         block length          |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_1 (SSRC of first receiver)               | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  |                         last RR (LRR)                         |   1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   delay since last RR (DLRR)                  |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_2 (SSRC of second receiver)              | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  :                               ...                             :   2
-
-void CreateDlrr(const std::vector<Xr::DlrrBlock>& dlrrs,
-                uint8_t* buffer,
-                size_t* pos) {
-  for (std::vector<Xr::DlrrBlock>::const_iterator it = dlrrs.begin();
-       it != dlrrs.end(); ++it) {
-    if ((*it).empty()) {
-      continue;
-    }
-    uint16_t block_length = 3 * (*it).size();
-    CreateXrBlockHeader(kBtDlrr, block_length, buffer, pos);
-    for (Xr::DlrrBlock::const_iterator it_block = (*it).begin();
-         it_block != (*it).end(); ++it_block) {
-      AssignUWord32(buffer, pos, (*it_block).SSRC);
-      AssignUWord32(buffer, pos, (*it_block).LastRR);
-      AssignUWord32(buffer, pos, (*it_block).DelayLastRR);
-    }
-  }
-}
-
-// VoIP Metrics Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=7      |   reserved    |       block length = 8        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                        SSRC of source                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   loss rate   | discard rate  | burst density |  gap density  |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |       burst duration          |         gap duration          |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     round trip delay          |       end system delay        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  | signal level  |  noise level  |     RERL      |     Gmin      |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   RX config   |   reserved    |          JB nominal           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |          JB maximum           |          JB abs max           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateVoipMetric(const std::vector<RTCPPacketXRVOIPMetricItem>& metrics,
-                      uint8_t* buffer,
-                      size_t* pos) {
-  const uint16_t kBlockLength = 8;
-  for (std::vector<RTCPPacketXRVOIPMetricItem>::const_iterator it =
-       metrics.begin(); it != metrics.end(); ++it) {
-    CreateXrBlockHeader(kBtVoipMetric, kBlockLength, buffer, pos);
-    AssignUWord32(buffer, pos, (*it).SSRC);
-    AssignUWord8(buffer, pos, (*it).lossRate);
-    AssignUWord8(buffer, pos, (*it).discardRate);
-    AssignUWord8(buffer, pos, (*it).burstDensity);
-    AssignUWord8(buffer, pos, (*it).gapDensity);
-    AssignUWord16(buffer, pos, (*it).burstDuration);
-    AssignUWord16(buffer, pos, (*it).gapDuration);
-    AssignUWord16(buffer, pos, (*it).roundTripDelay);
-    AssignUWord16(buffer, pos, (*it).endSystemDelay);
-    AssignUWord8(buffer, pos, (*it).signalLevel);
-    AssignUWord8(buffer, pos, (*it).noiseLevel);
-    AssignUWord8(buffer, pos, (*it).RERL);
-    AssignUWord8(buffer, pos, (*it).Gmin);
-    AssignUWord8(buffer, pos, (*it).Rfactor);
-    AssignUWord8(buffer, pos, (*it).extRfactor);
-    AssignUWord8(buffer, pos, (*it).MOSLQ);
-    AssignUWord8(buffer, pos, (*it).MOSCQ);
-    AssignUWord8(buffer, pos, (*it).RXconfig);
-    AssignUWord8(buffer, pos, 0);
-    AssignUWord16(buffer, pos, (*it).JBnominal);
-    AssignUWord16(buffer, pos, (*it).JBmax);
-    AssignUWord16(buffer, pos, (*it).JBabsMax);
-  }
-}
 }  // namespace
 
 void RtcpPacket::Append(RtcpPacket* packet) {
@@ -751,17 +390,6 @@ void RtcpPacket::CreateHeader(
   AssignUWord16(buffer, pos, length);
 }
 
-bool Empty::Create(uint8_t* packet,
-                   size_t* index,
-                   size_t max_length,
-                   RtcpPacket::PacketReadyCallback* callback) const {
-  return true;
-}
-
-size_t Empty::BlockLength() const {
-  return 0;
-}
-
 bool SenderReport::Create(uint8_t* packet,
                           size_t* index,
                           size_t max_length,
@@ -781,58 +409,11 @@ bool SenderReport::WithReportBlock(const ReportBlock& block) {
     LOG(LS_WARNING) << "Max report blocks reached.";
     return false;
   }
-  report_blocks_.push_back(block.report_block_);
+  report_blocks_.push_back(block);
   sr_.NumberOfReportBlocks = report_blocks_.size();
   return true;
 }
 
-bool ReceiverReport::Create(uint8_t* packet,
-                            size_t* index,
-                            size_t max_length,
-                            RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  CreateHeader(rr_.NumberOfReportBlocks, PT_RR, HeaderLength(), packet, index);
-  CreateReceiverReport(rr_, packet, index);
-  CreateReportBlocks(report_blocks_, packet, index);
-  return true;
-}
-
-bool ReceiverReport::WithReportBlock(const ReportBlock& block) {
-  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
-    LOG(LS_WARNING) << "Max report blocks reached.";
-    return false;
-  }
-  report_blocks_.push_back(block.report_block_);
-  rr_.NumberOfReportBlocks = report_blocks_.size();
-  return true;
-}
-
-bool Ij::Create(uint8_t* packet,
-                size_t* index,
-                size_t max_length,
-                RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  size_t length = ij_items_.size();
-  CreateHeader(length, PT_IJ, length, packet, index);
-  CreateIj(ij_items_, packet, index);
-  return true;
-}
-
-bool Ij::WithJitterItem(uint32_t jitter) {
-  if (ij_items_.size() >= kMaxNumberOfIjItems) {
-    LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
-    return false;
-  }
-  ij_items_.push_back(jitter);
-  return true;
-}
-
 bool Sdes::Create(uint8_t* packet,
                   size_t* index,
                   size_t max_length,
@@ -876,129 +457,6 @@ size_t Sdes::BlockLength() const {
   return length;
 }
 
-bool Bye::Create(uint8_t* packet,
-                 size_t* index,
-                 size_t max_length,
-                 RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  size_t length = HeaderLength();
-  CreateHeader(length, PT_BYE, length, packet, index);
-  CreateBye(bye_, csrcs_, packet, index);
-  return true;
-}
-
-bool Bye::WithCsrc(uint32_t csrc) {
-  if (csrcs_.size() >= kMaxNumberOfCsrcs) {
-    LOG(LS_WARNING) << "Max CSRC size reached.";
-    return false;
-  }
-  csrcs_.push_back(csrc);
-  return true;
-}
-
-bool App::Create(uint8_t* packet,
-                 size_t* index,
-                 size_t max_length,
-                 RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  CreateHeader(app_.SubType, PT_APP, HeaderLength(), packet, index);
-  CreateApp(app_, ssrc_, packet, index);
-  return true;
-}
-
-bool Pli::Create(uint8_t* packet,
-                 size_t* index,
-                 size_t max_length,
-                 RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  const uint8_t kFmt = 1;
-  CreateHeader(kFmt, PT_PSFB, HeaderLength(), packet, index);
-  CreatePli(pli_, packet, index);
-  return true;
-}
-
-bool Sli::Create(uint8_t* packet,
-                 size_t* index,
-                 size_t max_length,
-                 RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  const uint8_t kFmt = 2;
-  CreateHeader(kFmt, PT_PSFB, HeaderLength(), packet, index);
-  CreateSli(sli_, sli_item_, packet, index);
-  return true;
-}
-
-bool Nack::Create(uint8_t* packet,
-                  size_t* index,
-                  size_t max_length,
-                  RtcpPacket::PacketReadyCallback* callback) const {
-  assert(!nack_fields_.empty());
-  // If nack list can't fit in packet, try to fragment.
-  size_t nack_index = 0;
-  do {
-    size_t bytes_left_in_buffer = max_length - *index;
-    if (bytes_left_in_buffer < kCommonFbFmtLength + 4) {
-      if (!OnBufferFull(packet, index, callback))
-        return false;
-      continue;
-    }
-    int64_t num_nack_fields =
-        std::min((bytes_left_in_buffer - kCommonFbFmtLength) / 4,
-                 nack_fields_.size() - nack_index);
-
-    const uint8_t kFmt = 1;
-    size_t size_bytes = (num_nack_fields * 4) + kCommonFbFmtLength;
-    size_t header_length = ((size_bytes + 3) / 4) - 1;  // As 32bit words - 1
-    CreateHeader(kFmt, PT_RTPFB, header_length, packet, index);
-    CreateNack(nack_, nack_fields_, nack_index, nack_index + num_nack_fields,
-               packet, index);
-
-    nack_index += num_nack_fields;
-  } while (nack_index < nack_fields_.size());
-
-  return true;
-}
-
-size_t Nack::BlockLength() const {
-  return (nack_fields_.size() * 4) + kCommonFbFmtLength;
-}
-
-void Nack::WithList(const uint16_t* nack_list, int length) {
-  assert(nack_list);
-  assert(nack_fields_.empty());
-  int i = 0;
-  while (i < length) {
-    uint16_t pid = nack_list[i++];
-    // Bitmask specifies losses in any of the 16 packets following the pid.
-    uint16_t bitmask = 0;
-    while (i < length) {
-      int shift = static_cast<uint16_t>(nack_list[i] - pid) - 1;
-      if (shift >= 0 && shift <= 15) {
-        bitmask |= (1 << shift);
-        ++i;
-      } else {
-        break;
-      }
-    }
-    RTCPUtility::RTCPPacketRTPFBNACKItem item;
-    item.PacketID = pid;
-    item.BitMask = bitmask;
-    nack_fields_.push_back(item);
-  }
-}
-
 bool Rpsi::Create(uint8_t* packet,
                   size_t* index,
                   size_t max_length,
@@ -1077,48 +535,6 @@ void Remb::AppliesTo(uint32_t ssrc) {
   remb_item_.SSRCs[remb_item_.NumberOfSSRCs++] = ssrc;
 }
 
-bool Tmmbr::Create(uint8_t* packet,
-                   size_t* index,
-                   size_t max_length,
-                   RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  const uint8_t kFmt = 3;
-  CreateHeader(kFmt, PT_RTPFB, HeaderLength(), packet, index);
-  CreateTmmbr(tmmbr_, tmmbr_item_, packet, index);
-  return true;
-}
-
-bool Tmmbn::WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead) {
-  assert(overhead <= 0x1ff);
-  if (tmmbn_items_.size() >= kMaxNumberOfTmmbrs) {
-    LOG(LS_WARNING) << "Max TMMBN size reached.";
-    return false;
-  }
-  RTCPPacketRTPFBTMMBRItem tmmbn_item;
-  tmmbn_item.SSRC = ssrc;
-  tmmbn_item.MaxTotalMediaBitRate = bitrate_kbps;
-  tmmbn_item.MeasuredOverhead = overhead;
-  tmmbn_items_.push_back(tmmbn_item);
-  return true;
-}
-
-bool Tmmbn::Create(uint8_t* packet,
-                   size_t* index,
-                   size_t max_length,
-                   RtcpPacket::PacketReadyCallback* callback) const {
-  while (*index + BlockLength() > max_length) {
-    if (!OnBufferFull(packet, index, callback))
-      return false;
-  }
-  const uint8_t kFmt = 4;
-  CreateHeader(kFmt, PT_RTPFB, HeaderLength(), packet, index);
-  CreateTmmbn(tmmbn_, tmmbn_items_, packet, index);
-  return true;
-}
-
 bool Xr::Create(uint8_t* packet,
                 size_t* index,
                 size_t max_length,
@@ -1129,29 +545,38 @@ bool Xr::Create(uint8_t* packet,
   }
   CreateHeader(0U, PT_XR, HeaderLength(), packet, index);
   CreateXrHeader(xr_header_, packet, index);
-  CreateRrtr(rrtr_blocks_, packet, index);
-  CreateDlrr(dlrr_blocks_, packet, index);
-  CreateVoipMetric(voip_metric_blocks_, packet, index);
+  for (const Rrtr& block : rrtr_blocks_) {
+    block.Create(packet + *index);
+    *index += Rrtr::kLength;
+  }
+  for (const Dlrr& block : dlrr_blocks_) {
+    block.Create(packet + *index);
+    *index += block.BlockLength();
+  }
+  for (const VoipMetric& block : voip_metric_blocks_) {
+    block.Create(packet + *index);
+    *index += VoipMetric::kLength;
+  }
   return true;
 }
 
 bool Xr::WithRrtr(Rrtr* rrtr) {
-  assert(rrtr);
+  RTC_DCHECK(rrtr);
   if (rrtr_blocks_.size() >= kMaxNumberOfRrtrBlocks) {
     LOG(LS_WARNING) << "Max RRTR blocks reached.";
     return false;
   }
-  rrtr_blocks_.push_back(rrtr->rrtr_block_);
+  rrtr_blocks_.push_back(*rrtr);
   return true;
 }
 
 bool Xr::WithDlrr(Dlrr* dlrr) {
-  assert(dlrr);
+  RTC_DCHECK(dlrr);
   if (dlrr_blocks_.size() >= kMaxNumberOfDlrrBlocks) {
     LOG(LS_WARNING) << "Max DLRR blocks reached.";
     return false;
   }
-  dlrr_blocks_.push_back(dlrr->dlrr_block_);
+  dlrr_blocks_.push_back(*dlrr);
   return true;
 }
 
@@ -1161,38 +586,18 @@ bool Xr::WithVoipMetric(VoipMetric* voip_metric) {
     LOG(LS_WARNING) << "Max Voip Metric blocks reached.";
     return false;
   }
-  voip_metric_blocks_.push_back(voip_metric->metric_);
+  voip_metric_blocks_.push_back(*voip_metric);
   return true;
 }
 
 size_t Xr::DlrrLength() const {
-  const size_t kBlockHeaderLen = 4;
-  const size_t kSubBlockLen = 12;
   size_t length = 0;
-  for (std::vector<DlrrBlock>::const_iterator it = dlrr_blocks_.begin();
-       it != dlrr_blocks_.end(); ++it) {
-    if (!(*it).empty()) {
-      length += kBlockHeaderLen + kSubBlockLen * (*it).size();
-    }
+  for (const Dlrr& block : dlrr_blocks_) {
+    length += block.BlockLength();
   }
   return length;
 }
 
-bool Dlrr::WithDlrrItem(uint32_t ssrc,
-                        uint32_t last_rr,
-                        uint32_t delay_last_rr) {
-  if (dlrr_block_.size() >= kMaxNumberOfDlrrItems) {
-    LOG(LS_WARNING) << "Max DLRR items reached.";
-    return false;
-  }
-  RTCPPacketXRDLRRReportBlockItem dlrr;
-  dlrr.SSRC = ssrc;
-  dlrr.LastRR = last_rr;
-  dlrr.DelayLastRR = delay_last_rr;
-  dlrr_block_.push_back(dlrr);
-  return true;
-}
-
 RawPacket::RawPacket(size_t buffer_length)
     : buffer_length_(buffer_length), length_(0) {
   buffer_.reset(new uint8_t[buffer_length]);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
index 3c34957c36..01c97c38ba 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -9,16 +9,20 @@
  *
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -27,10 +31,7 @@ namespace rtcp {
 static const int kCommonFbFmtLength = 12;
 static const int kReportBlockLength = 24;
 
-class Dlrr;
 class RawPacket;
-class Rrtr;
-class VoipMetric;
 
 // Class for building RTCP packets.
 //
@@ -115,93 +116,17 @@ class RtcpPacket {
   size_t HeaderLength() const;
 
   static const size_t kHeaderLength = 4;
+  std::vector<RtcpPacket*> appended_packets_;
 
  private:
   bool CreateAndAddAppended(uint8_t* packet,
                             size_t* index,
                             size_t max_length,
                             PacketReadyCallback* callback) const;
-
-  std::vector<RtcpPacket*> appended_packets_;
 };
 
 // TODO(sprang): Move RtcpPacket subclasses out to separate files.
 
-class Empty : public RtcpPacket {
- public:
-  Empty() : RtcpPacket() {}
-
-  virtual ~Empty() {}
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
-  size_t BlockLength() const override;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(Empty);
-};
-
-// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
-//
-// RTCP report block (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_1 (SSRC of first source)                 |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  | fraction lost |       cumulative number of packets lost       |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |           extended highest sequence number received           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                      interarrival jitter                      |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                         last SR (LSR)                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   delay since last SR (DLSR)                  |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-
-class ReportBlock {
- public:
-  ReportBlock() {
-    // TODO(asapersson): Consider adding a constructor to struct.
-    memset(&report_block_, 0, sizeof(report_block_));
-  }
-
-  ~ReportBlock() {}
-
-  void To(uint32_t ssrc) {
-    report_block_.SSRC = ssrc;
-  }
-  void WithFractionLost(uint8_t fraction_lost) {
-    report_block_.FractionLost = fraction_lost;
-  }
-  void WithCumulativeLost(uint32_t cumulative_lost) {
-    report_block_.CumulativeNumOfPacketsLost = cumulative_lost;
-  }
-  void WithExtHighestSeqNum(uint32_t ext_highest_seq_num) {
-    report_block_.ExtendedHighestSequenceNumber = ext_highest_seq_num;
-  }
-  void WithJitter(uint32_t jitter) {
-    report_block_.Jitter = jitter;
-  }
-  void WithLastSr(uint32_t last_sr) {
-    report_block_.LastSR = last_sr;
-  }
-  void WithDelayLastSr(uint32_t delay_last_sr) {
-    report_block_.DelayLastSR = delay_last_sr;
-  }
-
- private:
-  friend class SenderReport;
-  friend class ReceiverReport;
-  RTCPUtility::RTCPPacketReportBlockItem report_block_;
-};
-
 // RTCP sender report (RFC 3550).
 //
 //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ -268,102 +193,11 @@ class SenderReport : public RtcpPacket {
   }
 
   RTCPUtility::RTCPPacketSR sr_;
-  std::vector<RTCPUtility::RTCPPacketReportBlockItem> report_blocks_;
+  std::vector<ReportBlock> report_blocks_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(SenderReport);
 };
 
-//
-// RTCP receiver report (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P|    RC   |   PT=RR=201   |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                     SSRC of packet sender                     |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                         report block(s)                       |
-//  |                            ....                               |
-
-class ReceiverReport : public RtcpPacket {
- public:
-  ReceiverReport() : RtcpPacket() {
-    memset(&rr_, 0, sizeof(rr_));
-  }
-
-  virtual ~ReceiverReport() {}
-
-  void From(uint32_t ssrc) {
-    rr_.SenderSSRC = ssrc;
-  }
-  bool WithReportBlock(const ReportBlock& block);
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  static const int kMaxNumberOfReportBlocks = 0x1F;
-
-  size_t BlockLength() const {
-    const size_t kRrHeaderLength = 8;
-    return kRrHeaderLength + report_blocks_.size() * kReportBlockLength;
-  }
-
-  RTCPUtility::RTCPPacketRR rr_;
-  std::vector<RTCPUtility::RTCPPacketReportBlockItem> report_blocks_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(ReceiverReport);
-};
-
-// Transmission Time Offsets in RTP Streams (RFC 5450).
-//
-//      0                   1                   2                   3
-//      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     .                                                               .
-//     .                                                               .
-//     .                                                               .
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//
-//  If present, this RTCP packet must be placed after a receiver report
-//  (inside a compound RTCP packet), and MUST have the same value for RC
-//  (reception report count) as the receiver report.
-
-class Ij : public RtcpPacket {
- public:
-  Ij() : RtcpPacket() {}
-
-  virtual ~Ij() {}
-
-  bool WithJitterItem(uint32_t jitter);
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  static const int kMaxNumberOfIjItems = 0x1f;
-
-  size_t BlockLength() const {
-    return kHeaderLength + 4 * ij_items_.size();
-  }
-
-  std::vector<uint32_t> ij_items_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Ij);
-};
-
 // Source Description (SDES) (RFC 3550).
 //
 //         0                   1                   2                   3
@@ -420,262 +254,6 @@ class Sdes : public RtcpPacket {
   RTC_DISALLOW_COPY_AND_ASSIGN(Sdes);
 };
 
-//
-// Bye packet (BYE) (RFC 3550).
-//
-//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |                           SSRC/CSRC                           |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       :                              ...                              :
-//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-// (opt) |     length    |               reason for leaving            ...
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Bye : public RtcpPacket {
- public:
-  Bye() : RtcpPacket() {
-    memset(&bye_, 0, sizeof(bye_));
-  }
-
-  virtual ~Bye() {}
-
-  void From(uint32_t ssrc) {
-    bye_.SenderSSRC = ssrc;
-  }
-
-  bool WithCsrc(uint32_t csrc);
-
-  // TODO(sprang): Add support for reason field?
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  static const int kMaxNumberOfCsrcs = 0x1f - 1;  // First item is sender SSRC.
-
-  size_t BlockLength() const {
-    size_t source_count = 1 + csrcs_.size();
-    return kHeaderLength + 4 * source_count;
-  }
-
-  RTCPUtility::RTCPPacketBYE bye_;
-  std::vector<uint32_t> csrcs_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Bye);
-};
-
-// Application-Defined packet (APP) (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P| subtype |   PT=APP=204  |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                           SSRC/CSRC                           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                          name (ASCII)                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   application-dependent data                ...
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class App : public RtcpPacket {
- public:
-  App()
-      : RtcpPacket(),
-        ssrc_(0) {
-    memset(&app_, 0, sizeof(app_));
-  }
-
-  virtual ~App() {}
-
-  void From(uint32_t ssrc) {
-    ssrc_ = ssrc;
-  }
-  void WithSubType(uint8_t subtype) {
-    assert(subtype <= 0x1f);
-    app_.SubType = subtype;
-  }
-  void WithName(uint32_t name) {
-    app_.Name = name;
-  }
-  void WithData(const uint8_t* data, uint16_t data_length) {
-    assert(data);
-    assert(data_length <= kRtcpAppCode_DATA_SIZE);
-    assert(data_length % 4 == 0);
-    memcpy(app_.Data, data, data_length);
-    app_.Size = data_length;
-  }
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  size_t BlockLength() const {
-    return 12 + app_.Size;
-  }
-
-  uint32_t ssrc_;
-  RTCPUtility::RTCPPacketAPP app_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(App);
-};
-
-// RFC 4585: Feedback format.
-//
-// Common packet format:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |V=2|P|   FMT   |       PT      |          length               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of packet sender                        |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of media source                         |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   :            Feedback Control Information (FCI)                 :
-//   :
-
-// Picture loss indication (PLI) (RFC 4585).
-//
-// FCI: no feedback control information.
-
-class Pli : public RtcpPacket {
- public:
-  Pli() : RtcpPacket() {
-    memset(&pli_, 0, sizeof(pli_));
-  }
-
-  virtual ~Pli() {}
-
-  void From(uint32_t ssrc) {
-    pli_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    pli_.MediaSSRC = ssrc;
-  }
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  size_t BlockLength() const {
-    return kCommonFbFmtLength;
-  }
-
-  RTCPUtility::RTCPPacketPSFBPLI pli_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Pli);
-};
-
-// Slice loss indication (SLI) (RFC 4585).
-//
-// FCI:
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            First        |        Number           | PictureID |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Sli : public RtcpPacket {
- public:
-  Sli() : RtcpPacket() {
-    memset(&sli_, 0, sizeof(sli_));
-    memset(&sli_item_, 0, sizeof(sli_item_));
-  }
-
-  virtual ~Sli() {}
-
-  void From(uint32_t ssrc) {
-    sli_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    sli_.MediaSSRC = ssrc;
-  }
-  void WithFirstMb(uint16_t first_mb) {
-    assert(first_mb <= 0x1fff);
-    sli_item_.FirstMB = first_mb;
-  }
-  void WithNumberOfMb(uint16_t number_mb) {
-    assert(number_mb <= 0x1fff);
-    sli_item_.NumberOfMB = number_mb;
-  }
-  void WithPictureId(uint8_t picture_id) {
-    assert(picture_id <= 0x3f);
-    sli_item_.PictureId = picture_id;
-  }
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  size_t BlockLength() const {
-    const size_t kFciLength = 4;
-    return kCommonFbFmtLength + kFciLength;
-  }
-
-  RTCPUtility::RTCPPacketPSFBSLI sli_;
-  RTCPUtility::RTCPPacketPSFBSLIItem sli_item_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Sli);
-};
-
-// Generic NACK (RFC 4585).
-//
-// FCI:
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            PID                |             BLP               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Nack : public RtcpPacket {
- public:
-  Nack() : RtcpPacket() {
-    memset(&nack_, 0, sizeof(nack_));
-  }
-
-  virtual ~Nack() {}
-
-  void From(uint32_t ssrc) {
-    nack_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    nack_.MediaSSRC = ssrc;
-  }
-  void WithList(const uint16_t* nack_list, int length);
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
-  size_t BlockLength() const override;
-
- private:
-
-  RTCPUtility::RTCPPacketRTPFBNACK nack_;
-  std::vector<RTCPUtility::RTCPPacketRTPFBNACKItem> nack_fields_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Nack);
-};
-
 // Reference picture selection indication (RPSI) (RFC 4585).
 //
 // FCI:
@@ -775,105 +353,6 @@ class Fir : public RtcpPacket {
   RTCPUtility::RTCPPacketPSFBFIRItem fir_item_;
 };
 
-// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Tmmbr : public RtcpPacket {
- public:
-  Tmmbr() : RtcpPacket() {
-    memset(&tmmbr_, 0, sizeof(tmmbr_));
-    memset(&tmmbr_item_, 0, sizeof(tmmbr_item_));
-  }
-
-  virtual ~Tmmbr() {}
-
-  void From(uint32_t ssrc) {
-    tmmbr_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    tmmbr_item_.SSRC = ssrc;
-  }
-  void WithBitrateKbps(uint32_t bitrate_kbps) {
-    tmmbr_item_.MaxTotalMediaBitRate = bitrate_kbps;
-  }
-  void WithOverhead(uint16_t overhead) {
-    assert(overhead <= 0x1ff);
-    tmmbr_item_.MeasuredOverhead = overhead;
-  }
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  size_t BlockLength() const {
-    const size_t kFciLen = 8;
-    return kCommonFbFmtLength + kFciLen;
-  }
-
-  RTCPUtility::RTCPPacketRTPFBTMMBR tmmbr_;
-  RTCPUtility::RTCPPacketRTPFBTMMBRItem tmmbr_item_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Tmmbr);
-};
-
-// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Tmmbn : public RtcpPacket {
- public:
-  Tmmbn() : RtcpPacket() {
-    memset(&tmmbn_, 0, sizeof(tmmbn_));
-  }
-
-  virtual ~Tmmbn() {}
-
-  void From(uint32_t ssrc) {
-    tmmbn_.SenderSSRC = ssrc;
-  }
-  // Max 50 TMMBR can be added per TMMBN.
-  bool WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead);
-
- protected:
-  bool Create(uint8_t* packet,
-              size_t* index,
-              size_t max_length,
-              RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
-  static const int kMaxNumberOfTmmbrs = 50;
-
-  size_t BlockLength() const {
-    const size_t kFciLen = 8;
-    return kCommonFbFmtLength + kFciLen * tmmbn_items_.size();
-  }
-
-  RTCPUtility::RTCPPacketRTPFBTMMBN tmmbn_;
-  std::vector<RTCPUtility::RTCPPacketRTPFBTMMBRItem> tmmbn_items_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Tmmbn);
-};
-
 // Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
 //
 //    0                   1                   2                   3
@@ -978,163 +457,22 @@ class Xr : public RtcpPacket {
     return kXrHeaderLength + RrtrLength() + DlrrLength() + VoipMetricLength();
   }
 
-  size_t RrtrLength() const {
-    const size_t kRrtrBlockLength = 12;
-    return kRrtrBlockLength * rrtr_blocks_.size();
-  }
+  size_t RrtrLength() const { return Rrtr::kLength * rrtr_blocks_.size(); }
 
   size_t DlrrLength() const;
 
   size_t VoipMetricLength() const {
-    const size_t kVoipMetricBlockLength = 36;
-    return kVoipMetricBlockLength * voip_metric_blocks_.size();
+    return VoipMetric::kLength * voip_metric_blocks_.size();
   }
 
   RTCPUtility::RTCPPacketXR xr_header_;
-  std::vector<RTCPUtility::RTCPPacketXRReceiverReferenceTimeItem> rrtr_blocks_;
-  std::vector<DlrrBlock> dlrr_blocks_;
-  std::vector<RTCPUtility::RTCPPacketXRVOIPMetricItem> voip_metric_blocks_;
+  std::vector<Rrtr> rrtr_blocks_;
+  std::vector<Dlrr> dlrr_blocks_;
+  std::vector<VoipMetric> voip_metric_blocks_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Xr);
 };
 
-// Receiver Reference Time Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=4      |   reserved    |       block length = 2        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |              NTP timestamp, most significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |             NTP timestamp, least significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Rrtr {
- public:
-  Rrtr() {
-    memset(&rrtr_block_, 0, sizeof(rrtr_block_));
-  }
-  ~Rrtr() {}
-
-  void WithNtpSec(uint32_t sec) {
-    rrtr_block_.NTPMostSignificant = sec;
-  }
-  void WithNtpFrac(uint32_t frac) {
-    rrtr_block_.NTPLeastSignificant = frac;
-  }
-
- private:
-  friend class Xr;
-  RTCPUtility::RTCPPacketXRReceiverReferenceTimeItem rrtr_block_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Rrtr);
-};
-
-// DLRR Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=5      |   reserved    |         block length          |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_1 (SSRC of first receiver)               | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  |                         last RR (LRR)                         |   1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   delay since last RR (DLRR)                  |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_2 (SSRC of second receiver)              | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  :                               ...                             :   2
-
-class Dlrr {
- public:
-  Dlrr() {}
-  ~Dlrr() {}
-
-  // Max 100 DLRR Items can be added per DLRR report block.
-  bool WithDlrrItem(uint32_t ssrc, uint32_t last_rr, uint32_t delay_last_rr);
-
- private:
-  friend class Xr;
-  static const int kMaxNumberOfDlrrItems = 100;
-
-  std::vector<RTCPUtility::RTCPPacketXRDLRRReportBlockItem> dlrr_block_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Dlrr);
-};
-
-// VoIP Metrics Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=7      |   reserved    |       block length = 8        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                        SSRC of source                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   loss rate   | discard rate  | burst density |  gap density  |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |       burst duration          |         gap duration          |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     round trip delay          |       end system delay        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  | signal level  |  noise level  |     RERL      |     Gmin      |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   RX config   |   reserved    |          JB nominal           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |          JB maximum           |          JB abs max           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class VoipMetric {
- public:
-  VoipMetric() {
-    memset(&metric_, 0, sizeof(metric_));
-  }
-  ~VoipMetric() {}
-
-  void To(uint32_t ssrc) { metric_.SSRC = ssrc; }
-  void LossRate(uint8_t loss_rate) { metric_.lossRate = loss_rate; }
-  void DiscardRate(uint8_t discard_rate) { metric_.discardRate = discard_rate; }
-  void BurstDensity(uint8_t burst_density) {
-    metric_.burstDensity = burst_density;
-  }
-  void GapDensity(uint8_t gap_density) { metric_.gapDensity = gap_density; }
-  void BurstDuration(uint16_t burst_duration) {
-    metric_.burstDuration = burst_duration;
-  }
-  void GapDuration(uint16_t gap_duration) {
-    metric_.gapDuration = gap_duration;
-  }
-  void RoundTripDelay(uint16_t round_trip_delay) {
-    metric_.roundTripDelay = round_trip_delay;
-  }
-  void EndSystemDelay(uint16_t end_system_delay) {
-    metric_.endSystemDelay = end_system_delay;
-  }
-  void SignalLevel(uint8_t signal_level) { metric_.signalLevel = signal_level; }
-  void NoiseLevel(uint8_t noise_level) { metric_.noiseLevel = noise_level; }
-  void Rerl(uint8_t rerl) { metric_.RERL = rerl; }
-  void Gmin(uint8_t gmin) { metric_.Gmin = gmin; }
-  void Rfactor(uint8_t rfactor) { metric_.Rfactor = rfactor; }
-  void ExtRfactor(uint8_t extrfactor) { metric_.extRfactor = extrfactor; }
-  void MosLq(uint8_t moslq) { metric_.MOSLQ = moslq; }
-  void MosCq(uint8_t moscq) { metric_.MOSCQ = moscq; }
-  void RxConfig(uint8_t rxconfig) { metric_.RXconfig = rxconfig; }
-  void JbNominal(uint16_t jbnominal) { metric_.JBnominal = jbnominal; }
-  void JbMax(uint16_t jbmax) { metric_.JBmax = jbmax; }
-  void JbAbsMax(uint16_t jbabsmax) { metric_.JBabsMax = jbabsmax; }
-
- private:
-  friend class Xr;
-  RTCPUtility::RTCPPacketXRVOIPMetricItem metric_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(VoipMetric);
-};
-
 // Class holding a RTCP packet.
 //
 // Takes a built rtcp packet.
@@ -1163,4 +501,4 @@ class RawPacket {
 
 }  // namespace rtcp
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
new file mode 100644
index 0000000000..a1ad8d6427
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// Application-Defined packet (APP) (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| subtype |   PT=APP=204  |             length            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |                           SSRC/CSRC                           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          name (ASCII)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |                   application-dependent data                ...
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool App::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  sub_type_ = header.count_or_format;
+  ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
+  name_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+  data_.SetData(&payload[8], header.payload_size_bytes - 8);
+  return true;
+}
+
+void App::WithSubType(uint8_t subtype) {
+  RTC_DCHECK_LE(subtype, 0x1f);
+  sub_type_ = subtype;
+}
+
+void App::WithData(const uint8_t* data, size_t data_length) {
+  RTC_DCHECK(data);
+  RTC_DCHECK_EQ(0u, data_length % 4) << "Data must be 32 bits aligned.";
+  RTC_DCHECK(data_length <= kMaxDataSize) << "App data size << " << data_length
+                                          << "exceed maximum of "
+                                          << kMaxDataSize << " bytes.";
+  data_.SetData(data, data_length);
+}
+
+bool App::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
+
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
+  memcpy(&packet[*index + 8], data_.data(), data_.size());
+  *index += (8 + data_.size());
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
new file mode 100644
index 0000000000..16bd3fc2a2
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class App : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 204;
+  // 28 bytes for UDP header
+  // 12 bytes for RTCP app header
+  static const size_t kMaxDataSize = IP_PACKET_SIZE - 12 - 28;
+  App() : sub_type_(0), ssrc_(0), name_(0) {}
+
+  virtual ~App() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void From(uint32_t ssrc) { ssrc_ = ssrc; }
+  void WithSubType(uint8_t subtype);
+  void WithName(uint32_t name) { name_ = name; }
+  void WithData(const uint8_t* data, size_t data_length);
+
+  uint8_t sub_type() const { return sub_type_; }
+  uint32_t ssrc() const { return ssrc_; }
+  uint32_t name() const { return name_; }
+  size_t data_size() const { return data_.size(); }
+  const uint8_t* data() const { return data_.data(); }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override { return 12 + data_.size(); }
+
+  uint8_t sub_type_;
+  uint32_t ssrc_;
+  uint32_t name_;
+  rtc::Buffer data_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(App);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
new file mode 100644
index 0000000000..4451fe8fb5
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using webrtc::rtcp::App;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kName = ((uint32_t)'n' << 24) | ((uint32_t)'a' << 16) |
+                       ((uint32_t)'m' << 8) | (uint32_t)'e';
+const uint32_t kSenderSsrc = 0x12345678;
+
+class RtcpPacketAppTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = app.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    // Check there is exactly one RTCP packet in the buffer.
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  App app;
+  rtc::scoped_ptr<RawPacket> packet;
+  const App& parsed() { return parsed_; }
+
+ private:
+  App parsed_;
+};
+
+TEST_F(RtcpPacketAppTest, WithNoData) {
+  app.WithSubType(30);
+  app.WithName(kName);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(30U, parsed().sub_type());
+  EXPECT_EQ(kName, parsed().name());
+  EXPECT_EQ(0u, parsed().data_size());
+}
+
+TEST_F(RtcpPacketAppTest, WithData) {
+  app.From(kSenderSsrc);
+  app.WithSubType(30);
+  app.WithName(kName);
+  const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't', 'a'};
+  const size_t kDataLength = sizeof(kData) / sizeof(kData[0]);
+  app.WithData(kData, kDataLength);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(30U, parsed().sub_type());
+  EXPECT_EQ(kName, parsed().name());
+  EXPECT_EQ(kDataLength, parsed().data_size());
+  EXPECT_EQ(0, memcmp(kData, parsed().data(), kDataLength));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
new file mode 100644
index 0000000000..4cfc921ce5
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// Bye packet (BYE) (RFC 3550).
+//
+//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |                           SSRC/CSRC                           |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       :                              ...                              :
+//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// (opt) |     length    |               reason for leaving            ...
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Bye::Bye() : sender_ssrc_(0) {}
+
+bool Bye::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t src_count = header.count_or_format;
+  // Validate packet.
+  if (header.payload_size_bytes < 4u * src_count) {
+    LOG(LS_WARNING)
+        << "Packet is too small to contain CSRCs it promise to have.";
+    return false;
+  }
+  bool has_reason = (header.payload_size_bytes > 4u * src_count);
+  uint8_t reason_length = 0;
+  if (has_reason) {
+    reason_length = payload[4u * src_count];
+    if (header.payload_size_bytes - 4u * src_count < 1u + reason_length) {
+      LOG(LS_WARNING) << "Invalid reason length: " << reason_length;
+      return false;
+    }
+  }
+  // Once sure packet is valid, copy values.
+  if (src_count == 0) {  // A count value of zero is valid, but useless.
+    sender_ssrc_ = 0;
+    csrcs_.clear();
+  } else {
+    sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(payload);
+    csrcs_.resize(src_count - 1);
+    for (size_t i = 1; i < src_count; ++i)
+      csrcs_[i - 1] = ByteReader<uint32_t>::ReadBigEndian(&payload[4 * i]);
+  }
+
+  if (has_reason) {
+    reason_.assign(reinterpret_cast<const char*>(&payload[4u * src_count + 1]),
+                   reason_length);
+  } else {
+    reason_.clear();
+  }
+
+  return true;
+}
+
+bool Bye::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(1 + csrcs_.size(), kPacketType, HeaderLength(), packet, index);
+  // Store srcs of the leaving clients.
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], sender_ssrc_);
+  *index += sizeof(uint32_t);
+  for (uint32_t csrc : csrcs_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], csrc);
+    *index += sizeof(uint32_t);
+  }
+  // Store the reason to leave.
+  if (!reason_.empty()) {
+    uint8_t reason_length = reason_.size();
+    packet[(*index)++] = reason_length;
+    memcpy(&packet[*index], reason_.data(), reason_length);
+    *index += reason_length;
+    // Add padding bytes if needed.
+    size_t bytes_to_pad = index_end - *index;
+    RTC_DCHECK_LE(bytes_to_pad, 3u);
+    if (bytes_to_pad > 0) {
+      memset(&packet[*index], 0, bytes_to_pad);
+      *index += bytes_to_pad;
+    }
+  }
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+bool Bye::WithCsrc(uint32_t csrc) {
+  if (csrcs_.size() >= kMaxNumberOfCsrcs) {
+    LOG(LS_WARNING) << "Max CSRC size reached.";
+    return false;
+  }
+  csrcs_.push_back(csrc);
+  return true;
+}
+
+void Bye::WithReason(const std::string& reason) {
+  RTC_DCHECK_LE(reason.size(), 0xffu);
+  reason_ = reason;
+}
+
+size_t Bye::BlockLength() const {
+  size_t src_count = (1 + csrcs_.size());
+  size_t reason_size_in_32bits = reason_.empty() ? 0 : (reason_.size() / 4 + 1);
+  return kHeaderLength + 4 * (src_count + reason_size_in_32bits);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
new file mode 100644
index 0000000000..6b4a181330
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Bye : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 203;
+
+  Bye();
+  virtual ~Bye() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  bool WithCsrc(uint32_t csrc);
+  void WithReason(const std::string& reason);
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  const std::vector<uint32_t>& csrcs() const { return csrcs_; }
+  const std::string& reason() const { return reason_; }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const int kMaxNumberOfCsrcs = 0x1f - 1;  // First item is sender SSRC.
+
+  size_t BlockLength() const override;
+
+  uint32_t sender_ssrc_;
+  std::vector<uint32_t> csrcs_;
+  std::string reason_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Bye);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
new file mode 100644
index 0000000000..d2ae8ed782
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using ::testing::ElementsAre;
+
+using webrtc::rtcp::Bye;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kCsrc1 = 0x22232425;
+const uint32_t kCsrc2 = 0x33343536;
+
+class RtcpPacketByeTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = bye.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    // Check that there is exactly one RTCP packet in the buffer.
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_bye.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  Bye bye;
+  rtc::scoped_ptr<RawPacket> packet;
+  Bye parsed_bye;
+};
+
+TEST_F(RtcpPacketByeTest, Bye) {
+  bye.From(kSenderSsrc);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST_F(RtcpPacketByeTest, WithCsrcs) {
+  bye.From(kSenderSsrc);
+  EXPECT_TRUE(bye.WithCsrc(kCsrc1));
+  EXPECT_TRUE(bye.WithCsrc(kCsrc2));
+  EXPECT_TRUE(bye.reason().empty());
+
+  BuildPacket();
+  EXPECT_EQ(16u, packet->Length());  // Header: 4, 3xSRCs: 12, Reason: 0.
+
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2));
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST_F(RtcpPacketByeTest, WithCsrcsAndReason) {
+  const std::string kReason = "Some Reason";
+
+  bye.From(kSenderSsrc);
+  EXPECT_TRUE(bye.WithCsrc(kCsrc1));
+  EXPECT_TRUE(bye.WithCsrc(kCsrc2));
+  bye.WithReason(kReason);
+
+  BuildPacket();
+  EXPECT_EQ(28u, packet->Length());  // Header: 4, 3xSRCs: 12, Reason: 12.
+
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2));
+  EXPECT_EQ(kReason, parsed_bye.reason());
+}
+
+TEST_F(RtcpPacketByeTest, WithTooManyCsrcs) {
+  bye.From(kSenderSsrc);
+  const int kMaxCsrcs = (1 << 5) - 2;  // 5 bit len, first item is sender SSRC.
+  for (int i = 0; i < kMaxCsrcs; ++i) {
+    EXPECT_TRUE(bye.WithCsrc(i));
+  }
+  EXPECT_FALSE(bye.WithCsrc(kMaxCsrcs));
+}
+
+TEST_F(RtcpPacketByeTest, WithAReason) {
+  const std::string kReason = "Some Random Reason";
+
+  bye.From(kSenderSsrc);
+  bye.WithReason(kReason);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_EQ(kReason, parsed_bye.reason());
+}
+
+TEST_F(RtcpPacketByeTest, WithReasons) {
+  // Test that packet creation/parsing behave with reasons of different length
+  // both when it require padding and when it does not.
+  for (size_t reminder = 0; reminder < 4; ++reminder) {
+    const std::string kReason(4 + reminder, 'a' + reminder);
+    bye.From(kSenderSsrc);
+    bye.WithReason(kReason);
+
+    BuildPacket();
+    ParsePacket();
+
+    EXPECT_EQ(kReason, parsed_bye.reason());
+  }
+}
+
+TEST_F(RtcpPacketByeTest, ParseEmptyPacket) {
+  RtcpCommonHeader header;
+  header.packet_type = Bye::kPacketType;
+  header.count_or_format = 0;
+  header.payload_size_bytes = 0;
+  uint8_t empty_payload[1];
+
+  EXPECT_TRUE(parsed_bye.Parse(header, empty_payload + 1));
+  EXPECT_EQ(0u, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST_F(RtcpPacketByeTest, ParseFailOnInvalidSrcCount) {
+  bye.From(kSenderSsrc);
+
+  BuildPacket();
+
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header);
+  header.count_or_format = 2;  // Lie there are 2 ssrcs, not one.
+
+  EXPECT_FALSE(parsed_bye.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+TEST_F(RtcpPacketByeTest, ParseFailOnInvalidReasonLength) {
+  bye.From(kSenderSsrc);
+  bye.WithReason("18 characters long");
+
+  BuildPacket();
+
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header);
+  header.payload_size_bytes -= 4;  // Payload is usually 32bit aligned.
+
+  EXPECT_FALSE(parsed_bye.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc
new file mode 100644
index 0000000000..8f5afd5dd1
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+bool CompoundPacket::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            RtcpPacket::PacketReadyCallback* callback) const {
+  return true;
+}
+
+size_t CompoundPacket::BlockLength() const {
+  return 0;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
new file mode 100644
index 0000000000..f2f49a8ffb
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class CompoundPacket : public RtcpPacket {
+ public:
+  CompoundPacket() : RtcpPacket() {}
+
+  virtual ~CompoundPacket() {}
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+  size_t BlockLength() const override;
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(CompoundPacket);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
new file mode 100644
index 0000000000..83dc5f6ed3
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/test/rtcp_packet_parser.h"
+
+using webrtc::rtcp::Bye;
+using webrtc::rtcp::CompoundPacket;
+using webrtc::rtcp::Fir;
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ReceiverReport;
+using webrtc::rtcp::ReportBlock;
+using webrtc::rtcp::SenderReport;
+using webrtc::test::RtcpPacketParser;
+
+namespace webrtc {
+
+const uint32_t kSenderSsrc = 0x12345678;
+
+TEST(RtcpCompoundPacketTest, AppendPacket) {
+  Fir fir;
+  ReportBlock rb;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  rtc::scoped_ptr<RawPacket> packet(rr.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
+  EXPECT_EQ(1, parser.report_block()->num_packets());
+  EXPECT_EQ(1, parser.fir()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, AppendPacketOnEmpty) {
+  CompoundPacket empty;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  empty.Append(&rr);
+
+  rtc::scoped_ptr<RawPacket> packet(empty.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(0, parser.report_block()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, AppendPacketWithOwnAppendedPacket) {
+  Fir fir;
+  Bye bye;
+  ReportBlock rb;
+
+  ReceiverReport rr;
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  SenderReport sr;
+  sr.Append(&bye);
+  sr.Append(&rr);
+
+  rtc::scoped_ptr<RawPacket> packet(sr.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.sender_report()->num_packets());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(1, parser.report_block()->num_packets());
+  EXPECT_EQ(1, parser.bye()->num_packets());
+  EXPECT_EQ(1, parser.fir()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, BuildWithInputBuffer) {
+  Fir fir;
+  ReportBlock rb;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  const size_t kRrLength = 8;
+  const size_t kReportBlockLength = 24;
+  const size_t kFirLength = 20;
+
+  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      RtcpPacketParser parser;
+      parser.Parse(data, length);
+      EXPECT_EQ(1, parser.receiver_report()->num_packets());
+      EXPECT_EQ(1, parser.report_block()->num_packets());
+      EXPECT_EQ(1, parser.fir()->num_packets());
+      ++packets_created_;
+    }
+
+    int packets_created_ = 0;
+  } verifier;
+  const size_t kBufferSize = kRrLength + kReportBlockLength + kFirLength;
+  uint8_t buffer[kBufferSize];
+  EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
+  EXPECT_EQ(1, verifier.packets_created_);
+}
+
+TEST(RtcpCompoundPacketTest, BuildWithTooSmallBuffer_FragmentedSend) {
+  Fir fir;
+  ReportBlock rb;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  const size_t kRrLength = 8;
+  const size_t kReportBlockLength = 24;
+
+  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      RtcpPacketParser parser;
+      parser.Parse(data, length);
+      switch (packets_created_++) {
+        case 0:
+          EXPECT_EQ(1, parser.receiver_report()->num_packets());
+          EXPECT_EQ(1, parser.report_block()->num_packets());
+          EXPECT_EQ(0, parser.fir()->num_packets());
+          break;
+        case 1:
+          EXPECT_EQ(0, parser.receiver_report()->num_packets());
+          EXPECT_EQ(0, parser.report_block()->num_packets());
+          EXPECT_EQ(1, parser.fir()->num_packets());
+          break;
+        default:
+          ADD_FAILURE() << "OnPacketReady not expected to be called "
+                        << packets_created_ << " times.";
+      }
+    }
+
+    int packets_created_ = 0;
+  } verifier;
+  const size_t kBufferSize = kRrLength + kReportBlockLength;
+  uint8_t buffer[kBufferSize];
+  EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
+  EXPECT_EQ(2, verifier.packets_created_);
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
new file mode 100644
index 0000000000..6d6c48fada
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+// DLRR Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=5      |   reserved    |         block length          |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_1 (SSRC of first receiver)               | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  |                         last RR (LRR)                         |   1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                   delay since last RR (DLRR)                  |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_2 (SSRC of second receiver)              | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  :                               ...                             :   2
+bool Dlrr::Parse(const uint8_t* buffer, uint16_t block_length_32bits) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // kReserved = buffer[1];
+  RTC_DCHECK_EQ(block_length_32bits,
+                ByteReader<uint16_t>::ReadBigEndian(&buffer[2]));
+  if (block_length_32bits % 3 != 0) {
+    LOG(LS_WARNING) << "Invalid size for dlrr block.";
+    return false;
+  }
+
+  size_t blocks_count = block_length_32bits / 3;
+  const uint8_t* read_at = buffer + kBlockHeaderLength;
+  sub_blocks_.resize(blocks_count);
+  for (SubBlock& sub_block : sub_blocks_) {
+    sub_block.ssrc = ByteReader<uint32_t>::ReadBigEndian(&read_at[0]);
+    sub_block.last_rr = ByteReader<uint32_t>::ReadBigEndian(&read_at[4]);
+    sub_block.delay_since_last_rr =
+        ByteReader<uint32_t>::ReadBigEndian(&read_at[8]);
+    read_at += kSubBlockLength;
+  }
+  return true;
+}
+
+size_t Dlrr::BlockLength() const {
+  if (sub_blocks_.empty())
+    return 0;
+  return kBlockHeaderLength + kSubBlockLength * sub_blocks_.size();
+}
+
+void Dlrr::Create(uint8_t* buffer) const {
+  if (sub_blocks_.empty())  // No subblocks, no need to write header either.
+    return;
+  // Create block header.
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], 3 * sub_blocks_.size());
+  // Create sub blocks.
+  uint8_t* write_at = buffer + kBlockHeaderLength;
+  for (const SubBlock& sub_block : sub_blocks_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[0], sub_block.ssrc);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[4], sub_block.last_rr);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[8],
+                                         sub_block.delay_since_last_rr);
+    write_at += kSubBlockLength;
+  }
+  RTC_DCHECK_EQ(buffer + BlockLength(), write_at);
+}
+
+bool Dlrr::WithDlrrItem(uint32_t ssrc,
+                        uint32_t last_rr,
+                        uint32_t delay_last_rr) {
+  if (sub_blocks_.size() >= kMaxNumberOfDlrrItems) {
+    LOG(LS_WARNING) << "Max DLRR items reached.";
+    return false;
+  }
+  SubBlock block;
+  block.ssrc = ssrc;
+  block.last_rr = last_rr;
+  block.delay_since_last_rr = delay_last_rr;
+  sub_blocks_.push_back(block);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
new file mode 100644
index 0000000000..9af2dedf3f
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// DLRR Report Block: Delay since the Last Receiver Report (RFC 3611).
+class Dlrr {
+ public:
+  struct SubBlock {
+    // RFC 3611 4.5
+    uint32_t ssrc;
+    uint32_t last_rr;
+    uint32_t delay_since_last_rr;
+  };
+
+  static const uint8_t kBlockType = 5;
+  static const size_t kMaxNumberOfDlrrItems = 100;
+
+  Dlrr() {}
+  Dlrr(const Dlrr& other) = default;
+  ~Dlrr() {}
+
+  Dlrr& operator=(const Dlrr& other) = default;
+
+  // Second parameter is value read from block header,
+  // i.e. size of block in 32bits excluding block header itself.
+  bool Parse(const uint8_t* buffer, uint16_t block_length_32bits);
+
+  size_t BlockLength() const;
+  // Fills buffer with the Dlrr.
+  // Consumes BlockLength() bytes.
+  void Create(uint8_t* buffer) const;
+
+  // Max 100 DLRR Items can be added per DLRR report block.
+  bool WithDlrrItem(uint32_t ssrc, uint32_t last_rr, uint32_t delay_last_rr);
+
+  const std::vector<SubBlock>& sub_blocks() const { return sub_blocks_; }
+
+ private:
+  static const size_t kBlockHeaderLength = 4;
+  static const size_t kSubBlockLength = 12;
+
+  std::vector<SubBlock> sub_blocks_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc
new file mode 100644
index 0000000000..c7c139c560
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::rtcp::Dlrr;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSsrc = 0x12345678;
+const uint32_t kLastRR = 0x23344556;
+const uint32_t kDelay = 0x33343536;
+const uint8_t kBlock[] = {0x05, 0x00, 0x00, 0x03, 0x12, 0x34, 0x56, 0x78,
+                          0x23, 0x34, 0x45, 0x56, 0x33, 0x34, 0x35, 0x36};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+
+TEST(RtcpPacketDlrrTest, Empty) {
+  Dlrr dlrr;
+
+  EXPECT_EQ(0u, dlrr.BlockLength());
+}
+
+TEST(RtcpPacketDlrrTest, Create) {
+  Dlrr dlrr;
+  EXPECT_TRUE(dlrr.WithDlrrItem(kSsrc, kLastRR, kDelay));
+
+  ASSERT_EQ(kBlockSizeBytes, dlrr.BlockLength());
+  uint8_t buffer[kBlockSizeBytes];
+
+  dlrr.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketDlrrTest, Parse) {
+  Dlrr dlrr;
+  uint16_t block_length = ByteReader<uint16_t>::ReadBigEndian(&kBlock[2]);
+  EXPECT_TRUE(dlrr.Parse(kBlock, block_length));
+
+  EXPECT_EQ(1u, dlrr.sub_blocks().size());
+  const Dlrr::SubBlock& block = dlrr.sub_blocks().front();
+  EXPECT_EQ(kSsrc, block.ssrc);
+  EXPECT_EQ(kLastRR, block.last_rr);
+  EXPECT_EQ(kDelay, block.delay_since_last_rr);
+}
+
+TEST(RtcpPacketDlrrTest, ParseFailsOnBadSize) {
+  const size_t kBigBufferSize = 0x100;  // More than enough.
+  uint8_t buffer[kBigBufferSize];
+  buffer[0] = Dlrr::kBlockType;
+  buffer[1] = 0;  // Reserved.
+  buffer[2] = 0;  // Most significant size byte.
+  for (uint8_t size = 3; size < 6; ++size) {
+    buffer[3] = size;
+    Dlrr dlrr;
+    // Parse should be successful only when size is multiple of 3.
+    EXPECT_EQ(size % 3 == 0, dlrr.Parse(buffer, static_cast<uint16_t>(size)));
+  }
+}
+
+TEST(RtcpPacketDlrrTest, FailsOnTooManySubBlocks) {
+  Dlrr dlrr;
+  for (size_t i = 1; i <= Dlrr::kMaxNumberOfDlrrItems; ++i) {
+    EXPECT_TRUE(dlrr.WithDlrrItem(kSsrc + i, kLastRR + i, kDelay + i));
+  }
+  EXPECT_FALSE(dlrr.WithDlrrItem(kSsrc, kLastRR, kDelay));
+}
+
+TEST(RtcpPacketDlrrTest, CreateAndParseMaxSubBlocks) {
+  const size_t kBufferSize = 0x1000;  // More than enough.
+  uint8_t buffer[kBufferSize];
+
+  // Create.
+  Dlrr dlrr;
+  for (size_t i = 1; i <= Dlrr::kMaxNumberOfDlrrItems; ++i) {
+    EXPECT_TRUE(dlrr.WithDlrrItem(kSsrc + i, kLastRR + i, kDelay + i));
+  }
+  size_t used_buffer_size = dlrr.BlockLength();
+  ASSERT_LE(used_buffer_size, kBufferSize);
+  dlrr.Create(buffer);
+
+  // Parse.
+  Dlrr parsed;
+  uint16_t block_length = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]);
+  EXPECT_EQ(used_buffer_size, (block_length + 1) * 4u);
+  EXPECT_TRUE(parsed.Parse(buffer, block_length));
+  EXPECT_TRUE(parsed.sub_blocks().size() == Dlrr::kMaxNumberOfDlrrItems);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
new file mode 100644
index 0000000000..030f9f81fa
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// Transmission Time Offsets in RTP Streams (RFC 5450).
+//
+//      0                   1                   2                   3
+//      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//     |                      inter-arrival jitter                     |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//     .                                                               .
+//     .                                                               .
+//     .                                                               .
+//     |                      inter-arrival jitter                     |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  If present, this RTCP packet must be placed after a receiver report
+//  (inside a compound RTCP packet), and MUST have the same value for RC
+//  (reception report count) as the receiver report.
+
+bool ExtendedJitterReport::Parse(const RtcpCommonHeader& header,
+                                 const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t jitters_count = header.count_or_format;
+  const size_t kJitterSizeBytes = 4u;
+
+  if (header.payload_size_bytes < jitters_count * kJitterSizeBytes) {
+    LOG(LS_WARNING) << "Packet is too small to contain all the jitter.";
+    return false;
+  }
+
+  inter_arrival_jitters_.resize(jitters_count);
+  for (size_t index = 0; index < jitters_count; ++index) {
+    inter_arrival_jitters_[index] =
+        ByteReader<uint32_t>::ReadBigEndian(&payload[index * kJitterSizeBytes]);
+  }
+
+  return true;
+}
+
+bool ExtendedJitterReport::WithJitter(uint32_t jitter) {
+  if (inter_arrival_jitters_.size() >= kMaxNumberOfJitters) {
+    LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
+    return false;
+  }
+  inter_arrival_jitters_.push_back(jitter);
+  return true;
+}
+
+bool ExtendedJitterReport::Create(
+    uint8_t* packet,
+    size_t* index,
+    size_t max_length,
+    RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  size_t length = inter_arrival_jitters_.size();
+  CreateHeader(length, kPacketType, length, packet, index);
+
+  for (uint32_t jitter : inter_arrival_jitters_) {
+    ByteWriter<uint32_t>::WriteBigEndian(packet + *index, jitter);
+    *index += sizeof(uint32_t);
+  }
+  // Sanity check.
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
new file mode 100644
index 0000000000..49de7be1a8
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
+
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class ExtendedJitterReport : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 195;
+
+  ExtendedJitterReport() : RtcpPacket() {}
+
+  virtual ~ExtendedJitterReport() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  bool WithJitter(uint32_t jitter);
+
+  size_t jitters_count() const { return inter_arrival_jitters_.size(); }
+  uint32_t jitter(size_t index) const {
+    RTC_DCHECK_LT(index, jitters_count());
+    return inter_arrival_jitters_[index];
+  }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const int kMaxNumberOfJitters = 0x1f;
+
+  size_t BlockLength() const override {
+    return kHeaderLength + 4 * inter_arrival_jitters_.size();
+  }
+
+  std::vector<uint32_t> inter_arrival_jitters_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ExtendedJitterReport);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
new file mode 100644
index 0000000000..09d7b6305f
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ExtendedJitterReport;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+class RtcpPacketExtendedJitterReportTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = ij.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  ExtendedJitterReport ij;
+  rtc::scoped_ptr<RawPacket> packet;
+  const ExtendedJitterReport& parsed() { return parsed_; }
+
+ private:
+  ExtendedJitterReport parsed_;
+};
+
+TEST_F(RtcpPacketExtendedJitterReportTest, NoItem) {
+  // No initialization because packet is empty.
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(0u, parsed().jitters_count());
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, OneItem) {
+  EXPECT_TRUE(ij.WithJitter(0x11121314));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(1u, parsed().jitters_count());
+  EXPECT_EQ(0x11121314U, parsed().jitter(0));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, TwoItems) {
+  EXPECT_TRUE(ij.WithJitter(0x11121418));
+  EXPECT_TRUE(ij.WithJitter(0x22242628));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(2u, parsed().jitters_count());
+  EXPECT_EQ(0x11121418U, parsed().jitter(0));
+  EXPECT_EQ(0x22242628U, parsed().jitter(1));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, TooManyItems) {
+  const int kMaxIjItems = (1 << 5) - 1;
+  for (int i = 0; i < kMaxIjItems; ++i) {
+    EXPECT_TRUE(ij.WithJitter(i));
+  }
+  EXPECT_FALSE(ij.WithJitter(kMaxIjItems));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, ParseFailWithTooManyItems) {
+  ij.WithJitter(0x11121418);
+  BuildPacket();
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header);
+  header.count_or_format++;  // Damage package.
+
+  ExtendedJitterReport parsed;
+
+  EXPECT_FALSE(parsed.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
new file mode 100644
index 0000000000..8b9b354a06
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+//
+// Generic NACK (RFC 4585).
+//
+// FCI:
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |            PID                |             BLP               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool Nack::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+  RTC_DCHECK(header.count_or_format == kFeedbackMessageType);
+
+  if (header.payload_size_bytes < kCommonFeedbackLength + kNackItemLength) {
+    LOG(LS_WARNING) << "Payload length " << header.payload_size_bytes
+                    << " is too small for a Nack.";
+    return false;
+  }
+  size_t nack_items =
+      (header.payload_size_bytes - kCommonFeedbackLength) / kNackItemLength;
+
+  ParseCommonFeedback(payload);
+  const uint8_t* next_nack = payload + kCommonFeedbackLength;
+
+  packet_ids_.clear();
+  packed_.resize(nack_items);
+  for (size_t index = 0; index < nack_items; ++index) {
+    packed_[index].first_pid = ByteReader<uint16_t>::ReadBigEndian(next_nack);
+    packed_[index].bitmask = ByteReader<uint16_t>::ReadBigEndian(next_nack + 2);
+    next_nack += kNackItemLength;
+  }
+  Unpack();
+
+  return true;
+}
+
+bool Nack::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  RtcpPacket::PacketReadyCallback* callback) const {
+  RTC_DCHECK(!packed_.empty());
+  // If nack list can't fit in packet, try to fragment.
+  size_t nack_index = 0;
+  const size_t kCommonFbFmtLength = kHeaderLength + kCommonFeedbackLength;
+  do {
+    size_t bytes_left_in_buffer = max_length - *index;
+    if (bytes_left_in_buffer < kCommonFbFmtLength + kNackItemLength) {
+      if (!OnBufferFull(packet, index, callback))
+        return false;
+      continue;
+    }
+    size_t num_nack_fields =
+        std::min((bytes_left_in_buffer - kCommonFbFmtLength) / kNackItemLength,
+                 packed_.size() - nack_index);
+
+    size_t size_bytes =
+        (num_nack_fields * kNackItemLength) + kCommonFbFmtLength;
+    size_t header_length = ((size_bytes + 3) / 4) - 1;  // As 32bit words - 1
+    CreateHeader(kFeedbackMessageType, kPacketType, header_length, packet,
+                 index);
+    CreateCommonFeedback(packet + *index);
+    *index += kCommonFeedbackLength;
+    size_t end_index = nack_index + num_nack_fields;
+    for (; nack_index < end_index; ++nack_index) {
+      const auto& item = packed_[nack_index];
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 0, item.first_pid);
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 2, item.bitmask);
+      *index += kNackItemLength;
+    }
+    RTC_DCHECK_LE(*index, max_length);
+  } while (nack_index < packed_.size());
+
+  return true;
+}
+
+size_t Nack::BlockLength() const {
+  return (packed_.size() * kNackItemLength) + kCommonFeedbackLength +
+         kHeaderLength;
+}
+
+void Nack::WithList(const uint16_t* nack_list, size_t length) {
+  RTC_DCHECK(nack_list);
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  packet_ids_.assign(nack_list, nack_list + length);
+  Pack();
+}
+
+void Nack::Pack() {
+  RTC_DCHECK(!packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  auto it = packet_ids_.begin();
+  const auto end = packet_ids_.end();
+  while (it != end) {
+    PackedNack item;
+    item.first_pid = *it++;
+    // Bitmask specifies losses in any of the 16 packets following the pid.
+    item.bitmask = 0;
+    while (it != end) {
+      uint16_t shift = static_cast<uint16_t>(*it - item.first_pid - 1);
+      if (shift <= 15) {
+        item.bitmask |= (1 << shift);
+        ++it;
+      } else {
+        break;
+      }
+    }
+    packed_.push_back(item);
+  }
+}
+
+void Nack::Unpack() {
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(!packed_.empty());
+  for (const PackedNack& item : packed_) {
+    packet_ids_.push_back(item.first_pid);
+    uint16_t pid = item.first_pid + 1;
+    for (uint16_t bitmask = item.bitmask; bitmask != 0; bitmask >>= 1, ++pid)
+      if (bitmask & 1)
+        packet_ids_.push_back(pid);
+  }
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
new file mode 100644
index 0000000000..fb2be113a2
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Nack : public Rtpfb {
+ public:
+  const uint8_t kFeedbackMessageType = 1;
+  Nack() {}
+
+  virtual ~Nack() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void WithList(const uint16_t* nack_list, size_t length);
+  const std::vector<uint16_t>& packet_ids() const { return packet_ids_; }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+  size_t BlockLength() const override;
+
+ private:
+  const size_t kNackItemLength = 4;
+  struct PackedNack {
+    uint16_t first_pid;
+    uint16_t bitmask;
+  };
+
+  void Pack();    // Fills packed_ using packed_ids_. (used in WithList).
+  void Unpack();  // Fills packet_ids_ using packed_. (used in Parse).
+
+  std::vector<PackedNack> packed_;
+  std::vector<uint16_t> packet_ids_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Nack);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
new file mode 100644
index 0000000000..01e30f5644
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using ::testing::_;
+using ::testing::ElementsAreArray;
+using ::testing::Invoke;
+using ::testing::UnorderedElementsAreArray;
+
+using webrtc::rtcp::Nack;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+const uint16_t kList[] = {0, 1, 3, 8, 16};
+const size_t kListLength = sizeof(kList) / sizeof(kList[0]);
+const uint8_t kPacket[] = {0x81, 205,  0x00, 0x03, 0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89, 0x00, 0x00, 0x80, 0x85};
+const size_t kPacketLength = sizeof(kPacket);
+
+const uint16_t kWrapList[] = {0xffdc, 0xffec, 0xfffe, 0xffff, 0x0000,
+                              0x0001, 0x0003, 0x0014, 0x0064};
+const size_t kWrapListLength = sizeof(kWrapList) / sizeof(kWrapList[0]);
+const uint8_t kWrapPacket[] = {0x81, 205,  0x00, 0x06, 0x12, 0x34, 0x56, 0x78,
+                               0x23, 0x45, 0x67, 0x89, 0xff, 0xdc, 0x80, 0x00,
+                               0xff, 0xfe, 0x00, 0x17, 0x00, 0x14, 0x00, 0x00,
+                               0x00, 0x64, 0x00, 0x00};
+const size_t kWrapPacketLength = sizeof(kWrapPacket);
+
+TEST(RtcpPacketNackTest, Create) {
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kList, kListLength);
+
+  rtc::scoped_ptr<RawPacket> packet = nack.Build();
+
+  EXPECT_EQ(kPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kPacket, packet->Buffer(), kPacketLength));
+}
+
+TEST(RtcpPacketNackTest, Parse) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  EXPECT_EQ(kPacketLength, header.BlockSize());
+  Nack parsed;
+
+  EXPECT_TRUE(
+      parsed.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+  const Nack& const_parsed = parsed;
+
+  EXPECT_EQ(kSenderSsrc, const_parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, const_parsed.media_ssrc());
+  EXPECT_THAT(const_parsed.packet_ids(), ElementsAreArray(kList));
+}
+
+TEST(RtcpPacketNackTest, CreateWrap) {
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kWrapList, kWrapListLength);
+
+  rtc::scoped_ptr<RawPacket> packet = nack.Build();
+
+  EXPECT_EQ(kWrapPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kWrapPacket, packet->Buffer(), kWrapPacketLength));
+}
+
+TEST(RtcpPacketNackTest, ParseWrap) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kWrapPacket, kWrapPacketLength, &header));
+  EXPECT_EQ(kWrapPacketLength, header.BlockSize());
+
+  Nack parsed;
+  EXPECT_TRUE(
+      parsed.Parse(header, kWrapPacket + RtcpCommonHeader::kHeaderSizeBytes));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_THAT(parsed.packet_ids(), ElementsAreArray(kWrapList));
+}
+
+TEST(RtcpPacketNackTest, BadOrder) {
+  // Does not guarantee optimal packing, but should guarantee correctness.
+  const uint16_t kUnorderedList[] = {1, 25, 13, 12, 9, 27, 29};
+  const size_t kUnorderedListLength =
+      sizeof(kUnorderedList) / sizeof(kUnorderedList[0]);
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kUnorderedList, kUnorderedListLength);
+
+  rtc::scoped_ptr<RawPacket> packet = nack.Build();
+
+  Nack parsed;
+  RtcpCommonHeader header;
+  EXPECT_TRUE(
+      RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+  EXPECT_TRUE(parsed.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_THAT(parsed.packet_ids(), UnorderedElementsAreArray(kUnorderedList));
+}
+
+TEST(RtcpPacketNackTest, CreateFragmented) {
+  Nack nack;
+  const uint16_t kList[] = {1, 100, 200, 300, 400};
+  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kList, kListLength);
+
+  class MockPacketReadyCallback : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    MOCK_METHOD2(OnPacketReady, void(uint8_t*, size_t));
+  } verifier;
+
+  class NackVerifier {
+   public:
+    explicit NackVerifier(std::vector<uint16_t> ids) : ids_(ids) {}
+    void operator()(uint8_t* data, size_t length) {
+      RtcpCommonHeader header;
+      EXPECT_TRUE(RtcpParseCommonHeader(data, length, &header));
+      EXPECT_EQ(length, header.BlockSize());
+      Nack nack;
+      EXPECT_TRUE(
+          nack.Parse(header, data + RtcpCommonHeader::kHeaderSizeBytes));
+      EXPECT_EQ(kSenderSsrc, nack.sender_ssrc());
+      EXPECT_EQ(kRemoteSsrc, nack.media_ssrc());
+      EXPECT_THAT(nack.packet_ids(), ElementsAreArray(ids_));
+    }
+    std::vector<uint16_t> ids_;
+  } packet1({1, 100, 200}), packet2({300, 400});
+
+  EXPECT_CALL(verifier, OnPacketReady(_, _))
+      .WillOnce(Invoke(packet1))
+      .WillOnce(Invoke(packet2));
+  const size_t kBufferSize = 12 + (3 * 4);  // Fits common header + 3 nack items
+  uint8_t buffer[kBufferSize];
+  EXPECT_TRUE(nack.BuildExternalBuffer(buffer, kBufferSize, &verifier));
+}
+
+TEST(RtcpPacketNackTest, CreateFailsWithTooSmallBuffer) {
+  const uint16_t kList[] = {1};
+  const size_t kMinNackBlockSize = 16;
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kList, 1);
+  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      ADD_FAILURE() << "Buffer should be too small.";
+    }
+  } verifier;
+  uint8_t buffer[kMinNackBlockSize - 1];
+  EXPECT_FALSE(
+      nack.BuildExternalBuffer(buffer, kMinNackBlockSize - 1, &verifier));
+}
+
+TEST(RtcpPacketNackTest, ParseFailsWithTooSmallBuffer) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  header.payload_size_bytes--;  // Damage the packet
+  Nack parsed;
+  EXPECT_FALSE(
+      parsed.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc
new file mode 100644
index 0000000000..3673491058
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+
+//
+// Picture loss indication (PLI) (RFC 4585).
+// FCI: no feedback control information.
+bool Pli::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+  RTC_DCHECK(header.count_or_format == kFeedbackMessageType);
+
+  if (header.payload_size_bytes < kCommonFeedbackLength) {
+    LOG(LS_WARNING) << "Packet is too small to be a valid PLI packet";
+    return false;
+  }
+
+  ParseCommonFeedback(payload);
+  return true;
+}
+
+bool Pli::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
new file mode 100644
index 0000000000..5567825830
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Picture loss indication (PLI) (RFC 4585).
+class Pli : public Psfb {
+ public:
+  static const uint8_t kFeedbackMessageType = 1;
+
+  Pli() {}
+  virtual ~Pli() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override {
+    return kHeaderLength + kCommonFeedbackLength;
+  }
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Pli);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
new file mode 100644
index 0000000000..1c47c3ffb1
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using webrtc::rtcp::Pli;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+// Manually created Pli packet matching constants above.
+const uint8_t kPacket[] = {0x81, 206,  0x00, 0x02,
+                           0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89};
+const size_t kPacketLength = sizeof(kPacket);
+
+TEST(RtcpPacketPliTest, Parse) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  Pli mutable_parsed;
+  EXPECT_TRUE(mutable_parsed.Parse(
+      header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+  const Pli& parsed = mutable_parsed;  // Read values from constant object.
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+}
+
+TEST(RtcpPacketPliTest, Create) {
+  Pli pli;
+  pli.From(kSenderSsrc);
+  pli.To(kRemoteSsrc);
+
+  rtc::scoped_ptr<RawPacket> packet(pli.Build());
+
+  ASSERT_EQ(kPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kPacket, packet->Buffer(), kPacketLength));
+}
+
+TEST(RtcpPacketPliTest, ParseFailsOnTooSmallPacket) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  header.payload_size_bytes--;
+
+  Pli parsed;
+  EXPECT_FALSE(
+      parsed.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc
new file mode 100644
index 0000000000..d1ee401dab
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+
+void Psfb::ParseCommonFeedback(const uint8_t* payload) {
+  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
+  media_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+}
+
+void Psfb::CreateCommonFeedback(uint8_t* payload) const {
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[0], sender_ssrc_);
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[4], media_ssrc_);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
new file mode 100644
index 0000000000..dddcdecba6
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// PSFB: Payload-specific feedback message.
+// RFC 4585, Section 6.3.
+class Psfb : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 206;
+
+  Psfb() : sender_ssrc_(0), media_ssrc_(0) {}
+  virtual ~Psfb() {}
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  void To(uint32_t ssrc) { media_ssrc_ = ssrc; }
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  uint32_t media_ssrc() const { return media_ssrc_; }
+
+ protected:
+  static const size_t kCommonFeedbackLength = 8;
+  void ParseCommonFeedback(const uint8_t* payload);
+  void CreateCommonFeedback(uint8_t* payload) const;
+
+ private:
+  uint32_t sender_ssrc_;
+  uint32_t media_ssrc_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
new file mode 100644
index 0000000000..ef64b4f51b
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+//
+// RTCP receiver report (RFC 3550).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|    RC   |   PT=RR=201   |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                     SSRC of packet sender                     |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                         report block(s)                       |
+//  |                            ....                               |
+bool ReceiverReport::Parse(const RTCPUtility::RtcpCommonHeader& header,
+                           const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t report_blocks_count = header.count_or_format;
+
+  if (header.payload_size_bytes <
+      kRrBaseLength + report_blocks_count * ReportBlock::kLength) {
+    LOG(LS_WARNING) << "Packet is too small to contain all the data.";
+    return false;
+  }
+
+  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(payload);
+
+  const uint8_t* next_report_block = payload + kRrBaseLength;
+
+  report_blocks_.resize(report_blocks_count);
+  for (ReportBlock& block : report_blocks_) {
+    block.Parse(next_report_block, ReportBlock::kLength);
+    next_report_block += ReportBlock::kLength;
+  }
+
+  RTC_DCHECK_LE(next_report_block, payload + header.payload_size_bytes);
+  return true;
+}
+
+bool ReceiverReport::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
+               index);
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, sender_ssrc_);
+  *index += kRrBaseLength;
+  for (const ReportBlock& block : report_blocks_) {
+    block.Create(packet + *index);
+    *index += ReportBlock::kLength;
+  }
+  return true;
+}
+
+bool ReceiverReport::WithReportBlock(const ReportBlock& block) {
+  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
+    LOG(LS_WARNING) << "Max report blocks reached.";
+    return false;
+  }
+  report_blocks_.push_back(block);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
new file mode 100644
index 0000000000..172a84ea2f
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class ReceiverReport : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 201;
+  ReceiverReport() : sender_ssrc_(0) {}
+
+  virtual ~ReceiverReport() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  bool WithReportBlock(const ReportBlock& block);
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  const std::vector<ReportBlock>& report_blocks() const {
+    return report_blocks_;
+  }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const size_t kRrBaseLength = 4;
+  static const size_t kMaxNumberOfReportBlocks = 0x1F;
+
+  size_t BlockLength() const {
+    return kHeaderLength + kRrBaseLength +
+           report_blocks_.size() * ReportBlock::kLength;
+  }
+
+  uint32_t sender_ssrc_;
+  std::vector<ReportBlock> report_blocks_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ReceiverReport);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
new file mode 100644
index 0000000000..ff3da600a5
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ReceiverReport;
+using webrtc::rtcp::ReportBlock;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+const uint32_t kCumulativeLost = 0x111213;
+const uint32_t kExtHighestSeqNum = 0x22232425;
+const uint32_t kJitter = 0x33343536;
+const uint32_t kLastSr = 0x44454647;
+const uint32_t kDelayLastSr = 0x55565758;
+// Manually created ReceiverReport with one ReportBlock matching constants
+// above.
+// Having this block allows to test Create and Parse separately.
+const uint8_t kPacket[] = {0x81, 201,  0x00, 0x07, 0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89, 55,   0x11, 0x12, 0x13,
+                           0x22, 0x23, 0x24, 0x25, 0x33, 0x34, 0x35, 0x36,
+                           0x44, 0x45, 0x46, 0x47, 0x55, 0x56, 0x57, 0x58};
+const size_t kPacketLength = sizeof(kPacket);
+
+class RtcpPacketReceiverReportTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = rr.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  ReceiverReport rr;
+  rtc::scoped_ptr<RawPacket> packet;
+  const ReceiverReport& parsed() { return parsed_; }
+
+ private:
+  ReceiverReport parsed_;
+};
+
+TEST_F(RtcpPacketReceiverReportTest, Parse) {
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(kPacket, kPacketLength, &header);
+  EXPECT_TRUE(rr.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+  const ReceiverReport& parsed = rr;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(1u, parsed.report_blocks().size());
+  const ReportBlock& rb = parsed.report_blocks().front();
+  EXPECT_EQ(kRemoteSsrc, rb.source_ssrc());
+  EXPECT_EQ(kFractionLost, rb.fraction_lost());
+  EXPECT_EQ(kCumulativeLost, rb.cumulative_lost());
+  EXPECT_EQ(kExtHighestSeqNum, rb.extended_high_seq_num());
+  EXPECT_EQ(kJitter, rb.jitter());
+  EXPECT_EQ(kLastSr, rb.last_sr());
+  EXPECT_EQ(kDelayLastSr, rb.delay_since_last_sr());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, ParseFailsOnIncorrectSize) {
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(kPacket, kPacketLength, &header);
+  header.count_or_format++;  // Damage the packet.
+  EXPECT_FALSE(rr.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+TEST_F(RtcpPacketReceiverReportTest, Create) {
+  rr.From(kSenderSsrc);
+  ReportBlock rb;
+  rb.To(kRemoteSsrc);
+  rb.WithFractionLost(kFractionLost);
+  rb.WithCumulativeLost(kCumulativeLost);
+  rb.WithExtHighestSeqNum(kExtHighestSeqNum);
+  rb.WithJitter(kJitter);
+  rb.WithLastSr(kLastSr);
+  rb.WithDelayLastSr(kDelayLastSr);
+  rr.WithReportBlock(rb);
+
+  BuildPacket();
+
+  ASSERT_EQ(kPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kPacket, packet->Buffer(), kPacketLength));
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithoutReportBlocks) {
+  rr.From(kSenderSsrc);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed().sender_ssrc());
+  EXPECT_EQ(0u, parsed().report_blocks().size());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithTwoReportBlocks) {
+  ReportBlock rb1;
+  rb1.To(kRemoteSsrc);
+  ReportBlock rb2;
+  rb2.To(kRemoteSsrc + 1);
+
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb1));
+  EXPECT_TRUE(rr.WithReportBlock(rb2));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed().sender_ssrc());
+  EXPECT_EQ(2u, parsed().report_blocks().size());
+  EXPECT_EQ(kRemoteSsrc, parsed().report_blocks()[0].source_ssrc());
+  EXPECT_EQ(kRemoteSsrc + 1, parsed().report_blocks()[1].source_ssrc());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithTooManyReportBlocks) {
+  rr.From(kSenderSsrc);
+  const size_t kMaxReportBlocks = (1 << 5) - 1;
+  ReportBlock rb;
+  for (size_t i = 0; i < kMaxReportBlocks; ++i) {
+    rb.To(kRemoteSsrc + i);
+    EXPECT_TRUE(rr.WithReportBlock(rb));
+  }
+  rb.To(kRemoteSsrc + kMaxReportBlocks);
+  EXPECT_FALSE(rr.WithReportBlock(rb));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
new file mode 100644
index 0000000000..4911dbf5b7
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTCP report block (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                 SSRC_1 (SSRC of first source)                 |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 | fraction lost |       cumulative number of packets lost       |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |           extended highest sequence number received           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                      interarrival jitter                      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |                         last SR (LSR)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |                   delay since last SR (DLSR)                  |
+// 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+ReportBlock::ReportBlock()
+    : source_ssrc_(0),
+      fraction_lost_(0),
+      cumulative_lost_(0),
+      extended_high_seq_num_(0),
+      jitter_(0),
+      last_sr_(0),
+      delay_since_last_sr_(0) {}
+
+bool ReportBlock::Parse(const uint8_t* buffer, size_t length) {
+  RTC_DCHECK(buffer != nullptr);
+  if (length < ReportBlock::kLength) {
+    LOG(LS_ERROR) << "Report Block should be 24 bytes long";
+    return false;
+  }
+
+  source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
+  fraction_lost_ = buffer[4];
+  cumulative_lost_ = ByteReader<uint32_t, 3>::ReadBigEndian(&buffer[5]);
+  extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
+  last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
+  delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]);
+
+  return true;
+}
+
+void ReportBlock::Create(uint8_t* buffer) const {
+  // Runtime check should be done while setting cumulative_lost.
+  RTC_DCHECK_LT(cumulative_lost(), (1u << 24));  // Have only 3 bytes for it.
+
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], source_ssrc());
+  ByteWriter<uint8_t>::WriteBigEndian(&buffer[4], fraction_lost());
+  ByteWriter<uint32_t, 3>::WriteBigEndian(&buffer[5], cumulative_lost());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], extended_high_seq_num());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[12], jitter());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[16], last_sr());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[20], delay_since_last_sr());
+}
+
+bool ReportBlock::WithCumulativeLost(uint32_t cumulative_lost) {
+  if (cumulative_lost >= (1u << 24)) {  // Have only 3 bytes to store it.
+    LOG(LS_WARNING) << "Cumulative lost is too big to fit into Report Block";
+    return false;
+  }
+  cumulative_lost_ = cumulative_lost;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
new file mode 100644
index 0000000000..ef99e17297
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+
+#include "webrtc/base/basictypes.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class ReportBlock {
+ public:
+  static const size_t kLength = 24;
+
+  ReportBlock();
+  ~ReportBlock() {}
+
+  bool Parse(const uint8_t* buffer, size_t length);
+
+  // Fills buffer with the ReportBlock.
+  // Consumes ReportBlock::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void To(uint32_t ssrc) { source_ssrc_ = ssrc; }
+  void WithFractionLost(uint8_t fraction_lost) {
+    fraction_lost_ = fraction_lost;
+  }
+  bool WithCumulativeLost(uint32_t cumulative_lost);
+  void WithExtHighestSeqNum(uint32_t ext_highest_seq_num) {
+    extended_high_seq_num_ = ext_highest_seq_num;
+  }
+  void WithJitter(uint32_t jitter) { jitter_ = jitter; }
+  void WithLastSr(uint32_t last_sr) { last_sr_ = last_sr; }
+  void WithDelayLastSr(uint32_t delay_last_sr) {
+    delay_since_last_sr_ = delay_last_sr;
+  }
+
+  uint32_t source_ssrc() const { return source_ssrc_; }
+  uint8_t fraction_lost() const { return fraction_lost_; }
+  uint32_t cumulative_lost() const { return cumulative_lost_; }
+  uint32_t extended_high_seq_num() const { return extended_high_seq_num_; }
+  uint32_t jitter() const { return jitter_; }
+  uint32_t last_sr() const { return last_sr_; }
+  uint32_t delay_since_last_sr() const { return delay_since_last_sr_; }
+
+ private:
+  uint32_t source_ssrc_;
+  uint8_t fraction_lost_;
+  uint32_t cumulative_lost_;
+  uint32_t extended_high_seq_num_;
+  uint32_t jitter_;
+  uint32_t last_sr_;
+  uint32_t delay_since_last_sr_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
new file mode 100644
index 0000000000..85bbb404a4
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+
+using webrtc::rtcp::ReportBlock;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+// Use values that are streamed differently LE and BE.
+const uint32_t kCumulativeLost = 0x111213;
+const uint32_t kExtHighestSeqNum = 0x22232425;
+const uint32_t kJitter = 0x33343536;
+const uint32_t kLastSr = 0x44454647;
+const uint32_t kDelayLastSr = 0x55565758;
+const size_t kBufferLength = ReportBlock::kLength;
+
+TEST(RtcpPacketReportBlockTest, ParseChecksLength) {
+  uint8_t buffer[kBufferLength];
+  memset(buffer, 0, sizeof(buffer));
+
+  ReportBlock rb;
+  EXPECT_FALSE(rb.Parse(buffer, kBufferLength - 1));
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseAnyData) {
+  uint8_t buffer[kBufferLength];
+  // Fill buffer with semi-random data.
+  Random generator(0x256F8A285EC829ull);
+  for (size_t i = 0; i < kBufferLength; ++i)
+    buffer[i] = static_cast<uint8_t>(generator.Rand(0, 0xff));
+
+  ReportBlock rb;
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseMatchCreate) {
+  ReportBlock rb;
+  rb.To(kRemoteSsrc);
+  rb.WithFractionLost(kFractionLost);
+  rb.WithCumulativeLost(kCumulativeLost);
+  rb.WithExtHighestSeqNum(kExtHighestSeqNum);
+  rb.WithJitter(kJitter);
+  rb.WithLastSr(kLastSr);
+  rb.WithDelayLastSr(kDelayLastSr);
+
+  uint8_t buffer[kBufferLength];
+  rb.Create(buffer);
+
+  ReportBlock parsed;
+  EXPECT_TRUE(parsed.Parse(buffer, kBufferLength));
+
+  EXPECT_EQ(kRemoteSsrc, parsed.source_ssrc());
+  EXPECT_EQ(kFractionLost, parsed.fraction_lost());
+  EXPECT_EQ(kCumulativeLost, parsed.cumulative_lost());
+  EXPECT_EQ(kExtHighestSeqNum, parsed.extended_high_seq_num());
+  EXPECT_EQ(kJitter, parsed.jitter());
+  EXPECT_EQ(kLastSr, parsed.last_sr());
+  EXPECT_EQ(kDelayLastSr, parsed.delay_since_last_sr());
+}
+
+TEST(RtcpPacketReportBlockTest, ValidateCumulativeLost) {
+  const uint32_t kMaxCumulativeLost = 0xffffff;
+  ReportBlock rb;
+  EXPECT_FALSE(rb.WithCumulativeLost(kMaxCumulativeLost + 1));
+  EXPECT_TRUE(rb.WithCumulativeLost(kMaxCumulativeLost));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc
new file mode 100644
index 0000000000..db4ae67326
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+// Receiver Reference Time Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=4      |   reserved    |       block length = 2        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |              NTP timestamp, most significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |             NTP timestamp, least significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void Rrtr::Parse(const uint8_t* buffer) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // reserved = buffer[1];
+  RTC_DCHECK(ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) == kBlockLength);
+  uint32_t seconds = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  uint32_t fraction = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  ntp_.Set(seconds, fraction);
+}
+
+void Rrtr::Create(uint8_t* buffer) const {
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockLength);
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ntp_.seconds());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], ntp_.fractions());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
new file mode 100644
index 0000000000..3354f61df6
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Rrtr {
+ public:
+  static const uint8_t kBlockType = 4;
+  static const uint16_t kBlockLength = 2;
+  static const size_t kLength = 4 * (kBlockLength + 1);  // 12
+
+  Rrtr() {}
+  Rrtr(const Rrtr&) = default;
+  ~Rrtr() {}
+
+  Rrtr& operator=(const Rrtr&) = default;
+
+  void Parse(const uint8_t* buffer);
+
+  // Fills buffer with the Rrtr.
+  // Consumes Rrtr::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void WithNtp(const NtpTime& ntp) { ntp_ = ntp; }
+
+  NtpTime ntp() const { return ntp_; }
+
+ private:
+  NtpTime ntp_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
new file mode 100644
index 0000000000..6536e06186
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+using webrtc::rtcp::Rrtr;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kNtpSec = 0x12345678;
+const uint32_t kNtpFrac = 0x23456789;
+const uint8_t kBlock[] = {0x04, 0x00, 0x00, 0x02,
+                          0x12, 0x34, 0x56, 0x78,
+                          0x23, 0x45, 0x67, 0x89};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+static_assert(
+    kBlockSizeBytes == Rrtr::kLength,
+    "Size of manually created Rrtr block should match class constant");
+
+TEST(RtcpPacketRrtrTest, Create) {
+  uint8_t buffer[Rrtr::kLength];
+  Rrtr rrtr;
+  rrtr.WithNtp(NtpTime(kNtpSec, kNtpFrac));
+
+  rrtr.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketRrtrTest, Parse) {
+  Rrtr read_rrtr;
+  read_rrtr.Parse(kBlock);
+
+  // Run checks on const object to ensure all accessors have const modifier.
+  const Rrtr& parsed = read_rrtr;
+
+  EXPECT_EQ(kNtpSec, parsed.ntp().seconds());
+  EXPECT_EQ(kNtpFrac, parsed.ntp().fractions());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc
new file mode 100644
index 0000000000..b5571d45a3
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585, Section 6.1: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+
+void Rtpfb::ParseCommonFeedback(const uint8_t* payload) {
+  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
+  media_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+}
+
+void Rtpfb::CreateCommonFeedback(uint8_t* payload) const {
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[0], sender_ssrc_);
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[4], media_ssrc_);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
new file mode 100644
index 0000000000..801aa085c4
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// RTPFB: Transport layer feedback message.
+// RFC4585, Section 6.2
+class Rtpfb : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 205;
+
+  Rtpfb() : sender_ssrc_(0), media_ssrc_(0) {}
+  virtual ~Rtpfb() {}
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  void To(uint32_t ssrc) { media_ssrc_ = ssrc; }
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  uint32_t media_ssrc() const { return media_ssrc_; }
+
+ protected:
+  static const size_t kCommonFeedbackLength = 8;
+  void ParseCommonFeedback(const uint8_t* payload);
+  void CreateCommonFeedback(uint8_t* payload) const;
+
+ private:
+  uint32_t sender_ssrc_;
+  uint32_t media_ssrc_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.cc
new file mode 100644
index 0000000000..829f3a9db9
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.cc
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+//
+// Slice loss indication (SLI) (RFC 4585).
+// FCI:
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |            First        |        Number           | PictureID |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Sli::Macroblocks::Macroblocks(uint8_t picture_id,
+                              uint16_t first,
+                              uint16_t number) {
+  RTC_DCHECK_LE(first, 0x1fff);
+  RTC_DCHECK_LE(number, 0x1fff);
+  RTC_DCHECK_LE(picture_id, 0x3f);
+  item_ = (first << 19) | (number << 6) | picture_id;
+}
+
+void Sli::Macroblocks::Parse(const uint8_t* buffer) {
+  item_ = ByteReader<uint32_t>::ReadBigEndian(buffer);
+}
+
+void Sli::Macroblocks::Create(uint8_t* buffer) const {
+  ByteWriter<uint32_t>::WriteBigEndian(buffer, item_);
+}
+
+bool Sli::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+  RTC_DCHECK(header.count_or_format == kFeedbackMessageType);
+
+  if (header.payload_size_bytes <
+      kCommonFeedbackLength + Macroblocks::kLength) {
+    LOG(LS_WARNING) << "Packet is too small to be a valid SLI packet";
+    return false;
+  }
+
+  size_t number_of_items =
+      (header.payload_size_bytes - kCommonFeedbackLength) /
+      Macroblocks::kLength;
+
+  ParseCommonFeedback(payload);
+  items_.resize(number_of_items);
+
+  const uint8_t* next_item = payload + kCommonFeedbackLength;
+  for (Macroblocks& item : items_) {
+    item.Parse(next_item);
+    next_item += Macroblocks::kLength;
+  }
+
+  return true;
+}
+
+bool Sli::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  RTC_DCHECK(!items_.empty());
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  for (const Macroblocks& item : items_) {
+    item.Create(packet + *index);
+    *index += Macroblocks::kLength;
+  }
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h
new file mode 100644
index 0000000000..5d9e6c93e9
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SLI_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SLI_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Slice loss indication (SLI) (RFC 4585).
+class Sli : public Psfb {
+ public:
+  static const uint8_t kFeedbackMessageType = 2;
+  class Macroblocks {
+   public:
+    static const size_t kLength = 4;
+    Macroblocks() : item_(0) {}
+    Macroblocks(uint8_t picture_id, uint16_t first, uint16_t number);
+    ~Macroblocks() {}
+
+    void Parse(const uint8_t* buffer);
+    void Create(uint8_t* buffer) const;
+
+    uint16_t first() const { return item_ >> 19; }
+    uint16_t number() const { return (item_ >> 6) & 0x1fff; }
+    uint8_t picture_id() const { return (item_ & 0x3f); }
+
+   private:
+    uint32_t item_;
+  };
+
+  Sli() {}
+  virtual ~Sli() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void WithPictureId(uint8_t picture_id,
+                     uint16_t first_macroblock = 0,
+                     uint16_t number_macroblocks = 0x1fff) {
+    items_.push_back(
+        Macroblocks(picture_id, first_macroblock, number_macroblocks));
+  }
+
+  const std::vector<Macroblocks>& macroblocks() const { return items_; }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override {
+    return RtcpPacket::kHeaderLength + Psfb::kCommonFeedbackLength +
+           items_.size() * Macroblocks::kLength;
+  }
+
+  std::vector<Macroblocks> items_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Sli);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SLI_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli_unittest.cc
new file mode 100644
index 0000000000..c2be16846b
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli_unittest.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using testing::ElementsAreArray;
+using testing::make_tuple;
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::Sli;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+const uint8_t kPictureId = 0x3f;
+const uint16_t kFirstMb = 0x1e61;
+const uint16_t kNumberOfMb = 0x1a0a;
+const uint32_t kSliItem = (static_cast<uint32_t>(kFirstMb) << 19) |
+                          (static_cast<uint32_t>(kNumberOfMb) << 6) |
+                           static_cast<uint32_t>(kPictureId);
+
+// Manually created Sli packet matching constants above.
+const uint8_t kPacket[] = {0x82,  206, 0x00, 0x03,
+                           0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89,
+                           (kSliItem >> 24) & 0xff,
+                           (kSliItem >> 16) & 0xff,
+                           (kSliItem >> 8) & 0xff,
+                           kSliItem & 0xff};
+const size_t kPacketLength = sizeof(kPacket);
+
+bool ParseSli(const uint8_t* buffer, size_t length, Sli* sli) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(buffer, length, &header));
+  EXPECT_EQ(length, header.BlockSize());
+  return sli->Parse(header, buffer + RtcpCommonHeader::kHeaderSizeBytes);
+}
+
+TEST(RtcpPacketSliTest, Create) {
+  Sli sli;
+  sli.From(kSenderSsrc);
+  sli.To(kRemoteSsrc);
+  sli.WithPictureId(kPictureId, kFirstMb, kNumberOfMb);
+
+  rtc::scoped_ptr<RawPacket> packet(sli.Build());
+
+  EXPECT_THAT(make_tuple(packet->Buffer(), packet->Length()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketSliTest, Parse) {
+  Sli mutable_parsed;
+  EXPECT_TRUE(ParseSli(kPacket, kPacketLength, &mutable_parsed));
+  const Sli& parsed = mutable_parsed;  // Read values from constant object.
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_EQ(1u, parsed.macroblocks().size());
+  EXPECT_EQ(kFirstMb, parsed.macroblocks()[0].first());
+  EXPECT_EQ(kNumberOfMb, parsed.macroblocks()[0].number());
+  EXPECT_EQ(kPictureId, parsed.macroblocks()[0].picture_id());
+}
+
+TEST(RtcpPacketSliTest, ParseFailsOnTooSmallPacket) {
+  Sli sli;
+  sli.From(kSenderSsrc);
+  sli.To(kRemoteSsrc);
+  sli.WithPictureId(kPictureId, kFirstMb, kNumberOfMb);
+
+  rtc::scoped_ptr<RawPacket> packet(sli.Build());
+  packet->MutableBuffer()[3]--;  // Decrease size by 1 word (4 bytes).
+
+  EXPECT_FALSE(ParseSli(packet->Buffer(), packet->Length() - 4, &sli));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
new file mode 100644
index 0000000000..fd0219cf82
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::PT_RTPFB;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBN;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBRItem;
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+const uint32_t kUnusedMediaSourceSsrc0 = 0;
+void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) {
+  buffer[(*offset)++] = value;
+}
+void AssignUWord32(uint8_t* buffer, size_t* offset, uint32_t value) {
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + *offset, value);
+  *offset += 4;
+}
+
+void ComputeMantissaAnd6bitBase2Exponent(uint32_t input_base10,
+                                         uint8_t bits_mantissa,
+                                         uint32_t* mantissa,
+                                         uint8_t* exp) {
+  // input_base10 = mantissa * 2^exp
+  assert(bits_mantissa <= 32);
+  uint32_t mantissa_max = (1 << bits_mantissa) - 1;
+  uint8_t exponent = 0;
+  for (uint32_t i = 0; i < 64; ++i) {
+    if (input_base10 <= (mantissa_max << i)) {
+      exponent = i;
+      break;
+    }
+  }
+  *exp = exponent;
+  *mantissa = (input_base10 >> exponent);
+}
+
+void CreateTmmbrItem(const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
+                     uint8_t* buffer,
+                     size_t* pos) {
+  uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000;
+  uint32_t mantissa = 0;
+  uint8_t exp = 0;
+  ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
+
+  AssignUWord32(buffer, pos, tmmbr_item.SSRC);
+  AssignUWord8(buffer, pos, (exp << 2) + ((mantissa >> 15) & 0x03));
+  AssignUWord8(buffer, pos, mantissa >> 7);
+  AssignUWord8(buffer, pos, (mantissa << 1) +
+                            ((tmmbr_item.MeasuredOverhead >> 8) & 0x01));
+  AssignUWord8(buffer, pos, tmmbr_item.MeasuredOverhead);
+}
+
+// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
+//
+// FCI:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |                              SSRC                             |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void CreateTmmbn(const RTCPPacketRTPFBTMMBN& tmmbn,
+                 const std::vector<RTCPPacketRTPFBTMMBRItem>& tmmbn_items,
+                 uint8_t* buffer,
+                 size_t* pos) {
+  AssignUWord32(buffer, pos, tmmbn.SenderSSRC);
+  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
+  for (uint8_t i = 0; i < tmmbn_items.size(); ++i) {
+    CreateTmmbrItem(tmmbn_items[i], buffer, pos);
+  }
+}
+}  // namespace
+
+bool Tmmbn::WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead) {
+  assert(overhead <= 0x1ff);
+  if (tmmbn_items_.size() >= kMaxNumberOfTmmbrs) {
+    LOG(LS_WARNING) << "Max TMMBN size reached.";
+    return false;
+  }
+  RTCPPacketRTPFBTMMBRItem tmmbn_item;
+  tmmbn_item.SSRC = ssrc;
+  tmmbn_item.MaxTotalMediaBitRate = bitrate_kbps;
+  tmmbn_item.MeasuredOverhead = overhead;
+  tmmbn_items_.push_back(tmmbn_item);
+  return true;
+}
+
+bool Tmmbn::Create(uint8_t* packet,
+                   size_t* index,
+                   size_t max_length,
+                   RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const uint8_t kFmt = 4;
+  CreateHeader(kFmt, PT_RTPFB, HeaderLength(), packet, index);
+  CreateTmmbn(tmmbn_, tmmbn_items_, packet, index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
new file mode 100644
index 0000000000..82bf9dd9e9
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
+
+#include <vector>
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
+class Tmmbn : public RtcpPacket {
+ public:
+  Tmmbn() : RtcpPacket() {
+    memset(&tmmbn_, 0, sizeof(tmmbn_));
+  }
+
+  virtual ~Tmmbn() {}
+
+  void From(uint32_t ssrc) {
+    tmmbn_.SenderSSRC = ssrc;
+  }
+  // Max 50 TMMBR can be added per TMMBN.
+  bool WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead);
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const int kMaxNumberOfTmmbrs = 50;
+
+  size_t BlockLength() const {
+    const size_t kFciLen = 8;
+    return kCommonFbFmtLength + kFciLen * tmmbn_items_.size();
+  }
+
+  RTCPUtility::RTCPPacketRTPFBTMMBN tmmbn_;
+  std::vector<RTCPUtility::RTCPPacketRTPFBTMMBRItem> tmmbn_items_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Tmmbn);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
new file mode 100644
index 0000000000..32d64a97b4
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/test/rtcp_packet_parser.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::Tmmbn;
+using webrtc::test::RtcpPacketParser;
+
+namespace webrtc {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+TEST(RtcpPacketTest, TmmbnWithNoItem) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
+  EXPECT_EQ(0, parser.tmmbn_items()->num_packets());
+}
+
+TEST(RtcpPacketTest, TmmbnWithOneItem) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc, 312, 60));
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
+  EXPECT_EQ(1, parser.tmmbn_items()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
+  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
+  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
+}
+
+TEST(RtcpPacketTest, TmmbnWithTwoItems) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc, 312, 60));
+  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc + 1, 1288, 40));
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
+  EXPECT_EQ(2, parser.tmmbn_items()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
+  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
+  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
+  EXPECT_EQ(kRemoteSsrc + 1, parser.tmmbn_items()->Ssrc(1));
+  EXPECT_EQ(1288U, parser.tmmbn_items()->BitrateKbps(1));
+  EXPECT_EQ(40U, parser.tmmbn_items()->Overhead(1));
+}
+
+TEST(RtcpPacketTest, TmmbnWithTooManyItems) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+  const int kMaxTmmbrItems = 50;
+  for (int i = 0; i < kMaxTmmbrItems; ++i)
+    EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc + i, 312, 60));
+
+  EXPECT_FALSE(tmmbn.WithTmmbr(kRemoteSsrc + kMaxTmmbrItems, 312, 60));
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
new file mode 100644
index 0000000000..4df167de79
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::PT_RTPFB;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBR;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBRItem;
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+const uint32_t kUnusedMediaSourceSsrc0 = 0;
+
+void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) {
+  buffer[(*offset)++] = value;
+}
+
+void AssignUWord32(uint8_t* buffer, size_t* offset, uint32_t value) {
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + *offset, value);
+  *offset += 4;
+}
+
+void ComputeMantissaAnd6bitBase2Exponent(uint32_t input_base10,
+                                         uint8_t bits_mantissa,
+                                         uint32_t* mantissa,
+                                         uint8_t* exp) {
+  // input_base10 = mantissa * 2^exp
+  assert(bits_mantissa <= 32);
+  uint32_t mantissa_max = (1 << bits_mantissa) - 1;
+  uint8_t exponent = 0;
+  for (uint32_t i = 0; i < 64; ++i) {
+    if (input_base10 <= (mantissa_max << i)) {
+      exponent = i;
+      break;
+    }
+  }
+  *exp = exponent;
+  *mantissa = (input_base10 >> exponent);
+}
+
+void CreateTmmbrItem(const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
+                     uint8_t* buffer,
+                     size_t* pos) {
+  uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000;
+  uint32_t mantissa = 0;
+  uint8_t exp = 0;
+  ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
+
+  AssignUWord32(buffer, pos, tmmbr_item.SSRC);
+  AssignUWord8(buffer, pos, (exp << 2) + ((mantissa >> 15) & 0x03));
+  AssignUWord8(buffer, pos, mantissa >> 7);
+  AssignUWord8(buffer, pos, (mantissa << 1) +
+                            ((tmmbr_item.MeasuredOverhead >> 8) & 0x01));
+  AssignUWord8(buffer, pos, tmmbr_item.MeasuredOverhead);
+}
+
+// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
+//
+// FCI:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |                              SSRC                             |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void CreateTmmbr(const RTCPPacketRTPFBTMMBR& tmmbr,
+                 const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
+                 uint8_t* buffer,
+                 size_t* pos) {
+  AssignUWord32(buffer, pos, tmmbr.SenderSSRC);
+  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
+  CreateTmmbrItem(tmmbr_item, buffer, pos);
+}
+}  // namespace
+
+bool Tmmbr::Create(uint8_t* packet,
+                   size_t* index,
+                   size_t max_length,
+                   RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const uint8_t kFmt = 3;
+  CreateHeader(kFmt, PT_RTPFB, HeaderLength(), packet, index);
+  CreateTmmbr(tmmbr_, tmmbr_item_, packet, index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
new file mode 100644
index 0000000000..84a4180ad3
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
+class Tmmbr : public RtcpPacket {
+ public:
+  Tmmbr() : RtcpPacket() {
+    memset(&tmmbr_, 0, sizeof(tmmbr_));
+    memset(&tmmbr_item_, 0, sizeof(tmmbr_item_));
+  }
+
+  virtual ~Tmmbr() {}
+
+  void From(uint32_t ssrc) {
+    tmmbr_.SenderSSRC = ssrc;
+  }
+  void To(uint32_t ssrc) {
+    tmmbr_item_.SSRC = ssrc;
+  }
+  void WithBitrateKbps(uint32_t bitrate_kbps) {
+    tmmbr_item_.MaxTotalMediaBitRate = bitrate_kbps;
+  }
+  void WithOverhead(uint16_t overhead) {
+    assert(overhead <= 0x1ff);
+    tmmbr_item_.MeasuredOverhead = overhead;
+  }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const {
+    const size_t kFciLen = 8;
+    return kCommonFbFmtLength + kFciLen;
+  }
+
+  RTCPUtility::RTCPPacketRTPFBTMMBR tmmbr_;
+  RTCPUtility::RTCPPacketRTPFBTMMBRItem tmmbr_item_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Tmmbr);
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
new file mode 100644
index 0000000000..6d71caa251
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/test/rtcp_packet_parser.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::Tmmbr;
+using webrtc::test::RtcpPacketParser;
+
+namespace webrtc {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+TEST(RtcpPacketTest, Tmmbr) {
+  Tmmbr tmmbr;
+  tmmbr.From(kSenderSsrc);
+  tmmbr.To(kRemoteSsrc);
+  tmmbr.WithBitrateKbps(312);
+  tmmbr.WithOverhead(60);
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbr.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbr()->Ssrc());
+  EXPECT_EQ(1, parser.tmmbr_item()->num_packets());
+  EXPECT_EQ(312U, parser.tmmbr_item()->BitrateKbps());
+  EXPECT_EQ(60U, parser.tmmbr_item()->Overhead());
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
index 4cc1f38479..ad6fd166f2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.cc
new file mode 100644
index 0000000000..a79d48e1ca
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+// VoIP Metrics Report Block (RFC 3611).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |     BT=7      |   reserved    |       block length = 8        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                        SSRC of source                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |   loss rate   | discard rate  | burst density |  gap density  |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |       burst duration          |         gap duration          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |     round trip delay          |       end system delay        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 | signal level  |  noise level  |     RERL      |     Gmin      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 24 |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 28 |   RX config   |   reserved    |          JB nominal           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 32 |          JB maximum           |          JB abs max           |
+// 36 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+VoipMetric::VoipMetric() : ssrc_(0) {
+  memset(&voip_metric_, 0, sizeof(voip_metric_));
+}
+
+void VoipMetric::Parse(const uint8_t* buffer) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // reserved = buffer[1];
+  RTC_DCHECK(ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) == kBlockLength);
+  ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  voip_metric_.lossRate = buffer[8];
+  voip_metric_.discardRate = buffer[9];
+  voip_metric_.burstDensity = buffer[10];
+  voip_metric_.gapDensity = buffer[11];
+  voip_metric_.burstDuration = ByteReader<uint16_t>::ReadBigEndian(&buffer[12]);
+  voip_metric_.gapDuration = ByteReader<uint16_t>::ReadBigEndian(&buffer[14]);
+  voip_metric_.roundTripDelay =
+      ByteReader<uint16_t>::ReadBigEndian(&buffer[16]);
+  voip_metric_.endSystemDelay =
+      ByteReader<uint16_t>::ReadBigEndian(&buffer[18]);
+  voip_metric_.signalLevel = buffer[20];
+  voip_metric_.noiseLevel = buffer[21];
+  voip_metric_.RERL = buffer[22];
+  voip_metric_.Gmin = buffer[23];
+  voip_metric_.Rfactor = buffer[24];
+  voip_metric_.extRfactor = buffer[25];
+  voip_metric_.MOSLQ = buffer[26];
+  voip_metric_.MOSCQ = buffer[27];
+  voip_metric_.RXconfig = buffer[28];
+  // reserved = buffer[29];
+  voip_metric_.JBnominal = ByteReader<uint16_t>::ReadBigEndian(&buffer[30]);
+  voip_metric_.JBmax = ByteReader<uint16_t>::ReadBigEndian(&buffer[32]);
+  voip_metric_.JBabsMax = ByteReader<uint16_t>::ReadBigEndian(&buffer[34]);
+}
+
+void VoipMetric::Create(uint8_t* buffer) const {
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockLength);
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ssrc_);
+  buffer[8] = voip_metric_.lossRate;
+  buffer[9] = voip_metric_.discardRate;
+  buffer[10] = voip_metric_.burstDensity;
+  buffer[11] = voip_metric_.gapDensity;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[12], voip_metric_.burstDuration);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[14], voip_metric_.gapDuration);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[16],
+                                       voip_metric_.roundTripDelay);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[18],
+                                       voip_metric_.endSystemDelay);
+  buffer[20] = voip_metric_.signalLevel;
+  buffer[21] = voip_metric_.noiseLevel;
+  buffer[22] = voip_metric_.RERL;
+  buffer[23] = voip_metric_.Gmin;
+  buffer[24] = voip_metric_.Rfactor;
+  buffer[25] = voip_metric_.extRfactor;
+  buffer[26] = voip_metric_.MOSLQ;
+  buffer[27] = voip_metric_.MOSCQ;
+  buffer[28] = voip_metric_.RXconfig;
+  buffer[29] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[30], voip_metric_.JBnominal);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[32], voip_metric_.JBmax);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[34], voip_metric_.JBabsMax);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h
new file mode 100644
index 0000000000..9e3e41995a
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_VOIP_METRIC_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_VOIP_METRIC_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class VoipMetric {
+ public:
+  static const uint8_t kBlockType = 7;
+  static const uint16_t kBlockLength = 8;
+  static const size_t kLength = 4 * (kBlockLength + 1);  // 36
+  VoipMetric();
+  VoipMetric(const VoipMetric&) = default;
+  ~VoipMetric() {}
+
+  VoipMetric& operator=(const VoipMetric&) = default;
+
+  void Parse(const uint8_t* buffer);
+
+  // Fills buffer with the VoipMetric.
+  // Consumes VoipMetric::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void To(uint32_t ssrc) { ssrc_ = ssrc; }
+  void WithVoipMetric(const RTCPVoIPMetric& voip_metric) {
+    voip_metric_ = voip_metric;
+  }
+
+  uint32_t ssrc() const { return ssrc_; }
+  const RTCPVoIPMetric& voip_metric() const { return voip_metric_; }
+
+ private:
+  uint32_t ssrc_;
+  RTCPVoIPMetric voip_metric_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_VOIP_METRIC_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc
new file mode 100644
index 0000000000..44c82d67a9
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kBlock[] = {0x07, 0x00, 0x00, 0x08, 0x23, 0x45, 0x67, 0x89,
+                          0x01, 0x02, 0x03, 0x04, 0x11, 0x12, 0x22, 0x23,
+                          0x33, 0x34, 0x44, 0x45, 0x05, 0x06, 0x07, 0x08,
+                          0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x00, 0x55, 0x56,
+                          0x66, 0x67, 0x77, 0x78};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+static_assert(
+    kBlockSizeBytes == VoipMetric::kLength,
+    "Size of manually created Voip Metric block should match class constant");
+
+TEST(RtcpPacketVoipMetricTest, Create) {
+  uint8_t buffer[VoipMetric::kLength];
+  RTCPVoIPMetric metric;
+  metric.lossRate = 1;
+  metric.discardRate = 2;
+  metric.burstDensity = 3;
+  metric.gapDensity = 4;
+  metric.burstDuration = 0x1112;
+  metric.gapDuration = 0x2223;
+  metric.roundTripDelay = 0x3334;
+  metric.endSystemDelay = 0x4445;
+  metric.signalLevel = 5;
+  metric.noiseLevel = 6;
+  metric.RERL = 7;
+  metric.Gmin = 8;
+  metric.Rfactor = 9;
+  metric.extRfactor = 10;
+  metric.MOSLQ = 11;
+  metric.MOSCQ = 12;
+  metric.RXconfig = 13;
+  metric.JBnominal = 0x5556;
+  metric.JBmax = 0x6667;
+  metric.JBabsMax = 0x7778;
+  VoipMetric metric_block;
+  metric_block.To(kRemoteSsrc);
+  metric_block.WithVoipMetric(metric);
+
+  metric_block.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketVoipMetricTest, Parse) {
+  VoipMetric read_metric;
+  read_metric.Parse(kBlock);
+
+  // Run checks on const object to ensure all accessors have const modifier.
+  const VoipMetric& parsed = read_metric;
+
+  EXPECT_EQ(kRemoteSsrc, parsed.ssrc());
+  EXPECT_EQ(1, parsed.voip_metric().lossRate);
+  EXPECT_EQ(2, parsed.voip_metric().discardRate);
+  EXPECT_EQ(3, parsed.voip_metric().burstDensity);
+  EXPECT_EQ(4, parsed.voip_metric().gapDensity);
+  EXPECT_EQ(0x1112, parsed.voip_metric().burstDuration);
+  EXPECT_EQ(0x2223, parsed.voip_metric().gapDuration);
+  EXPECT_EQ(0x3334, parsed.voip_metric().roundTripDelay);
+  EXPECT_EQ(0x4445, parsed.voip_metric().endSystemDelay);
+  EXPECT_EQ(5, parsed.voip_metric().signalLevel);
+  EXPECT_EQ(6, parsed.voip_metric().noiseLevel);
+  EXPECT_EQ(7, parsed.voip_metric().RERL);
+  EXPECT_EQ(8, parsed.voip_metric().Gmin);
+  EXPECT_EQ(9, parsed.voip_metric().Rfactor);
+  EXPECT_EQ(10, parsed.voip_metric().extRfactor);
+  EXPECT_EQ(11, parsed.voip_metric().MOSLQ);
+  EXPECT_EQ(12, parsed.voip_metric().MOSCQ);
+  EXPECT_EQ(13, parsed.voip_metric().RXconfig);
+  EXPECT_EQ(0x5556, parsed.voip_metric().JBnominal);
+  EXPECT_EQ(0x6667, parsed.voip_metric().JBmax);
+  EXPECT_EQ(0x7778, parsed.voip_metric().JBabsMax);
+}
+
+}  // namespace
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
index 77520b633b..22f61f5cab 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
@@ -14,6 +14,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
 #include "webrtc/test/rtcp_packet_parser.h"
 
 using ::testing::ElementsAre;
@@ -21,23 +24,15 @@ using ::testing::ElementsAre;
 using webrtc::rtcp::App;
 using webrtc::rtcp::Bye;
 using webrtc::rtcp::Dlrr;
-using webrtc::rtcp::Empty;
 using webrtc::rtcp::Fir;
-using webrtc::rtcp::Ij;
-using webrtc::rtcp::Nack;
-using webrtc::rtcp::Pli;
-using webrtc::rtcp::Sdes;
-using webrtc::rtcp::SenderReport;
-using webrtc::rtcp::Sli;
 using webrtc::rtcp::RawPacket;
 using webrtc::rtcp::ReceiverReport;
 using webrtc::rtcp::Remb;
 using webrtc::rtcp::ReportBlock;
 using webrtc::rtcp::Rpsi;
 using webrtc::rtcp::Rrtr;
+using webrtc::rtcp::Sdes;
 using webrtc::rtcp::SenderReport;
-using webrtc::rtcp::Tmmbn;
-using webrtc::rtcp::Tmmbr;
 using webrtc::rtcp::VoipMetric;
 using webrtc::rtcp::Xr;
 using webrtc::test::RtcpPacketParser;
@@ -47,81 +42,6 @@ namespace webrtc {
 const uint32_t kSenderSsrc = 0x12345678;
 const uint32_t kRemoteSsrc = 0x23456789;
 
-TEST(RtcpPacketTest, Rr) {
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-
-  rtc::scoped_ptr<RawPacket> packet(rr.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(0, parser.report_block()->num_packets());
-}
-
-TEST(RtcpPacketTest, RrWithOneReportBlock) {
-  ReportBlock rb;
-  rb.To(kRemoteSsrc);
-  rb.WithFractionLost(55);
-  rb.WithCumulativeLost(0x111111);
-  rb.WithExtHighestSeqNum(0x22222222);
-  rb.WithJitter(0x33333333);
-  rb.WithLastSr(0x44444444);
-  rb.WithDelayLastSr(0x55555555);
-
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  EXPECT_TRUE(rr.WithReportBlock(rb));
-
-  rtc::scoped_ptr<RawPacket> packet(rr.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(1, parser.report_block()->num_packets());
-  EXPECT_EQ(kRemoteSsrc, parser.report_block()->Ssrc());
-  EXPECT_EQ(55U, parser.report_block()->FractionLost());
-  EXPECT_EQ(0x111111U, parser.report_block()->CumPacketLost());
-  EXPECT_EQ(0x22222222U, parser.report_block()->ExtHighestSeqNum());
-  EXPECT_EQ(0x33333333U, parser.report_block()->Jitter());
-  EXPECT_EQ(0x44444444U, parser.report_block()->LastSr());
-  EXPECT_EQ(0x55555555U, parser.report_block()->DelayLastSr());
-}
-
-TEST(RtcpPacketTest, RrWithTwoReportBlocks) {
-  ReportBlock rb1;
-  rb1.To(kRemoteSsrc);
-  ReportBlock rb2;
-  rb2.To(kRemoteSsrc + 1);
-
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  EXPECT_TRUE(rr.WithReportBlock(rb1));
-  EXPECT_TRUE(rr.WithReportBlock(rb2));
-
-  rtc::scoped_ptr<RawPacket> packet(rr.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(2, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc));
-  EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc + 1));
-}
-
-TEST(RtcpPacketTest, RrWithTooManyReportBlocks) {
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  const int kMaxReportBlocks = (1 << 5) - 1;
-  ReportBlock rb;
-  for (int i = 0; i < kMaxReportBlocks; ++i) {
-    rb.To(kRemoteSsrc + i);
-    EXPECT_TRUE(rr.WithReportBlock(rb));
-  }
-  rb.To(kRemoteSsrc + kMaxReportBlocks);
-  EXPECT_FALSE(rr.WithReportBlock(rb));
-}
-
 TEST(RtcpPacketTest, Sr) {
   SenderReport sr;
   sr.From(kSenderSsrc);
@@ -196,50 +116,6 @@ TEST(RtcpPacketTest, SrWithTooManyReportBlocks) {
   EXPECT_FALSE(sr.WithReportBlock(rb));
 }
 
-TEST(RtcpPacketTest, IjNoItem) {
-  Ij ij;
-
-  rtc::scoped_ptr<RawPacket> packet(ij.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.ij()->num_packets());
-  EXPECT_EQ(0, parser.ij_item()->num_packets());
-}
-
-TEST(RtcpPacketTest, IjOneItem) {
-  Ij ij;
-  EXPECT_TRUE(ij.WithJitterItem(0x11111111));
-
-  rtc::scoped_ptr<RawPacket> packet(ij.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.ij()->num_packets());
-  EXPECT_EQ(1, parser.ij_item()->num_packets());
-  EXPECT_EQ(0x11111111U, parser.ij_item()->Jitter());
-}
-
-TEST(RtcpPacketTest, IjTwoItems) {
-  Ij ij;
-  EXPECT_TRUE(ij.WithJitterItem(0x11111111));
-  EXPECT_TRUE(ij.WithJitterItem(0x22222222));
-
-  rtc::scoped_ptr<RawPacket> packet(ij.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.ij()->num_packets());
-  EXPECT_EQ(2, parser.ij_item()->num_packets());
-  EXPECT_EQ(0x22222222U, parser.ij_item()->Jitter());
-}
-
-TEST(RtcpPacketTest, IjTooManyItems) {
-  Ij ij;
-  const int kMaxIjItems = (1 << 5) - 1;
-  for (int i = 0; i < kMaxIjItems; ++i) {
-    EXPECT_TRUE(ij.WithJitterItem(i));
-  }
-  EXPECT_FALSE(ij.WithJitterItem(kMaxIjItems));
-}
-
 TEST(RtcpPacketTest, AppWithNoData) {
   App app;
   app.WithSubType(30);
@@ -339,140 +215,6 @@ TEST(RtcpPacketTest, CnameItemWithEmptyString) {
   EXPECT_EQ("", parser.sdes_chunk()->Cname());
 }
 
-TEST(RtcpPacketTest, Pli) {
-  Pli pli;
-  pli.From(kSenderSsrc);
-  pli.To(kRemoteSsrc);
-
-  rtc::scoped_ptr<RawPacket> packet(pli.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.pli()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.pli()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.pli()->MediaSsrc());
-}
-
-TEST(RtcpPacketTest, Sli) {
-  const uint16_t kFirstMb = 7777;
-  const uint16_t kNumberOfMb = 6666;
-  const uint8_t kPictureId = 60;
-  Sli sli;
-  sli.From(kSenderSsrc);
-  sli.To(kRemoteSsrc);
-  sli.WithFirstMb(kFirstMb);
-  sli.WithNumberOfMb(kNumberOfMb);
-  sli.WithPictureId(kPictureId);
-
-  rtc::scoped_ptr<RawPacket> packet(sli.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.sli()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.sli()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.sli()->MediaSsrc());
-  EXPECT_EQ(1, parser.sli_item()->num_packets());
-  EXPECT_EQ(kFirstMb, parser.sli_item()->FirstMb());
-  EXPECT_EQ(kNumberOfMb, parser.sli_item()->NumberOfMb());
-  EXPECT_EQ(kPictureId, parser.sli_item()->PictureId());
-}
-
-TEST(RtcpPacketTest, Nack) {
-  Nack nack;
-  const uint16_t kList[] = {0, 1, 3, 8, 16};
-  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
-  nack.From(kSenderSsrc);
-  nack.To(kRemoteSsrc);
-  nack.WithList(kList, kListLength);
-  rtc::scoped_ptr<RawPacket> packet(nack.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.nack()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc());
-  EXPECT_EQ(1, parser.nack_item()->num_packets());
-  std::vector<uint16_t> seqs = parser.nack_item()->last_nack_list();
-  EXPECT_EQ(kListLength, seqs.size());
-  for (size_t i = 0; i < kListLength; ++i) {
-    EXPECT_EQ(kList[i], seqs[i]);
-  }
-}
-
-TEST(RtcpPacketTest, NackWithWrap) {
-  Nack nack;
-  const uint16_t kList[] = {65500, 65516, 65534, 65535, 0, 1, 3, 20, 100};
-  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
-  nack.From(kSenderSsrc);
-  nack.To(kRemoteSsrc);
-  nack.WithList(kList, kListLength);
-  rtc::scoped_ptr<RawPacket> packet(nack.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.nack()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc());
-  EXPECT_EQ(4, parser.nack_item()->num_packets());
-  std::vector<uint16_t> seqs = parser.nack_item()->last_nack_list();
-  EXPECT_EQ(kListLength, seqs.size());
-  for (size_t i = 0; i < kListLength; ++i) {
-    EXPECT_EQ(kList[i], seqs[i]);
-  }
-}
-
-TEST(RtcpPacketTest, NackFragmented) {
-  Nack nack;
-  const uint16_t kList[] = {1, 100, 200, 300, 400};
-  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
-  nack.From(kSenderSsrc);
-  nack.To(kRemoteSsrc);
-  nack.WithList(kList, kListLength);
-
-  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
-   public:
-    void OnPacketReady(uint8_t* data, size_t length) override {
-      ++packets_created_;
-      RtcpPacketParser parser;
-      parser.Parse(data, length);
-      EXPECT_EQ(1, parser.nack()->num_packets());
-      EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc());
-      EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc());
-      switch (packets_created_) {
-        case 1:
-          EXPECT_THAT(parser.nack_item()->last_nack_list(),
-                      ElementsAre(1, 100, 200));
-          break;
-        case 2:
-          EXPECT_THAT(parser.nack_item()->last_nack_list(),
-                      ElementsAre(300, 400));
-          break;
-        default:
-          ADD_FAILURE() << "Unexpected packet count: " << packets_created_;
-      }
-    }
-    int packets_created_ = 0;
-  } verifier;
-  const size_t kBufferSize = 12 + (3 * 4);  // Fits common header + 3 nack items
-  uint8_t buffer[kBufferSize];
-  EXPECT_TRUE(nack.BuildExternalBuffer(buffer, kBufferSize, &verifier));
-  EXPECT_EQ(2, verifier.packets_created_);
-}
-
-TEST(RtcpPacketTest, NackWithTooSmallBuffer) {
-  const uint16_t kList[] = {1};
-  const size_t kMinNackBlockSize = 16;
-  Nack nack;
-  nack.From(kSenderSsrc);
-  nack.To(kRemoteSsrc);
-  nack.WithList(kList, 1);
-  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
-   public:
-    void OnPacketReady(uint8_t* data, size_t length) override {
-      ADD_FAILURE() << "Buffer should be too small.";
-    }
-  } verifier;
-  uint8_t buffer[kMinNackBlockSize - 1];
-  EXPECT_FALSE(
-      nack.BuildExternalBuffer(buffer, kMinNackBlockSize - 1, &verifier));
-}
-
 TEST(RtcpPacketTest, Rpsi) {
   Rpsi rpsi;
   // 1000001 (7 bits = 1 byte in native string).
@@ -562,127 +304,6 @@ TEST(RtcpPacketTest, Fir) {
   EXPECT_EQ(123U, parser.fir_item()->SeqNum());
 }
 
-TEST(RtcpPacketTest, AppendPacket) {
-  Fir fir;
-  ReportBlock rb;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  EXPECT_TRUE(rr.WithReportBlock(rb));
-  rr.Append(&fir);
-
-  rtc::scoped_ptr<RawPacket> packet(rr.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(1, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.fir()->num_packets());
-}
-
-TEST(RtcpPacketTest, AppendPacketOnEmpty) {
-  Empty empty;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  empty.Append(&rr);
-
-  rtc::scoped_ptr<RawPacket> packet(empty.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(0, parser.report_block()->num_packets());
-}
-
-TEST(RtcpPacketTest, AppendPacketWithOwnAppendedPacket) {
-  Fir fir;
-  Bye bye;
-  ReportBlock rb;
-
-  ReceiverReport rr;
-  EXPECT_TRUE(rr.WithReportBlock(rb));
-  rr.Append(&fir);
-
-  SenderReport sr;
-  sr.Append(&bye);
-  sr.Append(&rr);
-
-  rtc::scoped_ptr<RawPacket> packet(sr.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.sender_report()->num_packets());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(1, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.bye()->num_packets());
-  EXPECT_EQ(1, parser.fir()->num_packets());
-}
-
-TEST(RtcpPacketTest, Bye) {
-  Bye bye;
-  bye.From(kSenderSsrc);
-
-  rtc::scoped_ptr<RawPacket> packet(bye.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.bye()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc());
-}
-
-TEST(RtcpPacketTest, ByeWithCsrcs) {
-  Fir fir;
-  Bye bye;
-  bye.From(kSenderSsrc);
-  EXPECT_TRUE(bye.WithCsrc(0x22222222));
-  EXPECT_TRUE(bye.WithCsrc(0x33333333));
-  bye.Append(&fir);
-
-  rtc::scoped_ptr<RawPacket> packet(bye.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.bye()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc());
-  EXPECT_EQ(1, parser.fir()->num_packets());
-}
-
-TEST(RtcpPacketTest, ByeWithTooManyCsrcs) {
-  Bye bye;
-  bye.From(kSenderSsrc);
-  const int kMaxCsrcs = (1 << 5) - 2;  // 5 bit len, first item is sender SSRC.
-  for (int i = 0; i < kMaxCsrcs; ++i) {
-    EXPECT_TRUE(bye.WithCsrc(i));
-  }
-  EXPECT_FALSE(bye.WithCsrc(kMaxCsrcs));
-}
-
-TEST(RtcpPacketTest, BuildWithInputBuffer) {
-  Fir fir;
-  ReportBlock rb;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  EXPECT_TRUE(rr.WithReportBlock(rb));
-  rr.Append(&fir);
-
-  const size_t kRrLength = 8;
-  const size_t kReportBlockLength = 24;
-  const size_t kFirLength = 20;
-
-  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
-   public:
-    void OnPacketReady(uint8_t* data, size_t length) override {
-      RtcpPacketParser parser;
-      parser.Parse(data, length);
-      EXPECT_EQ(1, parser.receiver_report()->num_packets());
-      EXPECT_EQ(1, parser.report_block()->num_packets());
-      EXPECT_EQ(1, parser.fir()->num_packets());
-      ++packets_created_;
-    }
-
-    int packets_created_ = 0;
-  } verifier;
-  const size_t kBufferSize = kRrLength + kReportBlockLength + kFirLength;
-  uint8_t buffer[kBufferSize];
-  EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
-  EXPECT_EQ(1, verifier.packets_created_);
-}
-
 TEST(RtcpPacketTest, BuildWithTooSmallBuffer) {
   ReportBlock rb;
   ReceiverReport rr;
@@ -703,47 +324,6 @@ TEST(RtcpPacketTest, BuildWithTooSmallBuffer) {
   EXPECT_FALSE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
 }
 
-TEST(RtcpPacketTest, BuildWithTooSmallBuffer_FragmentedSend) {
-  Fir fir;
-  ReportBlock rb;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  EXPECT_TRUE(rr.WithReportBlock(rb));
-  rr.Append(&fir);
-
-  const size_t kRrLength = 8;
-  const size_t kReportBlockLength = 24;
-
-  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
-   public:
-    void OnPacketReady(uint8_t* data, size_t length) override {
-      RtcpPacketParser parser;
-      parser.Parse(data, length);
-      switch (packets_created_++) {
-        case 0:
-          EXPECT_EQ(1, parser.receiver_report()->num_packets());
-          EXPECT_EQ(1, parser.report_block()->num_packets());
-          EXPECT_EQ(0, parser.fir()->num_packets());
-          break;
-        case 1:
-          EXPECT_EQ(0, parser.receiver_report()->num_packets());
-          EXPECT_EQ(0, parser.report_block()->num_packets());
-          EXPECT_EQ(1, parser.fir()->num_packets());
-          break;
-        default:
-          ADD_FAILURE() << "OnPacketReady not expected to be called "
-                        << packets_created_ << " times.";
-      }
-    }
-
-    int packets_created_ = 0;
-  } verifier;
-  const size_t kBufferSize = kRrLength + kReportBlockLength;
-  uint8_t buffer[kBufferSize];
-  EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
-  EXPECT_EQ(2, verifier.packets_created_);
-}
-
 TEST(RtcpPacketTest, Remb) {
   Remb remb;
   remb.From(kSenderSsrc);
@@ -765,81 +345,6 @@ TEST(RtcpPacketTest, Remb) {
   EXPECT_EQ(kRemoteSsrc + 2, ssrcs[2]);
 }
 
-TEST(RtcpPacketTest, Tmmbr) {
-  Tmmbr tmmbr;
-  tmmbr.From(kSenderSsrc);
-  tmmbr.To(kRemoteSsrc);
-  tmmbr.WithBitrateKbps(312);
-  tmmbr.WithOverhead(60);
-
-  rtc::scoped_ptr<RawPacket> packet(tmmbr.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.tmmbr()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbr()->Ssrc());
-  EXPECT_EQ(1, parser.tmmbr_item()->num_packets());
-  EXPECT_EQ(312U, parser.tmmbr_item()->BitrateKbps());
-  EXPECT_EQ(60U, parser.tmmbr_item()->Overhead());
-}
-
-TEST(RtcpPacketTest, TmmbnWithNoItem) {
-  Tmmbn tmmbn;
-  tmmbn.From(kSenderSsrc);
-
-  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.tmmbn()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
-  EXPECT_EQ(0, parser.tmmbn_items()->num_packets());
-}
-
-TEST(RtcpPacketTest, TmmbnWithOneItem) {
-  Tmmbn tmmbn;
-  tmmbn.From(kSenderSsrc);
-  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc, 312, 60));
-
-  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.tmmbn()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
-  EXPECT_EQ(1, parser.tmmbn_items()->num_packets());
-  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
-  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
-  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
-}
-
-TEST(RtcpPacketTest, TmmbnWithTwoItems) {
-  Tmmbn tmmbn;
-  tmmbn.From(kSenderSsrc);
-  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc, 312, 60));
-  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc + 1, 1288, 40));
-
-  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
-  RtcpPacketParser parser;
-  parser.Parse(packet->Buffer(), packet->Length());
-  EXPECT_EQ(1, parser.tmmbn()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
-  EXPECT_EQ(2, parser.tmmbn_items()->num_packets());
-  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
-  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
-  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
-  EXPECT_EQ(kRemoteSsrc + 1, parser.tmmbn_items()->Ssrc(1));
-  EXPECT_EQ(1288U, parser.tmmbn_items()->BitrateKbps(1));
-  EXPECT_EQ(40U, parser.tmmbn_items()->Overhead(1));
-}
-
-TEST(RtcpPacketTest, TmmbnWithTooManyItems) {
-  Tmmbn tmmbn;
-  tmmbn.From(kSenderSsrc);
-  const int kMaxTmmbrItems = 50;
-  for (int i = 0; i < kMaxTmmbrItems; ++i)
-    EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc + i, 312, 60));
-
-  EXPECT_FALSE(tmmbn.WithTmmbr(kRemoteSsrc + kMaxTmmbrItems, 312, 60));
-}
-
 TEST(RtcpPacketTest, XrWithNoReportBlocks) {
   Xr xr;
   xr.From(kSenderSsrc);
@@ -853,8 +358,7 @@ TEST(RtcpPacketTest, XrWithNoReportBlocks) {
 
 TEST(RtcpPacketTest, XrWithRrtr) {
   Rrtr rrtr;
-  rrtr.WithNtpSec(0x11111111);
-  rrtr.WithNtpFrac(0x22222222);
+  rrtr.WithNtp(NtpTime(0x11111111, 0x22222222));
   Xr xr;
   xr.From(kSenderSsrc);
   EXPECT_TRUE(xr.WithRrtr(&rrtr));
@@ -871,11 +375,9 @@ TEST(RtcpPacketTest, XrWithRrtr) {
 
 TEST(RtcpPacketTest, XrWithTwoRrtrBlocks) {
   Rrtr rrtr1;
-  rrtr1.WithNtpSec(0x11111111);
-  rrtr1.WithNtpFrac(0x22222222);
+  rrtr1.WithNtp(NtpTime(0x11111111, 0x22222222));
   Rrtr rrtr2;
-  rrtr2.WithNtpSec(0x33333333);
-  rrtr2.WithNtpFrac(0x44444444);
+  rrtr2.WithNtp(NtpTime(0x33333333, 0x44444444));
   Xr xr;
   xr.From(kSenderSsrc);
   EXPECT_TRUE(xr.WithRrtr(&rrtr1));
@@ -967,32 +469,33 @@ TEST(RtcpPacketTest, XrWithTwoDlrrBlocks) {
 }
 
 TEST(RtcpPacketTest, XrWithVoipMetric) {
-  VoipMetric metric;
-  metric.To(kRemoteSsrc);
-  metric.LossRate(1);
-  metric.DiscardRate(2);
-  metric.BurstDensity(3);
-  metric.GapDensity(4);
-  metric.BurstDuration(0x1111);
-  metric.GapDuration(0x2222);
-  metric.RoundTripDelay(0x3333);
-  metric.EndSystemDelay(0x4444);
-  metric.SignalLevel(5);
-  metric.NoiseLevel(6);
-  metric.Rerl(7);
-  metric.Gmin(8);
-  metric.Rfactor(9);
-  metric.ExtRfactor(10);
-  metric.MosLq(11);
-  metric.MosCq(12);
-  metric.RxConfig(13);
-  metric.JbNominal(0x5555);
-  metric.JbMax(0x6666);
-  metric.JbAbsMax(0x7777);
-
+  RTCPVoIPMetric metric;
+  metric.lossRate = 1;
+  metric.discardRate = 2;
+  metric.burstDensity = 3;
+  metric.gapDensity = 4;
+  metric.burstDuration = 0x1111;
+  metric.gapDuration = 0x2222;
+  metric.roundTripDelay = 0x3333;
+  metric.endSystemDelay = 0x4444;
+  metric.signalLevel = 5;
+  metric.noiseLevel = 6;
+  metric.RERL = 7;
+  metric.Gmin = 8;
+  metric.Rfactor = 9;
+  metric.extRfactor = 10;
+  metric.MOSLQ = 11;
+  metric.MOSCQ = 12;
+  metric.RXconfig = 13;
+  metric.JBnominal = 0x5555;
+  metric.JBmax = 0x6666;
+  metric.JBabsMax = 0x7777;
+  VoipMetric metric_block;
+  metric_block.To(kRemoteSsrc);
+  metric_block.WithVoipMetric(metric);
   Xr xr;
   xr.From(kSenderSsrc);
-  EXPECT_TRUE(xr.WithVoipMetric(&metric));
+  EXPECT_TRUE(xr.WithVoipMetric(&metric_block));
 
   rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index b914838109..d65b04c8ab 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -23,8 +23,13 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 
 namespace webrtc {
-using namespace RTCPUtility;
-using namespace RTCPHelp;
+using RTCPHelp::RTCPPacketInformation;
+using RTCPHelp::RTCPReceiveInformation;
+using RTCPHelp::RTCPReportBlockInformation;
+using RTCPUtility::kBtVoipMetric;
+using RTCPUtility::RTCPCnameInformation;
+using RTCPUtility::RTCPPacketReportBlockItem;
+using RTCPUtility::RTCPPacketTypes;
 
 // The number of RTCP time intervals needed to trigger a timeout.
 const int kRrTimeoutIntervals = 3;
@@ -741,7 +746,7 @@ bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
   return updateBoundingSet;
 }
 
-int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
+int32_t RTCPReceiver::BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec) {
   CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPReceiveInformation*>::iterator receiveInfoIt =
@@ -761,7 +766,7 @@ int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
         i++) {
       if(receiveInfo->TmmbnBoundingSet.Ssrc(i) == main_ssrc_) {
         // owner of bounding set
-        tmmbrOwner = true;
+        *tmmbrOwner = true;
       }
       boundingSetRec->SetEntry(i,
                                receiveInfo->TmmbnBoundingSet.Tmmbr(i),
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 272397675b..24861bd49e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -12,11 +12,11 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
 
 #include <map>
-#include <vector>
 #include <set>
+#include <vector>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
@@ -103,7 +103,7 @@ public:
 
     bool UpdateRTCPReceiveInformationTimers();
 
-    int32_t BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec);
+    int32_t BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec);
 
     int32_t UpdateTMMBR();
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
index 718990d10b..a5c0e28282 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
@@ -17,7 +17,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
-using namespace RTCPHelp;
+namespace RTCPHelp {
 
 RTCPPacketInformation::RTCPPacketInformation()
     : rtcpPacketTypeFlags(0),
@@ -190,4 +190,5 @@ void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(
     const uint32_t minimumSize) {
   TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
 }
+}  // namespace RTCPHelp
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
index 37b7b88370..a792841962 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -11,10 +11,12 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 
+#include <list>
+#include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"  // RTCPReportBlock
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPReportBlock
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 #include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 1581845476..5d2fda347e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -15,17 +15,23 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
-// Note: This file has no directory. Lint warning must be ignored.
 #include "webrtc/common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
 
@@ -384,20 +390,20 @@ TEST_F(RtcpReceiverTest, GetRtt) {
 }
 
 TEST_F(RtcpReceiverTest, InjectIjWithNoItem) {
-  rtcp::Ij ij;
+  rtcp::ExtendedJitterReport ij;
   rtc::scoped_ptr<rtcp::RawPacket> packet(ij.Build());
   EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
 TEST_F(RtcpReceiverTest, InjectIjWithOneItem) {
-  rtcp::Ij ij;
-  ij.WithJitterItem(0x11111111);
+  rtcp::ExtendedJitterReport ij;
+  ij.WithJitter(0x11213141);
 
   rtc::scoped_ptr<rtcp::RawPacket> packet(ij.Build());
   EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpTransmissionTimeOffset, rtcp_packet_info_.rtcpPacketTypeFlags);
-  EXPECT_EQ(0x11111111U, rtcp_packet_info_.interArrivalJitter);
+  EXPECT_EQ(0x11213141U, rtcp_packet_info_.interArrivalJitter);
 }
 
 TEST_F(RtcpReceiverTest, InjectAppWithNoData) {
@@ -586,7 +592,9 @@ TEST_F(RtcpReceiverTest, InjectXrVoipPacket) {
   const uint8_t kLossRate = 123;
   rtcp::VoipMetric voip_metric;
   voip_metric.To(kSourceSsrc);
-  voip_metric.LossRate(kLossRate);
+  RTCPVoIPMetric metric;
+  metric.lossRate = kLossRate;
+  voip_metric.WithVoipMetric(metric);
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithVoipMetric(&voip_metric);
@@ -615,8 +623,7 @@ TEST_F(RtcpReceiverTest, XrVoipPacketNotToUsIgnored) {
 
 TEST_F(RtcpReceiverTest, InjectXrReceiverReferenceTimePacket) {
   rtcp::Rrtr rrtr;
-  rrtr.WithNtpSec(0x10203);
-  rrtr.WithNtpFrac(0x40506);
+  rrtr.WithNtp(NtpTime(0x10203, 0x40506));
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithRrtr(&rrtr);
@@ -751,13 +758,12 @@ TEST_F(RtcpReceiverTest, LastReceivedXrReferenceTimeInfoInitiallyFalse) {
 
 TEST_F(RtcpReceiverTest, GetLastReceivedXrReferenceTimeInfo) {
   const uint32_t kSenderSsrc = 0x123456;
-  const uint32_t kNtpSec = 0x10203;
-  const uint32_t kNtpFrac = 0x40506;
-  const uint32_t kNtpMid = RTCPUtility::MidNtp(kNtpSec, kNtpFrac);
+  const NtpTime kNtp(0x10203, 0x40506);
+  const uint32_t kNtpMid =
+      RTCPUtility::MidNtp(kNtp.seconds(), kNtp.fractions());
 
   rtcp::Rrtr rrtr;
-  rrtr.WithNtpSec(kNtpSec);
-  rrtr.WithNtpFrac(kNtpFrac);
+  rrtr.WithNtp(kNtp);
   rtcp::Xr xr;
   xr.From(kSenderSsrc);
   xr.WithRrtr(&rrtr);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 22b9477e05..848d73b2c4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -11,17 +11,26 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 
 #include <assert.h>  // assert
-#include <stdlib.h>  // rand
 #include <string.h>  // memcpy
 
 #include <algorithm>  // min
 #include <limits>     // max
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
@@ -31,13 +40,11 @@ namespace webrtc {
 using RTCPUtility::RTCPCnameInformation;
 
 NACKStringBuilder::NACKStringBuilder()
-    : stream_(""), count_(0), prevNack_(0), consecutive_(false) {
-}
+    : stream_(""), count_(0), prevNack_(0), consecutive_(false) {}
 
 NACKStringBuilder::~NACKStringBuilder() {}
 
-void NACKStringBuilder::PushNACK(uint16_t nack)
-{
+void NACKStringBuilder::PushNACK(uint16_t nack) {
   if (count_ == 0) {
     stream_ << nack;
   } else if (nack == prevNack_ + 1) {
@@ -71,64 +78,63 @@ RTCPSender::FeedbackState::FeedbackState()
       last_rr_ntp_frac(0),
       remote_sr(0),
       has_last_xr_rr(false),
-      module(nullptr) {
-}
+      module(nullptr) {}
 
-struct RTCPSender::RtcpContext {
-  RtcpContext(const FeedbackState& feedback_state,
-              int32_t nack_size,
-              const uint16_t* nack_list,
-              bool repeat,
-              uint64_t picture_id,
-              uint8_t* buffer,
-              uint32_t buffer_size)
-      : feedback_state(feedback_state),
-        nack_size(nack_size),
-        nack_list(nack_list),
-        repeat(repeat),
-        picture_id(picture_id),
-        buffer(buffer),
-        buffer_size(buffer_size),
-        ntp_sec(0),
-        ntp_frac(0),
-        position(0) {}
-
-  uint8_t* AllocateData(uint32_t bytes) {
-    RTC_DCHECK_LE(position + bytes, buffer_size);
-    uint8_t* ptr = &buffer[position];
-    position += bytes;
-    return ptr;
+class PacketContainer : public rtcp::CompoundPacket,
+                        public rtcp::RtcpPacket::PacketReadyCallback {
+ public:
+  explicit PacketContainer(Transport* transport)
+      : transport_(transport), bytes_sent_(0) {}
+  virtual ~PacketContainer() {
+    for (RtcpPacket* packet : appended_packets_)
+      delete packet;
   }
 
-  const FeedbackState& feedback_state;
-  int32_t nack_size;
-  const uint16_t* nack_list;
-  bool repeat;
-  uint64_t picture_id;
-  uint8_t* buffer;
-  uint32_t buffer_size;
-  uint32_t ntp_sec;
-  uint32_t ntp_frac;
-  uint32_t position;
-};
-
-// TODO(sprang): Once all builders use RtcpPacket, call SendToNetwork() here.
-class RTCPSender::PacketBuiltCallback
-    : public rtcp::RtcpPacket::PacketReadyCallback {
- public:
-  PacketBuiltCallback(RtcpContext* context) : context_(context) {}
-  virtual ~PacketBuiltCallback() {}
   void OnPacketReady(uint8_t* data, size_t length) override {
-    context_->position += length;
+    if (transport_->SendRtcp(data, length))
+      bytes_sent_ += length;
   }
-  bool BuildPacket(const rtcp::RtcpPacket& packet) {
-    return packet.BuildExternalBuffer(
-        &context_->buffer[context_->position],
-        context_->buffer_size - context_->position, this);
+
+  size_t SendPackets() {
+    rtcp::CompoundPacket::Build(this);
+    return bytes_sent_;
   }
 
  private:
-  RtcpContext* const context_;
+  Transport* transport_;
+  size_t bytes_sent_;
+};
+
+class RTCPSender::RtcpContext {
+ public:
+  RtcpContext(const FeedbackState& feedback_state,
+              int32_t nack_size,
+              const uint16_t* nack_list,
+              bool repeat,
+              uint64_t picture_id,
+              uint32_t ntp_sec,
+              uint32_t ntp_frac,
+              PacketContainer* container)
+      : feedback_state_(feedback_state),
+        nack_size_(nack_size),
+        nack_list_(nack_list),
+        repeat_(repeat),
+        picture_id_(picture_id),
+        ntp_sec_(ntp_sec),
+        ntp_frac_(ntp_frac),
+        container_(container) {}
+
+  virtual ~RtcpContext() {}
+
+  const FeedbackState& feedback_state_;
+  const int32_t nack_size_;
+  const uint16_t* nack_list_;
+  const bool repeat_;
+  const uint64_t picture_id_;
+  const uint32_t ntp_sec_;
+  const uint32_t ntp_frac_;
+
+  PacketContainer* const container_;
 };
 
 RTCPSender::RTCPSender(
@@ -139,6 +145,7 @@ RTCPSender::RTCPSender(
     Transport* outgoing_transport)
     : audio_(audio),
       clock_(clock),
+      random_(clock_->TimeInMicroseconds()),
       method_(RtcpMode::kOff),
       transport_(outgoing_transport),
 
@@ -193,8 +200,7 @@ RTCPSender::RTCPSender(
   builders_[kRtcpXrDlrrReportBlock] = &RTCPSender::BuildDlrr;
 }
 
-RTCPSender::~RTCPSender() {
-}
+RTCPSender::~RTCPSender() {}
 
 RtcpMode RTCPSender::Status() const {
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
@@ -340,63 +346,63 @@ int32_t RTCPSender::RemoveMixedCNAME(uint32_t SSRC) {
 }
 
 bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
-/*
-    For audio we use a fix 5 sec interval
+  /*
+      For audio we use a fix 5 sec interval
 
-    For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
-        technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
-        that should be extremely rare
+      For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
+          technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
+          that should be extremely rare
 
 
-From RFC 3550
+  From RFC 3550
 
-    MAX RTCP BW is 5% if the session BW
-        A send report is approximately 65 bytes inc CNAME
-        A receiver report is approximately 28 bytes
+      MAX RTCP BW is 5% if the session BW
+          A send report is approximately 65 bytes inc CNAME
+          A receiver report is approximately 28 bytes
 
-    The RECOMMENDED value for the reduced minimum in seconds is 360
-      divided by the session bandwidth in kilobits/second.  This minimum
-      is smaller than 5 seconds for bandwidths greater than 72 kb/s.
+      The RECOMMENDED value for the reduced minimum in seconds is 360
+        divided by the session bandwidth in kilobits/second.  This minimum
+        is smaller than 5 seconds for bandwidths greater than 72 kb/s.
 
-    If the participant has not yet sent an RTCP packet (the variable
-      initial is true), the constant Tmin is set to 2.5 seconds, else it
-      is set to 5 seconds.
+      If the participant has not yet sent an RTCP packet (the variable
+        initial is true), the constant Tmin is set to 2.5 seconds, else it
+        is set to 5 seconds.
 
-    The interval between RTCP packets is varied randomly over the
-      range [0.5,1.5] times the calculated interval to avoid unintended
-      synchronization of all participants
+      The interval between RTCP packets is varied randomly over the
+        range [0.5,1.5] times the calculated interval to avoid unintended
+        synchronization of all participants
 
-    if we send
-    If the participant is a sender (we_sent true), the constant C is
-      set to the average RTCP packet size (avg_rtcp_size) divided by 25%
-      of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
-      number of senders.
+      if we send
+      If the participant is a sender (we_sent true), the constant C is
+        set to the average RTCP packet size (avg_rtcp_size) divided by 25%
+        of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
+        number of senders.
 
-    if we receive only
-      If we_sent is not true, the constant C is set
-      to the average RTCP packet size divided by 75% of the RTCP
-      bandwidth.  The constant n is set to the number of receivers
-      (members - senders).  If the number of senders is greater than
-      25%, senders and receivers are treated together.
+      if we receive only
+        If we_sent is not true, the constant C is set
+        to the average RTCP packet size divided by 75% of the RTCP
+        bandwidth.  The constant n is set to the number of receivers
+        (members - senders).  If the number of senders is greater than
+        25%, senders and receivers are treated together.
 
-    reconsideration NOT required for peer-to-peer
-      "timer reconsideration" is
-      employed.  This algorithm implements a simple back-off mechanism
-      which causes users to hold back RTCP packet transmission if the
-      group sizes are increasing.
+      reconsideration NOT required for peer-to-peer
+        "timer reconsideration" is
+        employed.  This algorithm implements a simple back-off mechanism
+        which causes users to hold back RTCP packet transmission if the
+        group sizes are increasing.
 
-      n = number of members
-      C = avg_size/(rtcpBW/4)
+        n = number of members
+        C = avg_size/(rtcpBW/4)
 
-   3. The deterministic calculated interval Td is set to max(Tmin, n*C).
+     3. The deterministic calculated interval Td is set to max(Tmin, n*C).
 
-   4. The calculated interval T is set to a number uniformly distributed
-      between 0.5 and 1.5 times the deterministic calculated interval.
+     4. The calculated interval T is set to a number uniformly distributed
+        between 0.5 and 1.5 times the deterministic calculated interval.
 
-   5. The resulting value of T is divided by e-3/2=1.21828 to compensate
-      for the fact that the timer reconsideration algorithm converges to
-      a value of the RTCP bandwidth below the intended average
-*/
+     5. The resulting value of T is divided by e-3/2=1.21828 to compensate
+        for the fact that the timer reconsideration algorithm converges to
+        a value of the RTCP bandwidth below the intended average
+  */
 
   int64_t now = clock_->TimeInMilliseconds();
 
@@ -451,32 +457,15 @@ bool RTCPSender::SendTimeOfXrRrReport(uint32_t mid_ntp,
   return true;
 }
 
-int32_t RTCPSender::AddReportBlock(const RTCPReportBlock& report_block) {
-  if (report_blocks_.size() >= RTCP_MAX_REPORT_BLOCKS) {
-    LOG(LS_WARNING) << "Too many report blocks.";
-    return -1;
-  }
-  rtcp::ReportBlock* block = &report_blocks_[report_block.remoteSSRC];
-  block->To(report_block.remoteSSRC);
-  block->WithFractionLost(report_block.fractionLost);
-  block->WithCumulativeLost(report_block.cumulativeLost);
-  block->WithExtHighestSeqNum(report_block.extendedHighSeqNum);
-  block->WithJitter(report_block.jitter);
-  block->WithLastSr(report_block.lastSR);
-  block->WithDelayLastSr(report_block.delaySinceLastSR);
-
-  return 0;
-}
-
-RTCPSender::BuildResult RTCPSender::BuildSR(RtcpContext* ctx) {
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
   for (int i = (RTCP_NUMBER_OF_SR - 2); i >= 0; i--) {
     // shift old
     last_send_report_[i + 1] = last_send_report_[i];
     last_rtcp_time_[i + 1] = last_rtcp_time_[i];
   }
 
-  last_rtcp_time_[0] = Clock::NtpToMs(ctx->ntp_sec, ctx->ntp_frac);
-  last_send_report_[0] = (ctx->ntp_sec << 16) + (ctx->ntp_frac >> 16);
+  last_rtcp_time_[0] = Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_);
+  last_send_report_[0] = (ctx.ntp_sec_ << 16) + (ctx.ntp_frac_ >> 16);
 
   // The timestamp of this RTCP packet should be estimated as the timestamp of
   // the frame being captured at this moment. We are calculating that
@@ -485,67 +474,52 @@ RTCPSender::BuildResult RTCPSender::BuildSR(RtcpContext* ctx) {
   uint32_t rtp_timestamp =
       start_timestamp_ + last_rtp_timestamp_ +
       (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) *
-          (ctx->feedback_state.frequency_hz / 1000);
+          (ctx.feedback_state_.frequency_hz / 1000);
 
-  rtcp::SenderReport report;
-  report.From(ssrc_);
-  report.WithNtpSec(ctx->ntp_sec);
-  report.WithNtpFrac(ctx->ntp_frac);
-  report.WithRtpTimestamp(rtp_timestamp);
-  report.WithPacketCount(ctx->feedback_state.packets_sent);
-  report.WithOctetCount(ctx->feedback_state.media_bytes_sent);
+  rtcp::SenderReport* report = new rtcp::SenderReport();
+  report->From(ssrc_);
+  report->WithNtpSec(ctx.ntp_sec_);
+  report->WithNtpFrac(ctx.ntp_frac_);
+  report->WithRtpTimestamp(rtp_timestamp);
+  report->WithPacketCount(ctx.feedback_state_.packets_sent);
+  report->WithOctetCount(ctx.feedback_state_.media_bytes_sent);
 
   for (auto it : report_blocks_)
-    report.WithReportBlock(it.second);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(report))
-    return BuildResult::kTruncated;
+    report->WithReportBlock(it.second);
 
   report_blocks_.clear();
-  return BuildResult::kSuccess;
+
+  return rtc::scoped_ptr<rtcp::SenderReport>(report);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildSDES(RtcpContext* ctx) {
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSDES(
+    const RtcpContext& ctx) {
   size_t length_cname = cname_.length();
   RTC_CHECK_LT(length_cname, static_cast<size_t>(RTCP_CNAME_SIZE));
 
-  rtcp::Sdes sdes;
-  sdes.WithCName(ssrc_, cname_);
+  rtcp::Sdes* sdes = new rtcp::Sdes();
+  sdes->WithCName(ssrc_, cname_);
 
   for (const auto it : csrc_cnames_)
-    sdes.WithCName(it.first, it.second);
+    sdes->WithCName(it.first, it.second);
 
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(sdes))
-    return BuildResult::kTruncated;
-
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Sdes>(sdes);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildRR(RtcpContext* ctx) {
-  rtcp::ReceiverReport report;
-  report.From(ssrc_);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildRR(const RtcpContext& ctx) {
+  rtcp::ReceiverReport* report = new rtcp::ReceiverReport();
+  report->From(ssrc_);
   for (auto it : report_blocks_)
-    report.WithReportBlock(it.second);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(report))
-    return BuildResult::kTruncated;
+    report->WithReportBlock(it.second);
 
   report_blocks_.clear();
-
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::ReceiverReport>(report);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildPLI(RtcpContext* ctx) {
-  rtcp::Pli pli;
-  pli.From(ssrc_);
-  pli.To(remote_ssrc_);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(pli))
-    return BuildResult::kTruncated;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildPLI(const RtcpContext& ctx) {
+  rtcp::Pli* pli = new rtcp::Pli();
+  pli->From(ssrc_);
+  pli->To(remote_ssrc_);
 
   TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTCPSender::PLI");
@@ -553,21 +527,17 @@ RTCPSender::BuildResult RTCPSender::BuildPLI(RtcpContext* ctx) {
   TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_PLICount",
                     ssrc_, packet_type_counter_.pli_packets);
 
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Pli>(pli);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildFIR(RtcpContext* ctx) {
-  if (!ctx->repeat)
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildFIR(const RtcpContext& ctx) {
+  if (!ctx.repeat_)
     ++sequence_number_fir_;  // Do not increase if repetition.
 
-  rtcp::Fir fir;
-  fir.From(ssrc_);
-  fir.To(remote_ssrc_);
-  fir.WithCommandSeqNum(sequence_number_fir_);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(fir))
-    return BuildResult::kTruncated;
+  rtcp::Fir* fir = new rtcp::Fir();
+  fir->From(ssrc_);
+  fir->To(remote_ssrc_);
+  fir->WithCommandSeqNum(sequence_number_fir_);
 
   TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTCPSender::FIR");
@@ -575,7 +545,7 @@ RTCPSender::BuildResult RTCPSender::BuildFIR(RtcpContext* ctx) {
   TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_FIRCount",
                     ssrc_, packet_type_counter_.fir_packets);
 
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Fir>(fir);
 }
 
 /*
@@ -585,20 +555,14 @@ RTCPSender::BuildResult RTCPSender::BuildFIR(RtcpContext* ctx) {
    |            First        |        Number           | PictureID |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 */
-RTCPSender::BuildResult RTCPSender::BuildSLI(RtcpContext* ctx) {
-  rtcp::Sli sli;
-  sli.From(ssrc_);
-  sli.To(remote_ssrc_);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSLI(const RtcpContext& ctx) {
+  rtcp::Sli* sli = new rtcp::Sli();
+  sli->From(ssrc_);
+  sli->To(remote_ssrc_);
   // Crop picture id to 6 least significant bits.
-  sli.WithPictureId(ctx->picture_id & 0x3F);
-  sli.WithFirstMb(0);
-  sli.WithNumberOfMb(0x1FFF);  // 13 bits, only ones for now.
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(sli))
-    return BuildResult::kTruncated;
+  sli->WithPictureId(ctx.picture_id_ & 0x3F);
 
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Sli>(sli);
 }
 
 /*
@@ -613,38 +577,32 @@ RTCPSender::BuildResult RTCPSender::BuildSLI(RtcpContext* ctx) {
 /*
 *    Note: not generic made for VP8
 */
-RTCPSender::BuildResult RTCPSender::BuildRPSI(RtcpContext* ctx) {
-  if (ctx->feedback_state.send_payload_type == 0xFF)
-    return BuildResult::kError;
-
-  rtcp::Rpsi rpsi;
-  rpsi.From(ssrc_);
-  rpsi.To(remote_ssrc_);
-  rpsi.WithPayloadType(ctx->feedback_state.send_payload_type);
-  rpsi.WithPictureId(ctx->picture_id);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(rpsi))
-    return BuildResult::kTruncated;
-
-  return BuildResult::kSuccess;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildRPSI(
+    const RtcpContext& ctx) {
+  if (ctx.feedback_state_.send_payload_type == 0xFF)
+    return nullptr;
+
+  rtcp::Rpsi* rpsi = new rtcp::Rpsi();
+  rpsi->From(ssrc_);
+  rpsi->To(remote_ssrc_);
+  rpsi->WithPayloadType(ctx.feedback_state_.send_payload_type);
+  rpsi->WithPictureId(ctx.picture_id_);
+
+  return rtc::scoped_ptr<rtcp::Rpsi>(rpsi);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildREMB(RtcpContext* ctx) {
-  rtcp::Remb remb;
-  remb.From(ssrc_);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildREMB(
+    const RtcpContext& ctx) {
+  rtcp::Remb* remb = new rtcp::Remb();
+  remb->From(ssrc_);
   for (uint32_t ssrc : remb_ssrcs_)
-    remb.AppliesTo(ssrc);
-  remb.WithBitrateBps(remb_bitrate_);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(remb))
-    return BuildResult::kTruncated;
+    remb->AppliesTo(ssrc);
+  remb->WithBitrateBps(remb_bitrate_);
 
   TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTCPSender::REMB");
 
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Remb>(remb);
 }
 
 void RTCPSender::SetTargetBitrate(unsigned int target_bitrate) {
@@ -652,9 +610,10 @@ void RTCPSender::SetTargetBitrate(unsigned int target_bitrate) {
   tmmbr_send_ = target_bitrate / 1000;
 }
 
-RTCPSender::BuildResult RTCPSender::BuildTMMBR(RtcpContext* ctx) {
-  if (ctx->feedback_state.module == NULL)
-    return BuildResult::kError;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBR(
+    const RtcpContext& ctx) {
+  if (ctx.feedback_state_.module == nullptr)
+    return nullptr;
   // Before sending the TMMBR check the received TMMBN, only an owner is
   // allowed to raise the bitrate:
   // * If the sender is an owner of the TMMBN -> send TMMBR
@@ -669,14 +628,14 @@ RTCPSender::BuildResult RTCPSender::BuildTMMBR(RtcpContext* ctx) {
   // will accuire criticalSectionRTCPReceiver_ is a potental deadlock but
   // since RTCPreceiver is not doing the reverse we should be fine
   int32_t lengthOfBoundingSet =
-      ctx->feedback_state.module->BoundingSet(tmmbrOwner, candidateSet);
+      ctx.feedback_state_.module->BoundingSet(&tmmbrOwner, candidateSet);
 
   if (lengthOfBoundingSet > 0) {
     for (int32_t i = 0; i < lengthOfBoundingSet; i++) {
       if (candidateSet->Tmmbr(i) == tmmbr_send_ &&
           candidateSet->PacketOH(i) == packet_oh_send_) {
-        // do not send the same tuple
-        return BuildResult::kAborted;
+        // Do not send the same tuple.
+        return nullptr;
       }
     }
     if (!tmmbrOwner) {
@@ -687,124 +646,69 @@ RTCPSender::BuildResult RTCPSender::BuildTMMBR(RtcpContext* ctx) {
       int numCandidates = lengthOfBoundingSet + 1;
 
       // find bounding set
-      TMMBRSet* boundingSet = NULL;
+      TMMBRSet* boundingSet = nullptr;
       int numBoundingSet = tmmbr_help_.FindTMMBRBoundingSet(boundingSet);
       if (numBoundingSet > 0 || numBoundingSet <= numCandidates)
         tmmbrOwner = tmmbr_help_.IsOwner(ssrc_, numBoundingSet);
       if (!tmmbrOwner) {
-        // did not enter bounding set, no meaning to send this request
-        return BuildResult::kAborted;
+        // Did not enter bounding set, no meaning to send this request.
+        return nullptr;
       }
     }
   }
 
-  if (tmmbr_send_) {
-    rtcp::Tmmbr tmmbr;
-    tmmbr.From(ssrc_);
-    tmmbr.To(remote_ssrc_);
-    tmmbr.WithBitrateKbps(tmmbr_send_);
-    tmmbr.WithOverhead(packet_oh_send_);
+  if (!tmmbr_send_)
+    return nullptr;
 
-    PacketBuiltCallback callback(ctx);
-    if (!callback.BuildPacket(tmmbr))
-      return BuildResult::kTruncated;
-  }
-  return BuildResult::kSuccess;
+  rtcp::Tmmbr* tmmbr = new rtcp::Tmmbr();
+  tmmbr->From(ssrc_);
+  tmmbr->To(remote_ssrc_);
+  tmmbr->WithBitrateKbps(tmmbr_send_);
+  tmmbr->WithOverhead(packet_oh_send_);
+
+  return rtc::scoped_ptr<rtcp::Tmmbr>(tmmbr);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildTMMBN(RtcpContext* ctx) {
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBN(
+    const RtcpContext& ctx) {
   TMMBRSet* boundingSet = tmmbr_help_.BoundingSetToSend();
-  if (boundingSet == NULL)
-    return BuildResult::kError;
+  if (boundingSet == nullptr)
+    return nullptr;
 
-  rtcp::Tmmbn tmmbn;
-  tmmbn.From(ssrc_);
+  rtcp::Tmmbn* tmmbn = new rtcp::Tmmbn();
+  tmmbn->From(ssrc_);
   for (uint32_t i = 0; i < boundingSet->lengthOfSet(); i++) {
     if (boundingSet->Tmmbr(i) > 0) {
-      tmmbn.WithTmmbr(boundingSet->Ssrc(i), boundingSet->Tmmbr(i),
-                      boundingSet->PacketOH(i));
+      tmmbn->WithTmmbr(boundingSet->Ssrc(i), boundingSet->Tmmbr(i),
+                       boundingSet->PacketOH(i));
     }
   }
 
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(tmmbn))
-    return BuildResult::kTruncated;
-
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Tmmbn>(tmmbn);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildAPP(RtcpContext* ctx) {
-  rtcp::App app;
-  app.From(ssrc_);
-  app.WithSubType(app_sub_type_);
-  app.WithName(app_name_);
-  app.WithData(app_data_.get(), app_length_);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(app))
-    return BuildResult::kTruncated;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildAPP(const RtcpContext& ctx) {
+  rtcp::App* app = new rtcp::App();
+  app->From(ssrc_);
+  app->WithSubType(app_sub_type_);
+  app->WithName(app_name_);
+  app->WithData(app_data_.get(), app_length_);
 
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::App>(app);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildNACK(RtcpContext* ctx) {
-  // sanity
-  if (ctx->position + 16 >= IP_PACKET_SIZE) {
-    LOG(LS_WARNING) << "Failed to build NACK.";
-    return BuildResult::kTruncated;
-  }
-
-  // int size, uint16_t* nack_list
-  // add nack list
-  uint8_t FMT = 1;
-  *ctx->AllocateData(1) = 0x80 + FMT;
-  *ctx->AllocateData(1) = 205;
-
-  *ctx->AllocateData(1) = 0;
-  int nack_size_pos_ = ctx->position;
-  *ctx->AllocateData(1) = 3;  // setting it to one kNACK signal as default
-
-  // Add our own SSRC
-  ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), ssrc_);
-
-  // Add the remote SSRC
-  ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), remote_ssrc_);
-
-  // Build NACK bitmasks and write them to the RTCP message.
-  // The nack list should be sorted and not contain duplicates if one
-  // wants to build the smallest rtcp nack packet.
-  int numOfNackFields = 0;
-  int maxNackFields =
-      std::min<int>(kRtcpMaxNackFields, (IP_PACKET_SIZE - ctx->position) / 4);
-  int i = 0;
-  while (i < ctx->nack_size && numOfNackFields < maxNackFields) {
-    uint16_t nack = ctx->nack_list[i++];
-    uint16_t bitmask = 0;
-    while (i < ctx->nack_size) {
-      int shift = static_cast<uint16_t>(ctx->nack_list[i] - nack) - 1;
-      if (shift >= 0 && shift <= 15) {
-        bitmask |= (1 << shift);
-        ++i;
-      } else {
-        break;
-      }
-    }
-    // Write the sequence number and the bitmask to the packet.
-    assert(ctx->position + 4 < IP_PACKET_SIZE);
-    ByteWriter<uint16_t>::WriteBigEndian(ctx->AllocateData(2), nack);
-    ByteWriter<uint16_t>::WriteBigEndian(ctx->AllocateData(2), bitmask);
-    numOfNackFields++;
-  }
-  ctx->buffer[nack_size_pos_] = static_cast<uint8_t>(2 + numOfNackFields);
-
-  if (i != ctx->nack_size)
-    LOG(LS_WARNING) << "Nack list too large for one packet.";
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildNACK(
+    const RtcpContext& ctx) {
+  rtcp::Nack* nack = new rtcp::Nack();
+  nack->From(ssrc_);
+  nack->To(remote_ssrc_);
+  nack->WithList(ctx.nack_list_, ctx.nack_size_);
 
   // Report stats.
   NACKStringBuilder stringBuilder;
-  for (int idx = 0; idx < i; ++idx) {
-    stringBuilder.PushNACK(ctx->nack_list[idx]);
-    nack_stats_.ReportRequest(ctx->nack_list[idx]);
+  for (int idx = 0; idx < ctx.nack_size_; ++idx) {
+    stringBuilder.PushNACK(ctx.nack_list_[idx]);
+    nack_stats_.ReportRequest(ctx.nack_list_[idx]);
   }
   packet_type_counter_.nack_requests = nack_stats_.requests();
   packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
@@ -816,101 +720,66 @@ RTCPSender::BuildResult RTCPSender::BuildNACK(RtcpContext* ctx) {
   TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_NACKCount",
                     ssrc_, packet_type_counter_.nack_packets);
 
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Nack>(nack);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildBYE(RtcpContext* ctx) {
-  rtcp::Bye bye;
-  bye.From(ssrc_);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildBYE(const RtcpContext& ctx) {
+  rtcp::Bye* bye = new rtcp::Bye();
+  bye->From(ssrc_);
   for (uint32_t csrc : csrcs_)
-    bye.WithCsrc(csrc);
+    bye->WithCsrc(csrc);
 
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(bye))
-    return BuildResult::kTruncated;
-
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Bye>(bye);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildReceiverReferenceTime(
-    RtcpContext* ctx) {
-
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildReceiverReferenceTime(
+    const RtcpContext& ctx) {
   if (last_xr_rr_.size() >= RTCP_NUMBER_OF_SR)
     last_xr_rr_.erase(last_xr_rr_.begin());
   last_xr_rr_.insert(std::pair<uint32_t, int64_t>(
-      RTCPUtility::MidNtp(ctx->ntp_sec, ctx->ntp_frac),
-      Clock::NtpToMs(ctx->ntp_sec, ctx->ntp_frac)));
+      RTCPUtility::MidNtp(ctx.ntp_sec_, ctx.ntp_frac_),
+      Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_)));
 
-  rtcp::Xr xr;
-  xr.From(ssrc_);
+  rtcp::Xr* xr = new rtcp::Xr();
+  xr->From(ssrc_);
 
   rtcp::Rrtr rrtr;
-  rrtr.WithNtpSec(ctx->ntp_sec);
-  rrtr.WithNtpFrac(ctx->ntp_frac);
+  rrtr.WithNtp(NtpTime(ctx.ntp_sec_, ctx.ntp_frac_));
 
-  xr.WithRrtr(&rrtr);
+  xr->WithRrtr(&rrtr);
 
   // TODO(sprang): Merge XR report sending to contain all of RRTR, DLRR, VOIP?
 
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(xr))
-    return BuildResult::kTruncated;
-
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Xr>(xr);
 }
 
-RTCPSender::BuildResult RTCPSender::BuildDlrr(RtcpContext* ctx) {
-  rtcp::Xr xr;
-  xr.From(ssrc_);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildDlrr(
+    const RtcpContext& ctx) {
+  rtcp::Xr* xr = new rtcp::Xr();
+  xr->From(ssrc_);
 
   rtcp::Dlrr dlrr;
-  const RtcpReceiveTimeInfo& info = ctx->feedback_state.last_xr_rr;
+  const RtcpReceiveTimeInfo& info = ctx.feedback_state_.last_xr_rr;
   dlrr.WithDlrrItem(info.sourceSSRC, info.lastRR, info.delaySinceLastRR);
 
-  xr.WithDlrr(&dlrr);
+  xr->WithDlrr(&dlrr);
 
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(xr))
-    return BuildResult::kTruncated;
-
-  return BuildResult::kSuccess;
+  return rtc::scoped_ptr<rtcp::Xr>(xr);
 }
 
 // TODO(sprang): Add a unit test for this, or remove if the code isn't used.
-RTCPSender::BuildResult RTCPSender::BuildVoIPMetric(RtcpContext* ctx) {
-  rtcp::Xr xr;
-  xr.From(ssrc_);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildVoIPMetric(
+    const RtcpContext& context) {
+  rtcp::Xr* xr = new rtcp::Xr();
+  xr->From(ssrc_);
 
   rtcp::VoipMetric voip;
   voip.To(remote_ssrc_);
-  voip.LossRate(xr_voip_metric_.lossRate);
-  voip.DiscardRate(xr_voip_metric_.discardRate);
-  voip.BurstDensity(xr_voip_metric_.burstDensity);
-  voip.GapDensity(xr_voip_metric_.gapDensity);
-  voip.BurstDuration(xr_voip_metric_.burstDuration);
-  voip.GapDuration(xr_voip_metric_.gapDuration);
-  voip.RoundTripDelay(xr_voip_metric_.roundTripDelay);
-  voip.EndSystemDelay(xr_voip_metric_.endSystemDelay);
-  voip.SignalLevel(xr_voip_metric_.signalLevel);
-  voip.NoiseLevel(xr_voip_metric_.noiseLevel);
-  voip.Rerl(xr_voip_metric_.RERL);
-  voip.Gmin(xr_voip_metric_.Gmin);
-  voip.Rfactor(xr_voip_metric_.Rfactor);
-  voip.ExtRfactor(xr_voip_metric_.extRfactor);
-  voip.MosLq(xr_voip_metric_.MOSLQ);
-  voip.MosCq(xr_voip_metric_.MOSCQ);
-  voip.RxConfig(xr_voip_metric_.RXconfig);
-  voip.JbNominal(xr_voip_metric_.JBnominal);
-  voip.JbMax(xr_voip_metric_.JBmax);
-  voip.JbAbsMax(xr_voip_metric_.JBabsMax);
-
-  xr.WithVoipMetric(&voip);
-
-  PacketBuiltCallback callback(ctx);
-  if (!callback.BuildPacket(xr))
-    return BuildResult::kTruncated;
-
-  return BuildResult::kSuccess;
+  voip.WithVoipMetric(xr_voip_metric_);
+
+  xr->WithVoipMetric(&voip);
+
+  return rtc::scoped_ptr<rtcp::Xr>(xr);
 }
 
 int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
@@ -926,43 +795,59 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
 
 int32_t RTCPSender::SendCompoundRTCP(
     const FeedbackState& feedback_state,
-    const std::set<RTCPPacketType>& packetTypes,
+    const std::set<RTCPPacketType>& packet_types,
     int32_t nack_size,
     const uint16_t* nack_list,
     bool repeat,
     uint64_t pictureID) {
+  PacketContainer container(transport_);
   {
     CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
     if (method_ == RtcpMode::kOff) {
       LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
       return -1;
     }
-  }
-  uint8_t rtcp_buffer[IP_PACKET_SIZE];
-  int rtcp_length =
-      PrepareRTCP(feedback_state, packetTypes, nack_size, nack_list, repeat,
-                  pictureID, rtcp_buffer, IP_PACKET_SIZE);
 
-  // Sanity don't send empty packets.
-  if (rtcp_length <= 0)
-    return -1;
+    // We need to send our NTP even if we haven't received any reports.
+    uint32_t ntp_sec;
+    uint32_t ntp_frac;
+    clock_->CurrentNtp(ntp_sec, ntp_frac);
+    RtcpContext context(feedback_state, nack_size, nack_list, repeat, pictureID,
+                        ntp_sec, ntp_frac, &container);
+
+    PrepareReport(packet_types, feedback_state);
+
+    auto it = report_flags_.begin();
+    while (it != report_flags_.end()) {
+      auto builder_it = builders_.find(it->type);
+      RTC_DCHECK(builder_it != builders_.end());
+      if (it->is_volatile) {
+        report_flags_.erase(it++);
+      } else {
+        ++it;
+      }
 
-  return SendToNetwork(rtcp_buffer, static_cast<size_t>(rtcp_length));
-}
+      BuilderFunc func = builder_it->second;
+      rtc::scoped_ptr<rtcp::RtcpPacket> packet = (this->*func)(context);
+      if (packet.get() == nullptr)
+        return -1;
+      container.Append(packet.release());
+    }
 
-int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
-                            const std::set<RTCPPacketType>& packetTypes,
-                            int32_t nack_size,
-                            const uint16_t* nack_list,
-                            bool repeat,
-                            uint64_t pictureID,
-                            uint8_t* rtcp_buffer,
-                            int buffer_size) {
-  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+    if (packet_type_counter_observer_ != nullptr) {
+      packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
+          remote_ssrc_, packet_type_counter_);
+    }
 
-  RtcpContext context(feedback_state, nack_size, nack_list, repeat, pictureID,
-                      rtcp_buffer, buffer_size);
+    RTC_DCHECK(AllVolatileFlagsConsumed());
+  }
+
+  size_t bytes_sent = container.SendPackets();
+  return bytes_sent == 0 ? -1 : 0;
+}
 
+void RTCPSender::PrepareReport(const std::set<RTCPPacketType>& packetTypes,
+                               const FeedbackState& feedback_state) {
   // Add all flags as volatile. Non volatile entries will not be overwritten
   // and all new volatile flags added will be consumed by the end of this call.
   SetFlags(packetTypes, true);
@@ -986,9 +871,6 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
   if (IsFlagPresent(kRtcpSr) || (IsFlagPresent(kRtcpRr) && !cname_.empty()))
     SetFlag(kRtcpSdes, true);
 
-  // We need to send our NTP even if we haven't received any reports.
-  clock_->CurrentNtp(context.ntp_sec, context.ntp_frac);
-
   if (generate_report) {
     if (!sending_ && xr_send_receiver_reference_time_enabled_)
       SetFlag(kRtcpXrReceiverReferenceTime, true);
@@ -996,15 +878,9 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
       SetFlag(kRtcpXrDlrrReportBlock, true);
 
     // generate next time to send an RTCP report
-    // seeded from RTP constructor
-    int32_t random = rand() % 1000;
-    int32_t timeToNext = RTCP_INTERVAL_AUDIO_MS;
-
-    if (audio_) {
-      timeToNext = (RTCP_INTERVAL_AUDIO_MS / 2) +
-                   (RTCP_INTERVAL_AUDIO_MS * random / 1000);
-    } else {
-      uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
+    uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
+
+    if (!audio_) {
       if (sending_) {
         // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
         uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
@@ -1013,74 +889,46 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
       }
       if (minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
         minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
-      timeToNext = (minIntervalMs / 2) + (minIntervalMs * random / 1000);
     }
+    // The interval between RTCP packets is varied randomly over the
+    // range [1/2,3/2] times the calculated interval.
+    uint32_t timeToNext =
+        random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
     next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
 
     StatisticianMap statisticians =
         receive_statistics_->GetActiveStatisticians();
-    if (!statisticians.empty()) {
-      for (auto it = statisticians.begin(); it != statisticians.end(); ++it) {
-        RTCPReportBlock report_block;
-        if (PrepareReport(feedback_state, it->first, it->second,
-                          &report_block)) {
-          AddReportBlock(report_block);
-        }
-      }
-    }
-  }
-
-  auto it = report_flags_.begin();
-  while (it != report_flags_.end()) {
-    auto builder = builders_.find(it->type);
-    RTC_DCHECK(builder != builders_.end());
-    if (it->is_volatile) {
-      report_flags_.erase(it++);
-    } else {
-      ++it;
-    }
-
-    uint32_t start_position = context.position;
-    BuildResult result = (this->*(builder->second))(&context);
-    switch (result) {
-      case BuildResult::kError:
-        return -1;
-      case BuildResult::kTruncated:
-        return context.position;
-      case BuildResult::kAborted:
-        context.position = start_position;
-        FALLTHROUGH();
-      case BuildResult::kSuccess:
-        continue;
-      default:
-        abort();
+    RTC_DCHECK(report_blocks_.empty());
+    for (auto& it : statisticians) {
+      AddReportBlock(feedback_state, it.first, it.second);
     }
   }
-
-  if (packet_type_counter_observer_ != NULL) {
-    packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
-        remote_ssrc_, packet_type_counter_);
-  }
-
-  RTC_DCHECK(AllVolatileFlagsConsumed());
-
-  return context.position;
 }
 
-bool RTCPSender::PrepareReport(const FeedbackState& feedback_state,
-                               uint32_t ssrc,
-                               StreamStatistician* statistician,
-                               RTCPReportBlock* report_block) {
+bool RTCPSender::AddReportBlock(const FeedbackState& feedback_state,
+                                uint32_t ssrc,
+                                StreamStatistician* statistician) {
   // Do we have receive statistics to send?
   RtcpStatistics stats;
   if (!statistician->GetStatistics(&stats, true))
     return false;
-  report_block->fractionLost = stats.fraction_lost;
-  report_block->cumulativeLost = stats.cumulative_lost;
-  report_block->extendedHighSeqNum =
-      stats.extended_max_sequence_number;
-  report_block->jitter = stats.jitter;
-  report_block->remoteSSRC = ssrc;
+
+  if (report_blocks_.size() >= RTCP_MAX_REPORT_BLOCKS) {
+    LOG(LS_WARNING) << "Too many report blocks.";
+    return false;
+  }
+  RTC_DCHECK(report_blocks_.find(ssrc) == report_blocks_.end());
+  rtcp::ReportBlock* block = &report_blocks_[ssrc];
+  block->To(ssrc);
+  block->WithFractionLost(stats.fraction_lost);
+  if (!block->WithCumulativeLost(stats.cumulative_lost)) {
+    report_blocks_.erase(ssrc);
+    LOG(LS_WARNING) << "Cumulative lost is oversized.";
+    return false;
+  }
+  block->WithExtHighestSeqNum(stats.extended_max_sequence_number);
+  block->WithJitter(stats.jitter);
+  block->WithLastSr(feedback_state.remote_sr);
 
   // TODO(sprang): Do we really need separate time stamps for each report?
   // Get our NTP as late as possible to avoid a race.
@@ -1089,7 +937,6 @@ bool RTCPSender::PrepareReport(const FeedbackState& feedback_state,
   clock_->CurrentNtp(ntp_secs, ntp_frac);
 
   // Delay since last received report.
-  uint32_t delaySinceLastReceivedSR = 0;
   if ((feedback_state.last_rr_ntp_secs != 0) ||
       (feedback_state.last_rr_ntp_frac != 0)) {
     // Get the 16 lowest bits of seconds and the 16 highest bits of fractions.
@@ -1101,19 +948,11 @@ bool RTCPSender::PrepareReport(const FeedbackState& feedback_state,
     receiveTime <<= 16;
     receiveTime += (feedback_state.last_rr_ntp_frac & 0xffff0000) >> 16;
 
-    delaySinceLastReceivedSR = now-receiveTime;
+    block->WithDelayLastSr(now - receiveTime);
   }
-  report_block->delaySinceLastSR = delaySinceLastReceivedSR;
-  report_block->lastSR = feedback_state.remote_sr;
   return true;
 }
 
-int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
-  if (transport_->SendRtcp(dataBuffer, length))
-    return 0;
-  return -1;
-}
-
 void RTCPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   assert(csrcs.size() <= kRtpCsrcSize);
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
@@ -1203,7 +1042,7 @@ bool RTCPSender::AllVolatileFlagsConsumed() const {
 bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
   class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
    public:
-    Sender(Transport* transport)
+    explicit Sender(Transport* transport)
         : transport_(transport), send_failure_(false) {}
 
     void OnPacketReady(uint8_t* data, size_t length) override {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 9ec928363b..dd3aec4c9f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -15,13 +15,15 @@
 #include <set>
 #include <sstream>
 #include <string>
+#include <vector>
 
+#include "webrtc/base/random.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
@@ -50,277 +52,251 @@ class NACKStringBuilder {
 };
 
 class RTCPSender {
-public:
- struct FeedbackState {
-   FeedbackState();
+ public:
+  struct FeedbackState {
+    FeedbackState();
+
+    uint8_t send_payload_type;
+    uint32_t frequency_hz;
+    uint32_t packets_sent;
+    size_t media_bytes_sent;
+    uint32_t send_bitrate;
+
+    uint32_t last_rr_ntp_secs;
+    uint32_t last_rr_ntp_frac;
+    uint32_t remote_sr;
+
+    bool has_last_xr_rr;
+    RtcpReceiveTimeInfo last_xr_rr;
+
+    // Used when generating TMMBR.
+    ModuleRtpRtcpImpl* module;
+  };
+
+  RTCPSender(bool audio,
+             Clock* clock,
+             ReceiveStatistics* receive_statistics,
+             RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+             Transport* outgoing_transport);
+  virtual ~RTCPSender();
+
+  RtcpMode Status() const;
+  void SetRTCPStatus(RtcpMode method);
+
+  bool Sending() const;
+  int32_t SetSendingStatus(const FeedbackState& feedback_state,
+                           bool enabled);  // combine the functions
+
+  int32_t SetNackStatus(bool enable);
 
-   uint8_t send_payload_type;
-   uint32_t frequency_hz;
-   uint32_t packets_sent;
-   size_t media_bytes_sent;
-   uint32_t send_bitrate;
+  void SetStartTimestamp(uint32_t start_timestamp);
 
-   uint32_t last_rr_ntp_secs;
-   uint32_t last_rr_ntp_frac;
-   uint32_t remote_sr;
+  void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
 
-   bool has_last_xr_rr;
-   RtcpReceiveTimeInfo last_xr_rr;
+  void SetSSRC(uint32_t ssrc);
 
-   // Used when generating TMMBR.
-   ModuleRtpRtcpImpl* module;
- };
+  void SetRemoteSSRC(uint32_t ssrc);
 
- RTCPSender(bool audio,
-            Clock* clock,
-            ReceiveStatistics* receive_statistics,
-            RtcpPacketTypeCounterObserver* packet_type_counter_observer,
-            Transport* outgoing_transport);
- virtual ~RTCPSender();
+  int32_t SetCNAME(const char* cName);
 
- RtcpMode Status() const;
- void SetRTCPStatus(RtcpMode method);
+  int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
 
- bool Sending() const;
- int32_t SetSendingStatus(const FeedbackState& feedback_state,
-                          bool enabled);  // combine the functions
+  int32_t RemoveMixedCNAME(uint32_t SSRC);
 
- int32_t SetNackStatus(bool enable);
+  int64_t SendTimeOfSendReport(uint32_t sendReport);
 
- void SetStartTimestamp(uint32_t start_timestamp);
+  bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
 
- void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
+  bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
 
- void SetSSRC(uint32_t ssrc);
+  int32_t SendRTCP(const FeedbackState& feedback_state,
+                   RTCPPacketType packetType,
+                   int32_t nackSize = 0,
+                   const uint16_t* nackList = 0,
+                   bool repeat = false,
+                   uint64_t pictureID = 0);
 
- void SetRemoteSSRC(uint32_t ssrc);
+  int32_t SendCompoundRTCP(const FeedbackState& feedback_state,
+                           const std::set<RTCPPacketType>& packetTypes,
+                           int32_t nackSize = 0,
+                           const uint16_t* nackList = 0,
+                           bool repeat = false,
+                           uint64_t pictureID = 0);
 
- int32_t SetCNAME(const char* cName);
+  bool REMB() const;
 
- int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
+  void SetREMBStatus(bool enable);
 
- int32_t RemoveMixedCNAME(uint32_t SSRC);
+  void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
 
- int64_t SendTimeOfSendReport(uint32_t sendReport);
+  bool TMMBR() const;
 
- bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
+  void SetTMMBRStatus(bool enable);
 
- bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
+  int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit);
 
- int32_t SendRTCP(const FeedbackState& feedback_state,
-                  RTCPPacketType packetType,
-                  int32_t nackSize = 0,
-                  const uint16_t* nackList = 0,
-                  bool repeat = false,
-                  uint64_t pictureID = 0);
-
- int32_t SendCompoundRTCP(const FeedbackState& feedback_state,
-                          const std::set<RTCPPacketType>& packetTypes,
-                          int32_t nackSize = 0,
-                          const uint16_t* nackList = 0,
-                          bool repeat = false,
-                          uint64_t pictureID = 0);
-
- bool REMB() const;
-
- void SetREMBStatus(bool enable);
-
- void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
-
- bool TMMBR() const;
-
- void SetTMMBRStatus(bool enable);
-
- int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit);
-
- int32_t SetApplicationSpecificData(uint8_t subType,
-                                    uint32_t name,
-                                    const uint8_t* data,
-                                    uint16_t length);
- int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
-
- void SendRtcpXrReceiverReferenceTime(bool enable);
-
- bool RtcpXrReceiverReferenceTime() const;
-
- void SetCsrcs(const std::vector<uint32_t>& csrcs);
-
- void SetTargetBitrate(unsigned int target_bitrate);
- bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
-
-private:
- struct RtcpContext;
-
- // The BuildResult indicates the outcome of a call to a builder method,
- // constructing a part of an RTCP packet:
- //
- // kError
- //   Building RTCP packet failed, propagate error out to caller.
- // kAbort
- //   The (partial) block being build should not be included. Reset current
- //   buffer position to the state before the method call and proceed to the
- //   next packet type.
- // kTruncated
- //   There is not enough room in the buffer to fit the data being constructed.
- //   (IP packet is full). Proceed to the next packet type, and call this
- //   method again when a new buffer has been allocated.
- // TODO(sprang): Actually allocate multiple packets if needed.
- // kSuccess
- //   Data has been successfully placed in the buffer.
-
- enum class BuildResult { kError, kAborted, kTruncated, kSuccess };
-
- int32_t SendToNetwork(const uint8_t* dataBuffer, size_t length);
-
- int32_t AddReportBlock(const RTCPReportBlock& report_block)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
-
- bool PrepareReport(const FeedbackState& feedback_state,
-                    uint32_t ssrc,
-                    StreamStatistician* statistician,
-                    RTCPReportBlock* report_block);
-
- int PrepareRTCP(const FeedbackState& feedback_state,
-                 const std::set<RTCPPacketType>& packetTypes,
-                 int32_t nackSize,
-                 const uint16_t* nackList,
-                 bool repeat,
-                 uint64_t pictureID,
-                 uint8_t* rtcp_buffer,
-                 int buffer_size);
-
- BuildResult BuildSR(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildRR(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildSDES(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildPLI(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildREMB(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildTMMBR(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildTMMBN(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildAPP(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildVoIPMetric(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildBYE(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildFIR(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildSLI(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildRPSI(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildNACK(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildReceiverReferenceTime(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildDlrr(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
-
-private:
- const bool audio_;
- Clock* const clock_;
- RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
-
- Transport* const transport_;
-
- rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
- bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
- bool sending_ GUARDED_BY(critical_section_rtcp_sender_);
- bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
-
- int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
-
- uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
- uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
- int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_);
- uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
- // SSRC that we receive on our RTP channel
- uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
- std::string cname_ GUARDED_BY(critical_section_rtcp_sender_);
-
- ReceiveStatistics* receive_statistics_
-     GUARDED_BY(critical_section_rtcp_sender_);
- std::map<uint32_t, rtcp::ReportBlock> report_blocks_
-     GUARDED_BY(critical_section_rtcp_sender_);
- std::map<uint32_t, std::string> csrc_cnames_
-     GUARDED_BY(critical_section_rtcp_sender_);
-
- // Sent
- uint32_t last_send_report_[RTCP_NUMBER_OF_SR] GUARDED_BY(
-     critical_section_rtcp_sender_);  // allow packet loss and RTT above 1 sec
- int64_t last_rtcp_time_[RTCP_NUMBER_OF_SR] GUARDED_BY(
-     critical_section_rtcp_sender_);
-
- // Sent XR receiver reference time report.
- // <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
- std::map<uint32_t, int64_t> last_xr_rr_
-     GUARDED_BY(critical_section_rtcp_sender_);
-
- // send CSRCs
- std::vector<uint32_t> csrcs_ GUARDED_BY(critical_section_rtcp_sender_);
-
- // Full intra request
- uint8_t sequence_number_fir_ GUARDED_BY(critical_section_rtcp_sender_);
-
- // REMB
- uint32_t remb_bitrate_ GUARDED_BY(critical_section_rtcp_sender_);
- std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(critical_section_rtcp_sender_);
-
- TMMBRHelp tmmbr_help_ GUARDED_BY(critical_section_rtcp_sender_);
- uint32_t tmmbr_send_ GUARDED_BY(critical_section_rtcp_sender_);
- uint32_t packet_oh_send_ GUARDED_BY(critical_section_rtcp_sender_);
-
- // APP
- uint8_t app_sub_type_ GUARDED_BY(critical_section_rtcp_sender_);
- uint32_t app_name_ GUARDED_BY(critical_section_rtcp_sender_);
- rtc::scoped_ptr<uint8_t[]> app_data_ GUARDED_BY(critical_section_rtcp_sender_);
- uint16_t app_length_ GUARDED_BY(critical_section_rtcp_sender_);
-
- // True if sending of XR Receiver reference time report is enabled.
- bool xr_send_receiver_reference_time_enabled_
-     GUARDED_BY(critical_section_rtcp_sender_);
-
- // XR VoIP metric
- RTCPVoIPMetric xr_voip_metric_ GUARDED_BY(critical_section_rtcp_sender_);
-
- RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
- RtcpPacketTypeCounter packet_type_counter_
-     GUARDED_BY(critical_section_rtcp_sender_);
-
- RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
-
- void SetFlag(RTCPPacketType type, bool is_volatile)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- bool IsFlagPresent(RTCPPacketType type) const
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- bool ConsumeFlag(RTCPPacketType type, bool forced = false)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- bool AllVolatileFlagsConsumed() const
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- struct ReportFlag {
-   ReportFlag(RTCPPacketType type, bool is_volatile)
-       : type(type), is_volatile(is_volatile) {}
-   bool operator<(const ReportFlag& flag) const { return type < flag.type; }
-   bool operator==(const ReportFlag& flag) const { return type == flag.type; }
-   const RTCPPacketType type;
-   const bool is_volatile;
- };
-
- std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_);
-
- typedef BuildResult (RTCPSender::*Builder)(RtcpContext*);
- std::map<RTCPPacketType, Builder> builders_;
-
- class PacketBuiltCallback;
+  int32_t SetApplicationSpecificData(uint8_t subType,
+                                     uint32_t name,
+                                     const uint8_t* data,
+                                     uint16_t length);
+  int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
+
+  void SendRtcpXrReceiverReferenceTime(bool enable);
+
+  bool RtcpXrReceiverReferenceTime() const;
+
+  void SetCsrcs(const std::vector<uint32_t>& csrcs);
+
+  void SetTargetBitrate(unsigned int target_bitrate);
+  bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
+
+ private:
+  class RtcpContext;
+
+  // Determine which RTCP messages should be sent and setup flags.
+  void PrepareReport(const std::set<RTCPPacketType>& packetTypes,
+                     const FeedbackState& feedback_state)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+
+  bool AddReportBlock(const FeedbackState& feedback_state,
+                      uint32_t ssrc,
+                      StreamStatistician* statistician)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBN(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildAPP(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildVoIPMetric(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildBYE(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildFIR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildSLI(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildRPSI(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildNACK(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildReceiverReferenceTime(
+      const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+
+ private:
+  const bool audio_;
+  Clock* const clock_;
+  Random random_ GUARDED_BY(critical_section_rtcp_sender_);
+  RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  Transport* const transport_;
+
+  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
+  bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
+  bool sending_ GUARDED_BY(critical_section_rtcp_sender_);
+  bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
+  int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
+  // SSRC that we receive on our RTP channel
+  uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
+  std::string cname_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  ReceiveStatistics* receive_statistics_
+      GUARDED_BY(critical_section_rtcp_sender_);
+  std::map<uint32_t, rtcp::ReportBlock> report_blocks_
+      GUARDED_BY(critical_section_rtcp_sender_);
+  std::map<uint32_t, std::string> csrc_cnames_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  // Sent
+  uint32_t last_send_report_[RTCP_NUMBER_OF_SR] GUARDED_BY(
+      critical_section_rtcp_sender_);  // allow packet loss and RTT above 1 sec
+  int64_t last_rtcp_time_[RTCP_NUMBER_OF_SR] GUARDED_BY(
+      critical_section_rtcp_sender_);
+
+  // Sent XR receiver reference time report.
+  // <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
+  std::map<uint32_t, int64_t> last_xr_rr_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  // send CSRCs
+  std::vector<uint32_t> csrcs_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // Full intra request
+  uint8_t sequence_number_fir_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // REMB
+  uint32_t remb_bitrate_ GUARDED_BY(critical_section_rtcp_sender_);
+  std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  TMMBRHelp tmmbr_help_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t tmmbr_send_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t packet_oh_send_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // APP
+  uint8_t app_sub_type_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t app_name_ GUARDED_BY(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<uint8_t[]> app_data_
+      GUARDED_BY(critical_section_rtcp_sender_);
+  uint16_t app_length_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // True if sending of XR Receiver reference time report is enabled.
+  bool xr_send_receiver_reference_time_enabled_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  // XR VoIP metric
+  RTCPVoIPMetric xr_voip_metric_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
+  RtcpPacketTypeCounter packet_type_counter_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  void SetFlag(RTCPPacketType type, bool is_volatile)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  bool IsFlagPresent(RTCPPacketType type) const
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  bool ConsumeFlag(RTCPPacketType type, bool forced = false)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  bool AllVolatileFlagsConsumed() const
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  struct ReportFlag {
+    ReportFlag(RTCPPacketType type, bool is_volatile)
+        : type(type), is_volatile(is_volatile) {}
+    bool operator<(const ReportFlag& flag) const { return type < flag.type; }
+    bool operator==(const ReportFlag& flag) const { return type == flag.type; }
+    const RTCPPacketType type;
+    const bool is_volatile;
+  };
+
+  std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  typedef rtc::scoped_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
+      const RtcpContext&);
+  std::map<RTCPPacketType, BuilderFunc> builders_;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
index d2b80438cc..e19499612d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
@@ -8,9 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 #include <assert.h>
 #include <math.h>   // ceil
@@ -19,6 +17,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
 
@@ -42,8 +41,8 @@ void NackStats::ReportRequest(uint16_t sequence_number) {
 
 uint32_t MidNtp(uint32_t ntp_sec, uint32_t ntp_frac) {
   return (ntp_sec << 16) + (ntp_frac >> 16);
-}  // end RTCPUtility
 }
+}  // namespace RTCPUtility
 
 // RTCPParserV2 : currently read only
 RTCPUtility::RTCPParserV2::RTCPParserV2(const uint8_t* rtcpData,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
index f05d512919..0b03ceb56e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
@@ -14,7 +14,7 @@
 #include <stddef.h> // size_t, ptrdiff_t
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
 
@@ -487,6 +487,6 @@ class RTCPPacketIterator {
 
   RtcpCommonHeader _header;
 };
-}  // RTCPUtility
+}  // namespace RTCPUtility
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
index 541f522f8d..80f961bd1e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -11,6 +11,7 @@
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
@@ -41,8 +42,12 @@ template <typename T> void ClearList(std::list<T*>* my_list) {
 class RtpFecTest : public ::testing::Test {
  protected:
   RtpFecTest()
-      : fec_(new ForwardErrorCorrection()), ssrc_(rand()), fec_seq_num_(0) {}
+      : random_(0xfec133700742),
+        fec_(new ForwardErrorCorrection()),
+        ssrc_(random_.Rand<uint32_t>()),
+        fec_seq_num_(0) {}
 
+  webrtc::Random random_;
   ForwardErrorCorrection* fec_;
   int ssrc_;
   uint16_t fec_seq_num_;
@@ -891,22 +896,20 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets,
   assert(num_media_packets > 0);
   ForwardErrorCorrection::Packet* media_packet = NULL;
   int sequence_number = start_seq_num;
-  int time_stamp = rand();
+  int time_stamp = random_.Rand<int>();
 
   for (int i = 0; i < num_media_packets; ++i) {
     media_packet = new ForwardErrorCorrection::Packet;
     media_packet_list_.push_back(media_packet);
-    media_packet->length = static_cast<size_t>(
-        (static_cast<float>(rand()) / RAND_MAX) *
-        (IP_PACKET_SIZE - kRtpHeaderSize - kTransportOverhead -
-         ForwardErrorCorrection::PacketOverhead()));
+    const uint32_t kMinPacketSize = kRtpHeaderSize;
+    const uint32_t kMaxPacketSize = IP_PACKET_SIZE - kRtpHeaderSize -
+                                    kTransportOverhead -
+                                    ForwardErrorCorrection::PacketOverhead();
+    media_packet->length = random_.Rand(kMinPacketSize, kMaxPacketSize);
 
-    if (media_packet->length < kRtpHeaderSize) {
-      media_packet->length = kRtpHeaderSize;
-    }
     // Generate random values for the first 2 bytes
-    media_packet->data[0] = static_cast<uint8_t>(rand() % 256);
-    media_packet->data[1] = static_cast<uint8_t>(rand() % 256);
+    media_packet->data[0] = random_.Rand<uint8_t>();
+    media_packet->data[1] = random_.Rand<uint8_t>();
 
     // The first two bits are assumed to be 10 by the FEC encoder.
     // In fact the FEC decoder will set the two first bits to 10 regardless of
@@ -929,7 +932,7 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets,
 
     // Generate random values for payload.
     for (size_t j = 12; j < media_packet->length; ++j) {
-      media_packet->data[j] = static_cast<uint8_t>(rand() % 256);
+      media_packet->data[j] = random_.Rand<uint8_t>();
     }
     sequence_number++;
   }
@@ -940,5 +943,5 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets,
 }
 
 int RtpFecTest::ConstructMediaPackets(int num_media_packets) {
-  return ConstructMediaPacketsSeqNum(num_media_packets, rand());
+  return ConstructMediaPacketsSeqNum(num_media_packets, random_.Rand<int>());
 }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h
index 18225f9bb4..3519499248 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -14,8 +14,8 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index aeef44364a..c422577c81 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -11,7 +11,7 @@
 #include <string.h>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 1a14b5554a..d29e3d4f21 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -13,7 +13,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 39b64c6ffa..b47e9b9359 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -11,7 +11,7 @@
 #include <string>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
index 63db349c74..d62ecba85f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -30,7 +30,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
index 2fe963251f..668476833d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -19,7 +19,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
index 0e76a8eae8..d2f22d5044 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
@@ -47,10 +47,6 @@ int16_t Tl0PicIdxField(const RTPVideoHeaderVP9& hdr, uint8_t def) {
   return (hdr.tl0_pic_idx == kNoTl0PicIdx) ? def : hdr.tl0_pic_idx;
 }
 
-uint8_t GofIdxField(const RTPVideoHeaderVP9& hdr, uint8_t def) {
-  return (hdr.gof_idx == kNoGofIdx) ? def : hdr.gof_idx;
-}
-
 // Picture ID:
 //
 //      +-+-+-+-+-+-+-+-+
@@ -74,19 +70,17 @@ bool PictureIdPresent(const RTPVideoHeaderVP9& hdr) {
 // Flexible mode (F=1):     Non-flexible mode (F=0):
 //
 //      +-+-+-+-+-+-+-+-+   +-+-+-+-+-+-+-+-+
-// L:   |  T  |U|  S  |D|   |GOF_IDX|  S  |D|
+// L:   |  T  |U|  S  |D|   |  T  |U|  S  |D|
 //      +-+-+-+-+-+-+-+-+   +-+-+-+-+-+-+-+-+
 //                          |   TL0PICIDX   |
 //                          +-+-+-+-+-+-+-+-+
 //
 size_t LayerInfoLength(const RTPVideoHeaderVP9& hdr) {
-  if (hdr.flexible_mode) {
-    return (hdr.temporal_idx == kNoTemporalIdx &&
-            hdr.spatial_idx == kNoSpatialIdx) ? 0 : 1;
-  } else {
-    return (hdr.gof_idx == kNoGofIdx &&
-            hdr.spatial_idx == kNoSpatialIdx) ? 0 : 2;
+  if (hdr.temporal_idx == kNoTemporalIdx &&
+      hdr.spatial_idx == kNoSpatialIdx) {
+    return 0;
   }
+  return hdr.flexible_mode ? 1 : 2;
 }
 
 bool LayerInfoPresent(const RTPVideoHeaderVP9& hdr) {
@@ -198,8 +192,8 @@ bool WritePictureId(const RTPVideoHeaderVP9& vp9,
 // L:   |  T  |U|  S  |D|
 //      +-+-+-+-+-+-+-+-+
 //
-bool WriteLayerInfoFlexibleMode(const RTPVideoHeaderVP9& vp9,
-                                rtc::BitBufferWriter* writer) {
+bool WriteLayerInfoCommon(const RTPVideoHeaderVP9& vp9,
+                          rtc::BitBufferWriter* writer) {
   RETURN_FALSE_ON_ERROR(writer->WriteBits(TemporalIdxField(vp9, 0), 3));
   RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.temporal_up_switch ? 1 : 0, 1));
   RETURN_FALSE_ON_ERROR(writer->WriteBits(SpatialIdxField(vp9, 0), 3));
@@ -210,27 +204,26 @@ bool WriteLayerInfoFlexibleMode(const RTPVideoHeaderVP9& vp9,
 // Non-flexible mode (F=0):
 //
 //      +-+-+-+-+-+-+-+-+
-// L:   |GOF_IDX|  S  |D|
+// L:   |  T  |U|  S  |D|
 //      +-+-+-+-+-+-+-+-+
 //      |   TL0PICIDX   |
 //      +-+-+-+-+-+-+-+-+
 //
 bool WriteLayerInfoNonFlexibleMode(const RTPVideoHeaderVP9& vp9,
                                    rtc::BitBufferWriter* writer) {
-  RETURN_FALSE_ON_ERROR(writer->WriteBits(GofIdxField(vp9, 0), 4));
-  RETURN_FALSE_ON_ERROR(writer->WriteBits(SpatialIdxField(vp9, 0), 3));
-  RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.inter_layer_predicted ? 1: 0, 1));
   RETURN_FALSE_ON_ERROR(writer->WriteUInt8(Tl0PicIdxField(vp9, 0)));
   return true;
 }
 
 bool WriteLayerInfo(const RTPVideoHeaderVP9& vp9,
                     rtc::BitBufferWriter* writer) {
-  if (vp9.flexible_mode) {
-    return WriteLayerInfoFlexibleMode(vp9, writer);
-  } else {
-    return WriteLayerInfoNonFlexibleMode(vp9, writer);
-  }
+  if (!WriteLayerInfoCommon(vp9, writer))
+    return false;
+
+  if (vp9.flexible_mode)
+    return true;
+
+  return WriteLayerInfoNonFlexibleMode(vp9, writer);
 }
 
 // Reference indices:
@@ -246,7 +239,7 @@ bool WriteRefIndices(const RTPVideoHeaderVP9& vp9,
       vp9.num_ref_pics == 0 || vp9.num_ref_pics > kMaxVp9RefPics) {
     return false;
   }
-  for (size_t i = 0; i < vp9.num_ref_pics; ++i) {
+  for (uint8_t i = 0; i < vp9.num_ref_pics; ++i) {
     bool n_bit = !(i == vp9.num_ref_pics - 1);
     RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.pid_diff[i], 7));
     RETURN_FALSE_ON_ERROR(writer->WriteBits(n_bit ? 1 : 0, 1));
@@ -301,7 +294,7 @@ bool WriteSsData(const RTPVideoHeaderVP9& vp9, rtc::BitBufferWriter* writer) {
         writer->WriteBits(vp9.gof.temporal_up_switch[i] ? 1 : 0, 1));
     RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.gof.num_ref_pics[i], 2));
     RETURN_FALSE_ON_ERROR(writer->WriteBits(kReservedBitValue0, 2));
-    for (size_t r = 0; r < vp9.gof.num_ref_pics[i]; ++r) {
+    for (uint8_t r = 0; r < vp9.gof.num_ref_pics[i]; ++r) {
       RETURN_FALSE_ON_ERROR(writer->WriteUInt8(vp9.gof.pid_diff[i][r]));
     }
   }
@@ -337,8 +330,7 @@ bool ParsePictureId(rtc::BitBuffer* parser, RTPVideoHeaderVP9* vp9) {
 // L:   |  T  |U|  S  |D|
 //      +-+-+-+-+-+-+-+-+
 //
-bool ParseLayerInfoFlexibleMode(rtc::BitBuffer* parser,
-                                RTPVideoHeaderVP9* vp9) {
+bool ParseLayerInfoCommon(rtc::BitBuffer* parser, RTPVideoHeaderVP9* vp9) {
   uint32_t t, u_bit, s, d_bit;
   RETURN_FALSE_ON_ERROR(parser->ReadBits(&t, 3));
   RETURN_FALSE_ON_ERROR(parser->ReadBits(&u_bit, 1));
@@ -354,32 +346,27 @@ bool ParseLayerInfoFlexibleMode(rtc::BitBuffer* parser,
 // Layer indices (non-flexible mode):
 //
 //      +-+-+-+-+-+-+-+-+
-// L:   |GOF_IDX|  S  |D|
+// L:   |  T  |U|  S  |D|
 //      +-+-+-+-+-+-+-+-+
 //      |   TL0PICIDX   |
 //      +-+-+-+-+-+-+-+-+
 //
 bool ParseLayerInfoNonFlexibleMode(rtc::BitBuffer* parser,
                                    RTPVideoHeaderVP9* vp9) {
-  uint32_t gof_idx, s, d_bit;
   uint8_t tl0picidx;
-  RETURN_FALSE_ON_ERROR(parser->ReadBits(&gof_idx, 4));
-  RETURN_FALSE_ON_ERROR(parser->ReadBits(&s, 3));
-  RETURN_FALSE_ON_ERROR(parser->ReadBits(&d_bit, 1));
   RETURN_FALSE_ON_ERROR(parser->ReadUInt8(&tl0picidx));
-  vp9->gof_idx = gof_idx;
-  vp9->spatial_idx = s;
-  vp9->inter_layer_predicted = d_bit ? true : false;
   vp9->tl0_pic_idx = tl0picidx;
   return true;
 }
 
 bool ParseLayerInfo(rtc::BitBuffer* parser, RTPVideoHeaderVP9* vp9) {
-  if (vp9->flexible_mode) {
-    return ParseLayerInfoFlexibleMode(parser, vp9);
-  } else {
-    return ParseLayerInfoNonFlexibleMode(parser, vp9);
-  }
+  if (!ParseLayerInfoCommon(parser, vp9))
+    return false;
+
+  if (vp9->flexible_mode)
+    return true;
+
+  return ParseLayerInfoNonFlexibleMode(parser, vp9);
 }
 
 // Reference indices:
@@ -466,7 +453,7 @@ bool ParseSsData(rtc::BitBuffer* parser, RTPVideoHeaderVP9* vp9) {
     vp9->gof.temporal_up_switch[i] = u_bit ? true : false;
     vp9->gof.num_ref_pics[i] = r;
 
-    for (size_t p = 0; p < vp9->gof.num_ref_pics[i]; ++p) {
+    for (uint8_t p = 0; p < vp9->gof.num_ref_pics[i]; ++p) {
       uint8_t p_diff;
       RETURN_FALSE_ON_ERROR(parser->ReadUInt8(&p_diff));
       vp9->gof.pid_diff[i][p] = p_diff;
@@ -604,7 +591,7 @@ bool RtpPacketizerVp9::NextPacket(uint8_t* buffer,
 //      +-+-+-+-+-+-+-+-+
 // M:   | EXTENDED PID  | (RECOMMENDED)
 //      +-+-+-+-+-+-+-+-+
-// L:   |GOF_IDX|  S  |D| (CONDITIONALLY RECOMMENDED)
+// L:   |  T  |U|  S  |D| (CONDITIONALLY RECOMMENDED)
 //      +-+-+-+-+-+-+-+-+
 //      |   TL0PICIDX   | (CONDITIONALLY REQUIRED)
 //      +-+-+-+-+-+-+-+-+
@@ -738,7 +725,8 @@ bool RtpDepacketizerVp9::Parse(ParsedPayload* parsed_payload,
       parsed_payload->type.Video.height = vp9->height[0];
     }
   }
-  parsed_payload->type.Video.isFirstPacket = b_bit && (vp9->spatial_idx == 0);
+  parsed_payload->type.Video.isFirstPacket =
+      b_bit && (!l_bit || !vp9->inter_layer_predicted);
 
   uint64_t rem_bits = parser.RemainingBitCount();
   assert(rem_bits % 8 == 0);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
index abce7e7791..3feca4392a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
@@ -25,7 +25,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
index 66ab5cdb71..5bbafe459d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
@@ -55,7 +55,7 @@ void VerifyHeader(const RTPVideoHeaderVP9& expected,
                 actual.gof.temporal_up_switch[i]);
       EXPECT_EQ(expected.gof.temporal_idx[i], actual.gof.temporal_idx[i]);
       EXPECT_EQ(expected.gof.num_ref_pics[i], actual.gof.num_ref_pics[i]);
-      for (size_t j = 0; j < expected.gof.num_ref_pics[i]; j++) {
+      for (uint8_t j = 0; j < expected.gof.num_ref_pics[i]; j++) {
         EXPECT_EQ(expected.gof.pid_diff[i][j], actual.gof.pid_diff[i][j]);
       }
     }
@@ -112,7 +112,7 @@ void ParseAndCheckPacket(const uint8_t* packet,
 //        +-+-+-+-+-+-+-+-+
 //   M:   | EXTENDED PID  | (RECOMMENDED)
 //        +-+-+-+-+-+-+-+-+
-//   L:   |GOF_IDX|  S  |D| (CONDITIONALLY RECOMMENDED)
+//   L:   |  T  |U|  S  |D| (CONDITIONALLY RECOMMENDED)
 //        +-+-+-+-+-+-+-+-+
 //        |   TL0PICIDX   | (CONDITIONALLY REQUIRED)
 //        +-+-+-+-+-+-+-+-+
@@ -255,7 +255,8 @@ TEST_F(RtpPacketizerVp9Test, TestLayerInfoWithNonFlexibleMode) {
   const size_t kFrameSize = 30;
   const size_t kPacketSize = 25;
 
-  expected_.gof_idx = 3;
+  expected_.temporal_idx = 3;
+  expected_.temporal_up_switch = true;  // U
   expected_.num_spatial_layers = 3;
   expected_.spatial_idx = 2;
   expected_.inter_layer_predicted = true;  // D
@@ -264,9 +265,9 @@ TEST_F(RtpPacketizerVp9Test, TestLayerInfoWithNonFlexibleMode) {
 
   // Two packets:
   //    | I:0, P:0, L:1, F:0, B:1, E:0, V:0 | (3hdr + 15 payload)
-  // L: | GOF_IDX:3, S:2, D:1 | TL0PICIDX:117 |
+  // L: | T:3, U:1, S:2, D:1 | TL0PICIDX:117 |
   //    | I:0, P:0, L:1, F:0, B:0, E:1, V:0 | (3hdr + 15 payload)
-  // L: | GOF_IDX:3, S:2, D:1 | TL0PICIDX:117 |
+  // L: | T:3, U:1, S:2, D:1 | TL0PICIDX:117 |
   const size_t kExpectedHdrSizes[] = {3, 3};
   const size_t kExpectedSizes[] = {18, 18};
   const size_t kExpectedNum = GTEST_ARRAY_SIZE_(kExpectedSizes);
@@ -505,16 +506,20 @@ TEST_F(RtpDepacketizerVp9Test, ParseTwoBytePictureId) {
 
 TEST_F(RtpDepacketizerVp9Test, ParseLayerInfoWithNonFlexibleMode) {
   const uint8_t kHeaderLength = 3;
-  const uint8_t kGofIdx = 7;
+  const uint8_t kTemporalIdx = 2;
+  const uint8_t kUbit = 1;
   const uint8_t kSpatialIdx = 1;
   const uint8_t kDbit = 1;
   const uint8_t kTl0PicIdx = 17;
   uint8_t packet[13] = {0};
   packet[0] = 0x20;  // I:0 P:0 L:1 F:0 B:0 E:0 V:0 R:0
-  packet[1] = (kGofIdx << 4) | (kSpatialIdx << 1) | kDbit;  // GOF_IDX:7 S:1 D:1
-  packet[2] = kTl0PicIdx;                                   // TL0PICIDX:17
+  packet[1] = (kTemporalIdx << 5) | (kUbit << 4) | (kSpatialIdx << 1) | kDbit;
+  packet[2] = kTl0PicIdx;
 
-  expected_.gof_idx = kGofIdx;
+  // T:2 U:1 S:1 D:1
+  // TL0PICIDX:17
+  expected_.temporal_idx = kTemporalIdx;
+  expected_.temporal_up_switch = kUbit ? true : false;
   expected_.spatial_idx = kSpatialIdx;
   expected_.inter_layer_predicted = kDbit ? true : false;
   expected_.tl0_pic_idx = kTl0PicIdx;
@@ -545,9 +550,9 @@ TEST_F(RtpDepacketizerVp9Test, ParseLayerInfoWithFlexibleMode) {
 TEST_F(RtpDepacketizerVp9Test, ParseRefIdx) {
   const uint8_t kHeaderLength = 6;
   const int16_t kPictureId = 17;
-  const int16_t kPdiff1 = 17;
-  const int16_t kPdiff2 = 18;
-  const int16_t kPdiff3 = 127;
+  const uint8_t kPdiff1 = 17;
+  const uint8_t kPdiff2 = 18;
+  const uint8_t kPdiff3 = 127;
   uint8_t packet[13] = {0};
   packet[0] = 0xD8;  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 R:0
   packet[1] = 0x80 | ((kPictureId >> 8) & 0x7F);  // Two byte pictureID.
@@ -577,7 +582,7 @@ TEST_F(RtpDepacketizerVp9Test, ParseRefIdx) {
 }
 
 TEST_F(RtpDepacketizerVp9Test, ParseRefIdxFailsWithNoPictureId) {
-  const int16_t kPdiff = 3;
+  const uint8_t kPdiff = 3;
   uint8_t packet[13] = {0};
   packet[0] = 0x58;            // I:0 P:1 L:0 F:1 B:1 E:0 V:0 R:0
   packet[1] = (kPdiff << 1);   // P,F:  P_DIFF:3 N:0
@@ -587,7 +592,7 @@ TEST_F(RtpDepacketizerVp9Test, ParseRefIdxFailsWithNoPictureId) {
 }
 
 TEST_F(RtpDepacketizerVp9Test, ParseRefIdxFailsWithTooManyRefPics) {
-  const int16_t kPdiff = 3;
+  const uint8_t kPdiff = 3;
   uint8_t packet[13] = {0};
   packet[0] = 0xD8;                  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 R:0
   packet[1] = kMaxOneBytePictureId;  // I:    PICTURE ID:127
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 7be3c2e5c4..342e38a1f2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
 
 #include <map>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -28,7 +28,7 @@ const size_t kVideoRotationLength = 2;
 const size_t kTransportSequenceNumberLength = 3;
 
 struct HeaderExtension {
-  HeaderExtension(RTPExtensionType extension_type)
+  explicit HeaderExtension(RTPExtensionType extension_type)
       : type(extension_type), length(0), active(true) {
     Init();
   }
@@ -112,6 +112,7 @@ class RtpHeaderExtensionMap {
   int32_t Register(const RTPExtensionType type, const uint8_t id, bool active);
   std::map<uint8_t, HeaderExtension*> extensionMap_;
 };
-}
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
 
-#endif // WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
index 520cf7a962..ca37750621 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
@@ -15,7 +15,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index 266bad8858..d4cbe544cc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
@@ -58,7 +58,7 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
     rtp_header_extension_map_.GetCopy(&map);
   }
 
-  const bool valid_rtpheader = rtp_parser.Parse(*header, &map);
+  const bool valid_rtpheader = rtp_parser.Parse(header, &map);
   if (!valid_rtpheader) {
     return false;
   }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index aa941d63ff..49f9d8530a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -13,6 +13,8 @@
 #include <assert.h>
 #include <stdlib.h>
 #include <string.h>   // memset
+
+#include <algorithm>
 #include <limits>
 #include <set>
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index e97d11eeaa..8e1a732b19 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -10,14 +10,14 @@
  *  Class for storing RTP packets.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
 
 #include <vector>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -29,7 +29,7 @@ static const size_t kMaxHistoryCapacity = 9600;
 
 class RTPPacketHistory {
  public:
-  RTPPacketHistory(Clock* clock);
+  explicit RTPPacketHistory(Clock* clock);
   ~RTPPacketHistory();
 
   void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
@@ -101,4 +101,4 @@ class RTPPacketHistory {
   std::vector<StoredPacket> stored_packets_ GUARDED_BY(critsect_);
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index 00a6ac7ed2..a406d8bc9b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -12,10 +12,9 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -29,7 +28,7 @@ class RtpPacketHistoryTest : public ::testing::Test {
   ~RtpPacketHistoryTest() {
     delete hist_;
   }
-  
+
   SimulatedClock fake_clock_;
   RTPPacketHistory* hist_;
   enum {kPayload = 127};
@@ -54,7 +53,7 @@ class RtpPacketHistoryTest : public ::testing::Test {
     array[(*cur_pos)++] = ssrc >> 16;
     array[(*cur_pos)++] = ssrc >> 8;
     array[(*cur_pos)++] = ssrc;
-  } 
+  }
 };
 
 TEST_F(RtpPacketHistoryTest, SetStoreStatus) {
@@ -268,6 +267,7 @@ TEST_F(RtpPacketHistoryTest, DynamicExpansion) {
 }
 
 TEST_F(RtpPacketHistoryTest, FullExpansion) {
+  static const int kSendSidePacketHistorySize = 600;
   hist_->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
   size_t len;
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 38d1450c23..ce0bcd7fed 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
@@ -40,7 +40,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payload_type,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate,
     bool* created_new_payload) {
   assert(payload_type >= 0);
@@ -139,7 +139,7 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const size_t payload_name_length,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate) {
   RtpUtility::PayloadTypeMap::iterator iterator = payload_type_map_.begin();
   for (; iterator != payload_type_map_.end(); ++iterator) {
@@ -171,7 +171,7 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
 int32_t RTPPayloadRegistry::ReceivePayloadType(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate,
     int8_t* payload_type) const {
   assert(payload_type);
@@ -343,17 +343,16 @@ bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type,
 
 int RTPPayloadRegistry::GetPayloadTypeFrequency(
     uint8_t payload_type) const {
-  RtpUtility::Payload* payload;
-  if (!PayloadTypeToPayload(payload_type, payload)) {
+  const RtpUtility::Payload* payload = PayloadTypeToPayload(payload_type);
+  if (!payload) {
     return -1;
   }
   CriticalSectionScoped cs(crit_sect_.get());
   return rtp_payload_strategy_->GetPayloadTypeFrequency(*payload);
 }
 
-bool RTPPayloadRegistry::PayloadTypeToPayload(
-    const uint8_t payload_type,
-    RtpUtility::Payload*& payload) const {
+const RtpUtility::Payload* RTPPayloadRegistry::PayloadTypeToPayload(
+    uint8_t payload_type) const {
   CriticalSectionScoped cs(crit_sect_.get());
 
   RtpUtility::PayloadTypeMap::const_iterator it =
@@ -361,11 +360,10 @@ bool RTPPayloadRegistry::PayloadTypeToPayload(
 
   // Check that this is a registered payload type.
   if (it == payload_type_map_.end()) {
-    return false;
+    return nullptr;
   }
 
-  payload = it->second;
-  return true;
+  return it->second;
 }
 
 void RTPPayloadRegistry::SetIncomingPayloadType(const RTPHeader& header) {
@@ -390,7 +388,7 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy {
 
   bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                            const uint32_t frequency,
-                           const uint8_t channels,
+                           const size_t channels,
                            const uint32_t rate) const override {
     return
         payload.audio &&
@@ -409,7 +407,7 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy {
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payloadType,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) const override {
     RtpUtility::Payload* payload = new RtpUtility::Payload;
     payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
@@ -433,7 +431,7 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy {
 
   bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                            const uint32_t frequency,
-                           const uint8_t channels,
+                           const size_t channels,
                            const uint32_t rate) const override {
     return !payload.audio;
   }
@@ -447,7 +445,7 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy {
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payloadType,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) const override {
     RtpVideoCodecTypes videoType = kRtpVideoGeneric;
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index 0b9bf2751e..b73666d1af 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
@@ -25,7 +25,7 @@ using ::testing::Return;
 using ::testing::_;
 
 static const char* kTypicalPayloadName = "name";
-static const uint8_t kTypicalChannels = 1;
+static const size_t kTypicalChannels = 1;
 static const int kTypicalFrequency = 44000;
 static const int kTypicalRate = 32 * 1024;
 
@@ -52,10 +52,9 @@ class RtpPayloadRegistryTest : public ::testing::Test {
     RtpUtility::Payload* returned_payload_on_heap =
         new RtpUtility::Payload(returned_payload);
     EXPECT_CALL(*mock_payload_strategy_,
-        CreatePayloadType(kTypicalPayloadName, payload_type,
-            kTypicalFrequency,
-            kTypicalChannels,
-            rate)).WillOnce(Return(returned_payload_on_heap));
+                CreatePayloadType(kTypicalPayloadName, payload_type,
+                                  kTypicalFrequency, kTypicalChannels, rate))
+        .WillOnce(Return(returned_payload_on_heap));
     return returned_payload_on_heap;
   }
 
@@ -70,14 +69,14 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) {
 
   bool new_payload_created = false;
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &new_payload_created));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &new_payload_created));
 
   EXPECT_TRUE(new_payload_created) << "A new payload WAS created.";
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                          retrieved_payload));
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+  EXPECT_TRUE(retrieved_payload);
 
   // We should get back the exact pointer to the payload returned by the
   // payload strategy.
@@ -85,32 +84,30 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) {
 
   // Now forget about it and verify it's gone.
   EXPECT_EQ(0, rtp_payload_registry_->DeRegisterReceivePayload(payload_type));
-  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type, retrieved_payload));
+  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type));
 }
 
 TEST_F(RtpPayloadRegistryTest, AudioRedWorkProperly) {
   const uint8_t kRedPayloadType = 127;
   const int kRedSampleRate = 8000;
-  const int kRedChannels = 1;
+  const size_t kRedChannels = 1;
   const int kRedBitRate = 0;
 
   // This creates an audio RTP payload strategy.
-  rtp_payload_registry_.reset(new RTPPayloadRegistry(
-      RTPPayloadStrategy::CreateStrategy(true)));
+  rtp_payload_registry_.reset(
+      new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true)));
 
   bool new_payload_created = false;
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      "red", kRedPayloadType, kRedSampleRate, kRedChannels, kRedBitRate,
-      &new_payload_created));
+                   "red", kRedPayloadType, kRedSampleRate, kRedChannels,
+                   kRedBitRate, &new_payload_created));
   EXPECT_TRUE(new_payload_created);
 
   EXPECT_EQ(kRedPayloadType, rtp_payload_registry_->red_payload_type());
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(kRedPayloadType,
-                                                          retrieved_payload));
-  ASSERT_TRUE(retrieved_payload);
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(kRedPayloadType);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_TRUE(retrieved_payload->audio);
   EXPECT_STRCASEEQ("red", retrieved_payload->name);
 
@@ -127,27 +124,29 @@ TEST_F(RtpPayloadRegistryTest,
   RtpUtility::Payload* first_payload_on_heap =
       ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
 
   EXPECT_EQ(-1, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored)) << "Adding same codec twice = bad.";
+                    kTypicalPayloadName, payload_type, kTypicalFrequency,
+                    kTypicalChannels, kTypicalRate, &ignored))
+      << "Adding same codec twice = bad.";
 
   RtpUtility::Payload* second_payload_on_heap =
       ExpectReturnOfTypicalAudioPayload(payload_type - 1, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
-      kTypicalChannels, kTypicalRate, &ignored)) <<
-          "With a different payload type is fine though.";
+                   kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored))
+      << "With a different payload type is fine though.";
 
   // Ensure both payloads are preserved.
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                          retrieved_payload));
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_EQ(first_payload_on_heap, retrieved_payload);
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1,
-                                                          retrieved_payload));
+  retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_EQ(second_payload_on_heap, retrieved_payload);
 
   // Ok, update the rate for one of the codecs. If either the incoming rate or
@@ -158,8 +157,8 @@ TEST_F(RtpPayloadRegistryTest,
   EXPECT_CALL(*mock_payload_strategy_,
               UpdatePayloadRate(first_payload_on_heap, kTypicalRate));
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
 }
 
 TEST_F(RtpPayloadRegistryTest,
@@ -174,35 +173,31 @@ TEST_F(RtpPayloadRegistryTest,
   bool ignored = false;
   ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
   ExpectReturnOfTypicalAudioPayload(payload_type - 1, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
-      kTypicalChannels, kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type, retrieved_payload)) << "The first payload should be "
-          "deregistered because the only thing that differs is payload type.";
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type - 1, retrieved_payload)) <<
-          "The second payload should still be registered though.";
+  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type))
+      << "The first payload should be "
+         "deregistered because the only thing that differs is payload type.";
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1))
+      << "The second payload should still be registered though.";
 
   // Now ensure non-compatible codecs aren't removed.
   ON_CALL(*mock_payload_strategy_, PayloadIsCompatible(_, _, _, _))
       .WillByDefault(Return(false));
   ExpectReturnOfTypicalAudioPayload(payload_type + 1, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type + 1, kTypicalFrequency,
-      kTypicalChannels, kTypicalRate, &ignored));
-
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type - 1, retrieved_payload)) <<
-          "Not compatible; both payloads should be kept.";
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type + 1, retrieved_payload)) <<
-          "Not compatible; both payloads should be kept.";
+                   kTypicalPayloadName, payload_type + 1, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
+
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1))
+      << "Not compatible; both payloads should be kept.";
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type + 1))
+      << "Not compatible; both payloads should be kept.";
 }
 
 TEST_F(RtpPayloadRegistryTest,
@@ -218,18 +213,17 @@ TEST_F(RtpPayloadRegistryTest,
   bool ignored;
   ExpectReturnOfTypicalAudioPayload(34, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, 34, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, 34, kTypicalFrequency, kTypicalChannels,
+                   kTypicalRate, &ignored));
 
   EXPECT_EQ(-1, rtp_payload_registry_->last_received_payload_type());
   media_type_unchanged = rtp_payload_registry_->ReportMediaPayloadType(18);
   EXPECT_FALSE(media_type_unchanged);
 }
 
-class ParameterizedRtpPayloadRegistryTest :
-    public RtpPayloadRegistryTest,
-    public ::testing::WithParamInterface<int> {
-};
+class ParameterizedRtpPayloadRegistryTest
+    : public RtpPayloadRegistryTest,
+      public ::testing::WithParamInterface<int> {};
 
 TEST_P(ParameterizedRtpPayloadRegistryTest,
        FailsToRegisterKnownPayloadsWeAreNotInterestedIn) {
@@ -237,26 +231,26 @@ TEST_P(ParameterizedRtpPayloadRegistryTest,
 
   bool ignored;
   EXPECT_EQ(-1, rtp_payload_registry_->RegisterReceivePayload(
-      "whatever", static_cast<uint8_t>(payload_type), 19, 1, 17, &ignored));
+                    "whatever", static_cast<uint8_t>(payload_type), 19, 1, 17,
+                    &ignored));
 }
 
 INSTANTIATE_TEST_CASE_P(TestKnownBadPayloadTypes,
                         ParameterizedRtpPayloadRegistryTest,
                         testing::Values(64, 72, 73, 74, 75, 76, 77, 78, 79));
 
-class RtpPayloadRegistryGenericTest :
-    public RtpPayloadRegistryTest,
-    public ::testing::WithParamInterface<int> {
-};
+class RtpPayloadRegistryGenericTest
+    : public RtpPayloadRegistryTest,
+      public ::testing::WithParamInterface<int> {};
 
 TEST_P(RtpPayloadRegistryGenericTest, RegisterGenericReceivePayloadType) {
   int payload_type = GetParam();
 
   bool ignored;
 
-  EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload("generic-codec",
-    static_cast<int8_t>(payload_type),
-    19, 1, 17, &ignored)); // dummy values, except for payload_type
+  EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
+                   "generic-codec", static_cast<int8_t>(payload_type), 19, 1,
+                   17, &ignored));  // dummy values, except for payload_type
 }
 
 // Generates an RTX packet for the given length and original sequence number.
@@ -395,7 +389,8 @@ TEST_F(RtpPayloadRegistryTest, InvalidRtxConfiguration) {
   TestRtxPacket(rtp_payload_registry_.get(), 106, 0, false);
 }
 
-INSTANTIATE_TEST_CASE_P(TestDynamicRange, RtpPayloadRegistryGenericTest,
-                        testing::Range(96, 127+1));
+INSTANTIATE_TEST_CASE_P(TestDynamicRange,
+                        RtpPayloadRegistryGenericTest,
+                        testing::Range(96, 127 + 1));
 
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index 176852e01e..1dd07d1cc9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -14,8 +14,8 @@
 #include <set>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/typedefs.h"
@@ -42,9 +42,7 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
   // Is TelephoneEvent configured with payload type payload_type
   bool TelephoneEventPayloadType(const int8_t payload_type) const;
 
-  TelephoneEventHandler* GetTelephoneEventHandler() {
-    return this;
-  }
+  TelephoneEventHandler* GetTelephoneEventHandler() { return this; }
 
   // Returns true if CNG is configured with payload type payload_type. If so,
   // the frequency and cng_payload_type_has_changed are filled in.
@@ -96,13 +94,11 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
 
  private:
-
-  int32_t ParseAudioCodecSpecific(
-      WebRtcRTPHeader* rtp_header,
-      const uint8_t* payload_data,
-      size_t payload_length,
-      const AudioPayload& audio_specific,
-      bool is_red);
+  int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
+                                  const uint8_t* payload_data,
+                                  size_t payload_length,
+                                  const AudioPayload& audio_specific,
+                                  bool is_red);
 
   uint32_t last_received_frequency_;
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index e1ebd0c8bb..2e21f230d3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -16,13 +16,12 @@
 #include <string.h>
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 
 namespace webrtc {
 
-using RtpUtility::GetCurrentRTP;
 using RtpUtility::Payload;
 using RtpUtility::StringCompare;
 
@@ -97,7 +96,7 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payload_type,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
@@ -170,7 +169,7 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
   bool is_red = false;
 
-  if (CheckPayloadChanged(rtp_header, first_payload_byte, is_red,
+  if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red,
                           &payload_specific) == -1) {
     if (payload_length == 0) {
       // OK, keep-alive packet.
@@ -253,7 +252,7 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
   bool new_ssrc = false;
   bool re_initialize_decoder = false;
   char payload_name[RTP_PAYLOAD_NAME_SIZE];
-  uint8_t channels = 1;
+  size_t channels = 1;
   uint32_t rate = 0;
 
   {
@@ -276,12 +275,11 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
         if (rtp_header.payloadType == last_received_payload_type) {
           re_initialize_decoder = true;
 
-          Payload* payload;
-          if (!rtp_payload_registry_->PayloadTypeToPayload(
-              rtp_header.payloadType, payload)) {
+          const Payload* payload = rtp_payload_registry_->PayloadTypeToPayload(
+              rtp_header.payloadType);
+          if (!payload) {
             return;
           }
-          assert(payload);
           payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
           strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
           if (payload->audio) {
@@ -321,7 +319,7 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
 // last known payload).
 int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
                                              const int8_t first_payload_byte,
-                                             bool& is_red,
+                                             bool* is_red,
                                              PayloadUnion* specific_payload) {
   bool re_initialize_decoder = false;
 
@@ -339,7 +337,7 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
       if (rtp_payload_registry_->red_payload_type() == payload_type) {
         // Get the real codec payload type.
         payload_type = first_payload_byte & 0x7f;
-        is_red = true;
+        *is_red = true;
 
         if (rtp_payload_registry_->red_payload_type() == payload_type) {
           // Invalid payload type, traced by caller. If we proceeded here,
@@ -361,16 +359,16 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
         &should_discard_changes);
 
       if (should_discard_changes) {
-        is_red = false;
+        *is_red = false;
         return 0;
       }
 
-      Payload* payload;
-      if (!rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload)) {
+      const Payload* payload =
+          rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+      if (!payload) {
         // Not a registered payload type.
         return -1;
       }
-      assert(payload);
       payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
       strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
 
@@ -391,7 +389,7 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
       }
     } else {
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
-      is_red = false;
+      *is_red = false;
     }
   }  // End critsect.
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 46741d59b4..5cf94c2859 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -12,8 +12,8 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
@@ -36,7 +36,7 @@ class RtpReceiverImpl : public RtpReceiver {
   int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
                                  const int8_t payload_type,
                                  const uint32_t frequency,
-                                 const uint8_t channels,
+                                 const size_t channels,
                                  const uint32_t rate) override;
 
   int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
@@ -71,7 +71,7 @@ class RtpReceiverImpl : public RtpReceiver {
   void CheckCSRC(const WebRtcRTPHeader& rtp_header);
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
                               const int8_t first_payload_byte,
-                              bool& is_red,
+                              bool* is_red,
                               PayloadUnion* payload);
 
   Clock* clock_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index 37c3e6e49a..0f7ad30e87 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -12,8 +12,8 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
@@ -95,7 +95,7 @@ class RTPReceiverStrategy {
   // Note: Implementations may call the callback for other reasons than calls
   // to ParseRtpPacket, for instance if the implementation somehow recovers a
   // packet.
-  RTPReceiverStrategy(RtpData* data_callback);
+  explicit RTPReceiverStrategy(RtpData* data_callback);
 
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   PayloadUnion last_payload_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 1af2d4857e..53051dd321 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -16,8 +16,8 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 23128df6e1..56f761a2e1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
index 7cfebd91a8..a2cd52736f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
@@ -13,36 +13,37 @@
 
 // Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
 namespace webrtc {
-enum { NACK_BYTECOUNT_SIZE      = 60};   // size of our NACK history
+enum { NACK_BYTECOUNT_SIZE = 60 };  // size of our NACK history
 // A sanity for the NACK list parsing at the send-side.
 enum { kSendSideNackListSizeSanity = 20000 };
 enum { kDefaultMaxReorderingThreshold = 50 };  // In sequence numbers.
 enum { kRtcpMaxNackFields = 253 };
 
-enum { RTCP_INTERVAL_VIDEO_MS       = 1000 };
-enum { RTCP_INTERVAL_AUDIO_MS       = 5000 };
-enum { RTCP_SEND_BEFORE_KEY_FRAME_MS= 100 };
-enum { RTCP_MAX_REPORT_BLOCKS       = 31};      // RFC 3550 page 37
-enum { RTCP_MIN_FRAME_LENGTH_MS     = 17};
-enum { kRtcpAppCode_DATA_SIZE           = 32*4};    // multiple of 4, this is not a limitation of the size
-enum { RTCP_RPSI_DATA_SIZE          = 30};
-enum { RTCP_NUMBER_OF_SR            = 60 };
-
-enum { MAX_NUMBER_OF_TEMPORAL_ID    = 8 };          // RFC
-enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID  = 128 };// RFC
+enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
+enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
+enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 };
+enum { RTCP_MAX_REPORT_BLOCKS = 31 };  // RFC 3550 page 37
+enum { RTCP_MIN_FRAME_LENGTH_MS = 17 };
+enum {
+  kRtcpAppCode_DATA_SIZE = 32 * 4
+};  // multiple of 4, this is not a limitation of the size
+enum { RTCP_RPSI_DATA_SIZE = 30 };
+enum { RTCP_NUMBER_OF_SR = 60 };
+
+enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 };              // RFC
+enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 };  // RFC
 enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 };
 
-enum { BW_HISTORY_SIZE          = 35};
+enum { BW_HISTORY_SIZE = 35 };
 
-#define MIN_AUDIO_BW_MANAGEMENT_BITRATE   6
-#define MIN_VIDEO_BW_MANAGEMENT_BITRATE   30
+#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
+#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
 
-enum { DTMF_OUTBAND_MAX         = 20};
+enum { DTMF_OUTBAND_MAX = 20 };
 
 enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
 enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
-enum { RTP_MAX_PACKETS_PER_FRAME= 512 }; // must be multiple of 32
+enum { RTP_MAX_PACKETS_PER_FRAME = 512 };  // must be multiple of 32
 }  // namespace webrtc
 
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 451f8bfa93..450eed698e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -183,8 +183,13 @@ int32_t ModuleRtpRtcpImpl::Process() {
       set_rtt_ms(rtt_stats_->LastProcessedRtt());
   }
 
-  if (rtcp_sender_.TimeToSendRTCPReport())
-    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+  // For sending streams, make sure to not send a SR before media has been sent.
+  if (rtcp_sender_.TimeToSendRTCPReport()) {
+    RTCPSender::FeedbackState state = GetFeedbackState();
+    // Prevent sending streams to send SR before any media has been sent.
+    if (!rtcp_sender_.Sending() || state.packets_sent > 0)
+      rtcp_sender_.SendRTCP(state, kRtcpReport);
+  }
 
   if (UpdateRTCPReceiveInformationTimers()) {
     // A receiver has timed out
@@ -402,6 +407,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     const RTPFragmentationHeader* fragmentation,
     const RTPVideoHeader* rtp_video_hdr) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
+  // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
       rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
   }
@@ -794,9 +800,8 @@ int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType(
 }
 
 // Get payload type for Redundant Audio Data RFC 2198.
-int32_t ModuleRtpRtcpImpl::SendREDPayloadType(
-    int8_t& payload_type) const {
-  return rtp_sender_.RED(&payload_type);
+int32_t ModuleRtpRtcpImpl::SendREDPayloadType(int8_t* payload_type) const {
+  return rtp_sender_.RED(payload_type);
 }
 
 void ModuleRtpRtcpImpl::SetTargetSendBitrate(uint32_t bitrate_bps) {
@@ -832,11 +837,10 @@ void ModuleRtpRtcpImpl::SetGenericFECStatus(
   rtp_sender_.SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
-void ModuleRtpRtcpImpl::GenericFECStatus(bool& enable,
-                                         uint8_t& payload_type_red,
-                                         uint8_t& payload_type_fec) {
-  rtp_sender_.GenericFECStatus(&enable, &payload_type_red,
-                                      &payload_type_fec);
+void ModuleRtpRtcpImpl::GenericFECStatus(bool* enable,
+                                         uint8_t* payload_type_red,
+                                         uint8_t* payload_type_fec) {
+  rtp_sender_.GenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
 int32_t ModuleRtpRtcpImpl::SetFecParameters(
@@ -952,8 +956,8 @@ bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
 }
 
 // Called from RTCPsender.
-int32_t ModuleRtpRtcpImpl::BoundingSet(bool& tmmbr_owner,
-                                       TMMBRSet*& bounding_set) {
+int32_t ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner,
+                                       TMMBRSet* bounding_set) {
   return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set);
 }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index c9b6686c0a..04e09c1217 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -12,10 +12,12 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
 #include <list>
+#include <set>
+#include <utility>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
@@ -258,7 +260,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   int32_t SetSendREDPayloadType(int8_t payload_type) override;
 
   // Get payload type for Redundant Audio Data RFC 2198.
-  int32_t SendREDPayloadType(int8_t& payload_type) const override;
+  int32_t SendREDPayloadType(int8_t* payload_type) const override;
 
   // Store the audio level in d_bov for header-extension-for-audio-level-
   // indication.
@@ -280,9 +282,9 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
                            uint8_t payload_type_red,
                            uint8_t payload_type_fec) override;
 
-  void GenericFECStatus(bool& enable,
-                        uint8_t& payload_type_red,
-                        uint8_t& payload_type_fec) override;
+  void GenericFECStatus(bool* enable,
+                        uint8_t* payload_type_red,
+                        uint8_t* payload_type_fec) override;
 
   int32_t SetFecParameters(const FecProtectionParams* delta_params,
                            const FecProtectionParams* key_params) override;
@@ -293,7 +295,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
 
-  virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
+  int32_t BoundingSet(bool* tmmbr_owner, TMMBRSet* bounding_set_rec);
 
   void BitrateSent(uint32_t* total_rate,
                    uint32_t* video_rate,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 4c94764ee6..8329f603f9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -8,13 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <map>
+#include <set>
+
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/system_wrappers/include/scoped_vector.h"
 #include "webrtc/test/rtcp_packet_parser.h"
@@ -94,7 +98,7 @@ class SendTransport : public Transport,
 
 class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
  public:
-  RtpRtcpModule(SimulatedClock* clock)
+  explicit RtpRtcpModule(SimulatedClock* clock)
       : receive_statistics_(ReceiveStatistics::Create(clock)) {
     RtpRtcp::Configuration config;
     config.audio = false;
@@ -346,6 +350,27 @@ TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
   EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
 }
 
+TEST_F(RtpRtcpImplTest, NoSrBeforeMedia) {
+  // Ignore fake transport delays in this test.
+  sender_.transport_.SimulateNetworkDelay(0, &clock_);
+  receiver_.transport_.SimulateNetworkDelay(0, &clock_);
+
+  sender_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+
+  // Verify no SR is sent before media has been sent, RR should still be sent
+  // from the receiving module though.
+  clock_.AdvanceTimeMilliseconds(2000);
+  int64_t current_time = clock_.TimeInMilliseconds();
+  sender_.impl_->Process();
+  receiver_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+  EXPECT_EQ(receiver_.RtcpSent().first_packet_time_ms, current_time);
+
+  SendFrame(&sender_, kBaseLayerTid);
+  EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, current_time);
+}
+
 TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_Nack) {
   EXPECT_EQ(-1, receiver_.RtcpSent().first_packet_time_ms);
   EXPECT_EQ(-1, sender_.RtcpReceived().first_packet_time_ms);
@@ -522,5 +547,4 @@ TEST_F(RtpRtcpImplTest, UniqueNackRequests) {
   EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
   EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
 }
-
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 50f476829d..f4933afdd9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -11,15 +11,17 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <stdlib.h>  // srand
+#include <algorithm>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
@@ -33,6 +35,7 @@ static const uint32_t kAbsSendTimeFraction = 18;
 namespace {
 
 const size_t kRtpHeaderLength = 12;
+const uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
 
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
@@ -125,6 +128,7 @@ RTPSender::RTPSender(
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
+      random_(clock_->TimeInMicroseconds()),
       bitrates_(new BitrateAggregator(bitrate_callback)),
       total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
       audio_configured_(audio),
@@ -182,8 +186,8 @@ RTPSender::RTPSender(
   ssrc_rtx_ = ssrc_db_.CreateSSRC();  // Can't be 0.
   bitrates_->set_ssrc(ssrc_);
   // Random start, 16 bits. Can't be 0.
-  sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
-  sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
+  sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
+  sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
 }
 
 RTPSender::~RTPSender() {
@@ -292,7 +296,7 @@ int32_t RTPSender::RegisterPayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     int8_t payload_number,
     uint32_t frequency,
-    uint8_t channels,
+    size_t channels,
     uint32_t rate) {
   assert(payload_name);
   CriticalSectionScoped cs(send_critsect_.get());
@@ -323,11 +327,11 @@ int32_t RTPSender::RegisterPayload(
     return -1;
   }
   int32_t ret_val = 0;
-  RtpUtility::Payload* payload = NULL;
+  RtpUtility::Payload* payload = nullptr;
   if (audio_configured_) {
     // TODO(mflodman): Change to CreateAudioPayload and make static.
     ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
-                                           frequency, channels, rate, payload);
+                                           frequency, channels, rate, &payload);
   } else {
     payload = video_->CreateVideoPayload(payload_name, payload_number, rate);
   }
@@ -452,7 +456,7 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
   }
   if (audio_configured_) {
     int8_t red_pl_type = -1;
-    if (audio_->RED(red_pl_type) == 0) {
+    if (audio_->RED(&red_pl_type) == 0) {
       // We have configured RED.
       if (red_pl_type == payload_type) {
         // And it's a match...
@@ -469,7 +473,8 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_type);
   if (it == payload_type_map_.end()) {
-    LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
+    LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
+                    << " not registered.";
     return -1;
   }
   SetSendPayloadType(payload_type);
@@ -512,7 +517,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
-    LOG(LS_ERROR) << "Don't send data with unknown payload type.";
+    LOG(LS_ERROR) << "Don't send data with unknown payload type: "
+                  << static_cast<int>(payload_type) << ".";
     return -1;
   }
 
@@ -573,7 +579,7 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
       break;
     RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
     bytes_left -= static_cast<int>(length - rtp_header.headerLength);
   }
   return bytes_to_send - bytes_left;
@@ -583,8 +589,7 @@ void RTPSender::BuildPaddingPacket(uint8_t* packet,
                                    size_t header_length,
                                    size_t padding_length) {
   packet[0] |= 0x20;  // Set padding bit.
-  int32_t *data =
-      reinterpret_cast<int32_t *>(&(packet[header_length]));
+  int32_t* data = reinterpret_cast<int32_t*>(&(packet[header_length]));
 
   // Fill data buffer with random data.
   for (size_t j = 0; j < (padding_length >> 2); ++j) {
@@ -665,7 +670,7 @@ size_t RTPSender::SendPadData(size_t bytes,
 
     RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
 
     if (capture_time_ms > 0) {
       UpdateTransmissionTimeOffset(
@@ -717,7 +722,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
   if (paced_sender_) {
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
-    if (!rtp_parser.Parse(header)) {
+    if (!rtp_parser.Parse(&header)) {
       assert(false);
       return -1;
     }
@@ -725,7 +730,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
     paced_sender_->InsertPacket(
-        RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
+        RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
         corrected_capture_tims_ms, length - header.headerLength, true);
 
     return length;
@@ -903,11 +908,11 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
                                      int64_t capture_time_ms,
                                      bool send_over_rtx,
                                      bool is_retransmit) {
-  uint8_t *buffer_to_send_ptr = buffer;
+  uint8_t* buffer_to_send_ptr = buffer;
 
   RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   if (!is_retransmit && rtp_header.markerBit) {
     TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
                            capture_time_ms);
@@ -996,14 +1001,14 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
   bool fec_enabled;
   uint8_t pt_red;
   uint8_t pt_fec;
-  video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
+  video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
-  if (bytes == 0)
+  if (audio_configured_ || bytes == 0)
     return 0;
   {
     CriticalSectionScoped cs(send_critsect_.get());
@@ -1026,7 +1031,7 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
   RtpUtility::RtpHeaderParser rtp_parser(buffer,
                                          payload_length + rtp_header_length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   int64_t now_ms = clock_->TimeInMilliseconds();
 
@@ -1169,7 +1174,7 @@ size_t RTPSender::CreateRtpHeader(uint8_t* header,
   int32_t rtp_header_length = kRtpHeaderLength;
 
   if (csrcs.size() > 0) {
-    uint8_t *ptr = &header[rtp_header_length];
+    uint8_t* ptr = &header[rtp_header_length];
     for (size_t i = 0; i < csrcs.size(); ++i) {
       ByteWriter<uint32_t>::WriteBigEndian(ptr, csrcs[i]);
       ptr += 4;
@@ -1638,7 +1643,7 @@ uint16_t RTPSender::UpdateTransportSequenceNumber(
 void RTPSender::SetSendingStatus(bool enabled) {
   if (enabled) {
     uint32_t frequency_hz = SendPayloadFrequency();
-    uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
+    uint32_t RTPtime = CurrentRtp(*clock_, frequency_hz);
 
     // Will be ignored if it's already configured via API.
     SetStartTimestamp(RTPtime, false);
@@ -1653,8 +1658,7 @@ void RTPSender::SetSendingStatus(bool enabled) {
     // Don't initialize seq number if SSRC passed externally.
     if (!sequence_number_forced_ && !ssrc_forced_) {
       // Generate a new sequence number.
-      sequence_number_ =
-          rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);  // NOLINT
+      sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
     }
   }
 }
@@ -1716,8 +1720,7 @@ void RTPSender::SetSSRC(uint32_t ssrc) {
   ssrc_ = ssrc;
   bitrates_->set_ssrc(ssrc_);
   if (!sequence_number_forced_) {
-    sequence_number_ =
-        rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);  // NOLINT
+    sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
 }
 
@@ -1775,7 +1778,7 @@ int32_t RTPSender::RED(int8_t *payload_type) const {
   if (!audio_configured_) {
     return -1;
   }
-  return audio_->RED(*payload_type);
+  return audio_->RED(payload_type);
 }
 
 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
@@ -1801,7 +1804,7 @@ void RTPSender::GenericFECStatus(bool* enable,
                                     uint8_t* payload_type_red,
                                     uint8_t* payload_type_fec) const {
   RTC_DCHECK(!audio_configured_);
-  video_->GenericFECStatus(*enable, *payload_type_red, *payload_type_fec);
+  video_->GenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
 int32_t RTPSender::SetFecParameters(
@@ -1823,7 +1826,7 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
       reinterpret_cast<const uint8_t*>(buffer), *length);
 
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   // Add original RTP header.
   memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
@@ -1836,7 +1839,7 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
   }
 
   // Replace sequence number.
-  uint8_t *ptr = data_buffer_rtx + 2;
+  uint8_t* ptr = data_buffer_rtx + 2;
   ByteWriter<uint16_t>::WriteBigEndian(ptr, sequence_number_rtx_++);
 
   // Replace SSRC.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index a134370c76..3c62336507 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -10,14 +10,16 @@
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
-#include <assert.h>
-#include <math.h>
 
+#include <list>
 #include <map>
+#include <utility>
+#include <vector>
 
+#include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
@@ -26,8 +28,6 @@
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 #include "webrtc/transport.h"
 
-#define MAX_INIT_RTP_SEQ_NUMBER 32767  // 2^15 -1.
-
 namespace webrtc {
 
 class BitrateAggregator;
@@ -116,7 +116,7 @@ class RTPSender : public RTPSenderInterface {
   int32_t RegisterPayload(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payload_type, const uint32_t frequency,
-      const uint8_t channels, const uint32_t rate);
+      const size_t channels, const uint32_t rate);
 
   int32_t DeRegisterSendPayload(const int8_t payload_type);
 
@@ -163,7 +163,7 @@ class RTPSender : public RTPSenderInterface {
   int32_t SetTransportSequenceNumber(uint16_t sequence_number);
 
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
-  virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
+  bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   size_t RtpHeaderExtensionTotalLength() const;
@@ -202,10 +202,10 @@ class RTPSender : public RTPSenderInterface {
                         bool is_voiced,
                         uint8_t dBov) const;
 
-  virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
-                                   size_t rtp_packet_length,
-                                   const RTPHeader& rtp_header,
-                                   VideoRotation rotation) const override;
+  bool UpdateVideoRotation(uint8_t* rtp_packet,
+                           size_t rtp_packet_length,
+                           const RTPHeader& rtp_header,
+                           VideoRotation rotation) const override;
 
   bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
                         bool retransmission);
@@ -386,6 +386,7 @@ class RTPSender : public RTPSenderInterface {
 
   Clock* clock_;
   int64_t clock_delta_ms_;
+  Random random_ GUARDED_BY(send_critsect_);
 
   rtc::scoped_ptr<BitrateAggregator> bitrates_;
   Bitrate total_bitrate_sent_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 1fc9a89ce1..2aa4961cdc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -10,12 +10,12 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
-#include <assert.h> //assert
-#include <string.h> //memcpy
+#include <string.h>
 
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -47,8 +47,7 @@ RTPSenderAudio::RTPSenderAudio(Clock* clock,
       _lastPayloadType(-1),
       _audioLevel_dBov(0) {}
 
-RTPSenderAudio::~RTPSenderAudio() {
-}
+RTPSenderAudio::~RTPSenderAudio() {}
 
 int RTPSenderAudio::AudioFrequency() const {
   return kDtmfFrequencyHz;
@@ -56,22 +55,20 @@ int RTPSenderAudio::AudioFrequency() const {
 
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
-int32_t
-RTPSenderAudio::SetAudioPacketSize(const uint16_t packetSizeSamples)
-{
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packetSizeSamples) {
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
 
-    _packetSizeSamples = packetSizeSamples;
-    return 0;
+  _packetSizeSamples = packetSizeSamples;
+  return 0;
 }
 
 int32_t RTPSenderAudio::RegisterAudioPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payloadType,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate,
-    RtpUtility::Payload*& payload) {
+    RtpUtility::Payload** payload) {
   if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     //  we can have multiple CNG payload types
@@ -99,72 +96,65 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
     return 0;
     // The default timestamp rate is 8000 Hz, but other rates may be defined.
   }
-  payload = new RtpUtility::Payload;
-  payload->typeSpecific.Audio.frequency = frequency;
-  payload->typeSpecific.Audio.channels = channels;
-  payload->typeSpecific.Audio.rate = rate;
-  payload->audio = true;
-  payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
-  strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+  *payload = new RtpUtility::Payload;
+  (*payload)->typeSpecific.Audio.frequency = frequency;
+  (*payload)->typeSpecific.Audio.channels = channels;
+  (*payload)->typeSpecific.Audio.rate = rate;
+  (*payload)->audio = true;
+  (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
+  strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
   return 0;
 }
 
-bool
-RTPSenderAudio::MarkerBit(const FrameType frameType,
-                          const int8_t payload_type)
-{
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
-    // for audio true for first packet in a speech burst
-    bool markerBit = false;
-    if (_lastPayloadType != payload_type) {
-      if (payload_type != -1 && (_cngNBPayloadType == payload_type ||
-                                 _cngWBPayloadType == payload_type ||
-                                 _cngSWBPayloadType == payload_type ||
-                                 _cngFBPayloadType == payload_type)) {
-        // Only set a marker bit when we change payload type to a non CNG
-        return false;
-      }
+bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  // for audio true for first packet in a speech burst
+  bool markerBit = false;
+  if (_lastPayloadType != payload_type) {
+    if (payload_type != -1 && (_cngNBPayloadType == payload_type ||
+                               _cngWBPayloadType == payload_type ||
+                               _cngSWBPayloadType == payload_type ||
+                               _cngFBPayloadType == payload_type)) {
+      // Only set a marker bit when we change payload type to a non CNG
+      return false;
+    }
 
-      // payload_type differ
-      if (_lastPayloadType == -1) {
-        if (frameType != kAudioFrameCN) {
-          // first packet and NOT CNG
-          return true;
-        } else {
-          // first packet and CNG
-          _inbandVADactive = true;
-          return false;
-        }
+    // payload_type differ
+    if (_lastPayloadType == -1) {
+      if (frameType != kAudioFrameCN) {
+        // first packet and NOT CNG
+        return true;
+      } else {
+        // first packet and CNG
+        _inbandVADactive = true;
+        return false;
       }
-
-      // not first packet AND
-      // not CNG AND
-      // payload_type changed
-
-      // set a marker bit when we change payload type
-      markerBit = true;
     }
 
-    // For G.723 G.729, AMR etc we can have inband VAD
-    if(frameType == kAudioFrameCN)
-    {
-        _inbandVADactive = true;
+    // not first packet AND
+    // not CNG AND
+    // payload_type changed
 
-    } else if(_inbandVADactive)
-    {
-        _inbandVADactive = false;
-        markerBit = true;
-    }
-    return markerBit;
+    // set a marker bit when we change payload type
+    markerBit = true;
+  }
+
+  // For G.723 G.729, AMR etc we can have inband VAD
+  if (frameType == kAudioFrameCN) {
+    _inbandVADactive = true;
+  } else if (_inbandVADactive) {
+    _inbandVADactive = false;
+    markerBit = true;
+  }
+  return markerBit;
 }
 
-int32_t RTPSenderAudio::SendAudio(
-    const FrameType frameType,
-    const int8_t payloadType,
-    const uint32_t captureTimeStamp,
-    const uint8_t* payloadData,
-    const size_t dataSize,
-    const RTPFragmentationHeader* fragmentation) {
+int32_t RTPSenderAudio::SendAudio(FrameType frameType,
+                                  int8_t payloadType,
+                                  uint32_t captureTimeStamp,
+                                  const uint8_t* payloadData,
+                                  size_t dataSize,
+                                  const RTPFragmentationHeader* fragmentation) {
   // TODO(pwestin) Breakup function in smaller functions.
   size_t payloadSize = dataSize;
   size_t maxPayloadLength = _rtpSender->MaxPayloadLength();
@@ -185,8 +175,8 @@ int32_t RTPSenderAudio::SendAudio(
 
   // Check if we have pending DTMFs to send
   if (!_dtmfEventIsOn && PendingDTMF()) {
-    int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() -
-        _dtmfTimeLastSent;
+    int64_t delaySinceLastDTMF =
+        _clock->TimeInMilliseconds() - _dtmfTimeLastSent;
 
     if (delaySinceLastDTMF > 100) {
       // New tone to play
@@ -294,128 +284,120 @@ int32_t RTPSenderAudio::SendAudio(
     // Too large payload buffer.
     return -1;
   }
-    if (red_payload_type >= 0 &&  // Have we configured RED?
-        fragmentation && fragmentation->fragmentationVectorSize > 1 &&
-        !markerBit) {
-      if (timestampOffset <= 0x3fff) {
-        if (fragmentation->fragmentationVectorSize != 2) {
-          // we only support 2 codecs when using RED
-          return -1;
-        }
-        // only 0x80 if we have multiple blocks
-        dataBuffer[rtpHeaderLength++] =
-            0x80 + fragmentation->fragmentationPlType[1];
-        size_t blockLength = fragmentation->fragmentationLength[1];
-
-        // sanity blockLength
-        if (blockLength > 0x3ff) {  // block length 10 bits 1023 bytes
-          return -1;
-        }
-        uint32_t REDheader = (timestampOffset << 10) + blockLength;
-        ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + rtpHeaderLength,
-                                             REDheader);
-        rtpHeaderLength += 3;
-
-        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-        // copy the RED data
-        memcpy(dataBuffer + rtpHeaderLength,
-               payloadData + fragmentation->fragmentationOffset[1],
-               fragmentation->fragmentationLength[1]);
-
-        // copy the normal data
-        memcpy(dataBuffer + rtpHeaderLength +
-                   fragmentation->fragmentationLength[1],
-               payloadData + fragmentation->fragmentationOffset[0],
-               fragmentation->fragmentationLength[0]);
-
-        payloadSize = fragmentation->fragmentationLength[0] +
-                      fragmentation->fragmentationLength[1];
-      } else {
-        // silence for too long send only new data
-        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-        memcpy(dataBuffer + rtpHeaderLength,
-               payloadData + fragmentation->fragmentationOffset[0],
-               fragmentation->fragmentationLength[0]);
+  if (red_payload_type >= 0 &&  // Have we configured RED?
+      fragmentation && fragmentation->fragmentationVectorSize > 1 &&
+      !markerBit) {
+    if (timestampOffset <= 0x3fff) {
+      if (fragmentation->fragmentationVectorSize != 2) {
+        // we only support 2 codecs when using RED
+        return -1;
+      }
+      // only 0x80 if we have multiple blocks
+      dataBuffer[rtpHeaderLength++] =
+          0x80 + fragmentation->fragmentationPlType[1];
+      size_t blockLength = fragmentation->fragmentationLength[1];
 
-        payloadSize = fragmentation->fragmentationLength[0];
+      // sanity blockLength
+      if (blockLength > 0x3ff) {  // block length 10 bits 1023 bytes
+        return -1;
       }
+      uint32_t REDheader = (timestampOffset << 10) + blockLength;
+      ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + rtpHeaderLength,
+                                           REDheader);
+      rtpHeaderLength += 3;
+
+      dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
+      // copy the RED data
+      memcpy(dataBuffer + rtpHeaderLength,
+             payloadData + fragmentation->fragmentationOffset[1],
+             fragmentation->fragmentationLength[1]);
+
+      // copy the normal data
+      memcpy(
+          dataBuffer + rtpHeaderLength + fragmentation->fragmentationLength[1],
+          payloadData + fragmentation->fragmentationOffset[0],
+          fragmentation->fragmentationLength[0]);
+
+      payloadSize = fragmentation->fragmentationLength[0] +
+                    fragmentation->fragmentationLength[1];
     } else {
-      if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
-        // use the fragment info if we have one
-        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-        memcpy(dataBuffer + rtpHeaderLength,
-               payloadData + fragmentation->fragmentationOffset[0],
-               fragmentation->fragmentationLength[0]);
-
-        payloadSize = fragmentation->fragmentationLength[0];
-      } else {
-        memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
-      }
+      // silence for too long send only new data
+      dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
+      memcpy(dataBuffer + rtpHeaderLength,
+             payloadData + fragmentation->fragmentationOffset[0],
+             fragmentation->fragmentationLength[0]);
+
+      payloadSize = fragmentation->fragmentationLength[0];
     }
-    {
-      CriticalSectionScoped cs(_sendAudioCritsect.get());
-      _lastPayloadType = payloadType;
+  } else {
+    if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
+      // use the fragment info if we have one
+      dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
+      memcpy(dataBuffer + rtpHeaderLength,
+             payloadData + fragmentation->fragmentationOffset[0],
+             fragmentation->fragmentationLength[0]);
+
+      payloadSize = fragmentation->fragmentationLength[0];
+    } else {
+      memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
     }
-    // Update audio level extension, if included.
-    size_t packetSize = payloadSize + rtpHeaderLength;
-    RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
-    RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
-    _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
-                                 (frameType == kAudioFrameSpeech),
-                                 audio_level_dbov);
-    TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
-                           _rtpSender->Timestamp(), "seqnum",
-                           _rtpSender->SequenceNumber());
-    return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
-                                     -1, kAllowRetransmission,
-                                     RtpPacketSender::kHighPriority);
   }
-
-    // Audio level magnitude and voice activity flag are set for each RTP packet
-int32_t
-RTPSenderAudio::SetAudioLevel(const uint8_t level_dBov)
-{
-    if (level_dBov > 127)
-    {
-        return -1;
-    }
+  {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
-    _audioLevel_dBov = level_dBov;
-    return 0;
+    _lastPayloadType = payloadType;
+  }
+  // Update audio level extension, if included.
+  size_t packetSize = payloadSize + rtpHeaderLength;
+  RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
+  RTPHeader rtp_header;
+  rtp_parser.Parse(&rtp_header);
+  _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
+                               (frameType == kAudioFrameSpeech),
+                               audio_level_dbov);
+  TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
+                         _rtpSender->Timestamp(), "seqnum",
+                         _rtpSender->SequenceNumber());
+  return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
+                                   TickTime::MillisecondTimestamp(),
+                                   kAllowRetransmission,
+                                   RtpPacketSender::kHighPriority);
 }
 
-    // Set payload type for Redundant Audio Data RFC 2198
-int32_t
-RTPSenderAudio::SetRED(const int8_t payloadType)
-{
-    if(payloadType < -1 )
-    {
-        return -1;
-    }
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
-    _REDPayloadType = payloadType;
-    return 0;
+// Audio level magnitude and voice activity flag are set for each RTP packet
+int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) {
+  if (level_dBov > 127) {
+    return -1;
+  }
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  _audioLevel_dBov = level_dBov;
+  return 0;
 }
 
-    // Get payload type for Redundant Audio Data RFC 2198
-int32_t
-RTPSenderAudio::RED(int8_t& payloadType) const
-{
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
-    if(_REDPayloadType == -1)
-    {
-        // not configured
-        return -1;
-    }
-    payloadType = _REDPayloadType;
-    return 0;
+// Set payload type for Redundant Audio Data RFC 2198
+int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
+  if (payloadType < -1) {
+    return -1;
+  }
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  _REDPayloadType = payloadType;
+  return 0;
+}
+
+// Get payload type for Redundant Audio Data RFC 2198
+int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  if (_REDPayloadType == -1) {
+    // not configured
+    return -1;
+  }
+  *payloadType = _REDPayloadType;
+  return 0;
 }
 
 // Send a TelephoneEvent tone using RFC 2833 (4733)
-int32_t RTPSenderAudio::SendTelephoneEvent(const uint8_t key,
-                                           const uint16_t time_ms,
-                                           const uint8_t level) {
+int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
+                                           uint16_t time_ms,
+                                           uint8_t level) {
   {
     CriticalSectionScoped lock(_sendAudioCritsect.get());
     if (_dtmfPayloadType < 0) {
@@ -426,63 +408,57 @@ int32_t RTPSenderAudio::SendTelephoneEvent(const uint8_t key,
   return AddDTMF(key, time_ms, level);
 }
 
-int32_t
-RTPSenderAudio::SendTelephoneEventPacket(bool ended,
-                                         int8_t dtmf_payload_type,
-                                         uint32_t dtmfTimeStamp,
-                                         uint16_t duration,
-                                         bool markerBit)
-{
-    uint8_t dtmfbuffer[IP_PACKET_SIZE];
-    uint8_t sendCount = 1;
-    int32_t retVal = 0;
-
-    if(ended)
-    {
-        // resend last packet in an event 3 times
-        sendCount = 3;
-    }
-    do
-    {
-        //Send DTMF data
-        _rtpSender->BuildRTPheader(dtmfbuffer, dtmf_payload_type, markerBit,
-                                   dtmfTimeStamp, _clock->TimeInMilliseconds());
-
-        // reset CSRC and X bit
-        dtmfbuffer[0] &= 0xe0;
-
-        //Create DTMF data
-        /*    From RFC 2833:
-
-         0                   1                   2                   3
-         0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-        |     event     |E|R| volume    |          duration             |
-        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-        */
-        // R bit always cleared
-        uint8_t R = 0x00;
-        uint8_t volume = _dtmfLevel;
-
-        // First packet un-ended
-        uint8_t E = ended ? 0x80 : 0x00;
-
-        // First byte is Event number, equals key number
-        dtmfbuffer[12] = _dtmfKey;
-        dtmfbuffer[13] = E|R|volume;
-        ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
-
-        TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                             "Audio::SendTelephoneEvent", "timestamp",
-                             dtmfTimeStamp, "seqnum",
-                             _rtpSender->SequenceNumber());
-        retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
-                                           kAllowRetransmission,
-                                           RtpPacketSender::kHighPriority);
-        sendCount--;
-
-    }while (sendCount > 0 && retVal == 0);
-
-    return retVal;
+int32_t RTPSenderAudio::SendTelephoneEventPacket(bool ended,
+                                                 int8_t dtmf_payload_type,
+                                                 uint32_t dtmfTimeStamp,
+                                                 uint16_t duration,
+                                                 bool markerBit) {
+  uint8_t dtmfbuffer[IP_PACKET_SIZE];
+  uint8_t sendCount = 1;
+  int32_t retVal = 0;
+
+  if (ended) {
+    // resend last packet in an event 3 times
+    sendCount = 3;
+  }
+  do {
+    // Send DTMF data
+    _rtpSender->BuildRTPheader(dtmfbuffer, dtmf_payload_type, markerBit,
+                               dtmfTimeStamp, _clock->TimeInMilliseconds());
+
+    // reset CSRC and X bit
+    dtmfbuffer[0] &= 0xe0;
+
+    // Create DTMF data
+    /*    From RFC 2833:
+
+     0                   1                   2                   3
+     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |     event     |E|R| volume    |          duration             |
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    */
+    // R bit always cleared
+    uint8_t R = 0x00;
+    uint8_t volume = _dtmfLevel;
+
+    // First packet un-ended
+    uint8_t E = ended ? 0x80 : 0x00;
+
+    // First byte is Event number, equals key number
+    dtmfbuffer[12] = _dtmfKey;
+    dtmfbuffer[13] = E | R | volume;
+    ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
+
+    TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                         "Audio::SendTelephoneEvent", "timestamp",
+                         dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
+    retVal = _rtpSender->SendToNetwork(
+        dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
+        kAllowRetransmission, RtpPacketSender::kHighPriority);
+    sendCount--;
+  } while (sendCount > 0 && retVal == 0);
+
+  return retVal;
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index dd16fe51b4..1e96d17a67 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -19,92 +19,91 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-class RTPSenderAudio: public DTMFqueue
-{
-public:
- RTPSenderAudio(Clock* clock,
-                RTPSender* rtpSender,
-                RtpAudioFeedback* audio_feedback);
-    virtual ~RTPSenderAudio();
-
-    int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                                 const int8_t payloadType,
-                                 const uint32_t frequency,
-                                 const uint8_t channels,
-                                 const uint32_t rate,
-                                 RtpUtility::Payload*& payload);
-
-    int32_t SendAudio(const FrameType frameType,
-                      const int8_t payloadType,
-                      const uint32_t captureTimeStamp,
-                      const uint8_t* payloadData,
-                      const size_t payloadSize,
-                      const RTPFragmentationHeader* fragmentation);
-
-    // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
-    int32_t SetAudioPacketSize(const uint16_t packetSizeSamples);
-
-    // Store the audio level in dBov for header-extension-for-audio-level-indication.
-    // Valid range is [0,100]. Actual value is negative.
-    int32_t SetAudioLevel(const uint8_t level_dBov);
-
-    // Send a DTMF tone using RFC 2833 (4733)
-    int32_t SendTelephoneEvent(const uint8_t key,
-                               const uint16_t time_ms,
-                               const uint8_t level);
-
-    int AudioFrequency() const;
-
-    // Set payload type for Redundant Audio Data RFC 2198
-    int32_t SetRED(const int8_t payloadType);
-
-    // Get payload type for Redundant Audio Data RFC 2198
-    int32_t RED(int8_t& payloadType) const;
-
-protected:
-    int32_t SendTelephoneEventPacket(bool ended,
-                                     int8_t dtmf_payload_type,
-                                     uint32_t dtmfTimeStamp,
-                                     uint16_t duration,
-                                     bool markerBit); // set on first packet in talk burst
-
-    bool MarkerBit(const FrameType frameType,
-                   const int8_t payloadType);
-
-private:
- Clock* const _clock;
- RTPSender* const _rtpSender;
- RtpAudioFeedback* const _audioFeedback;
-
- rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
-
- uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
-
- // DTMF
- bool _dtmfEventIsOn;
- bool _dtmfEventFirstPacketSent;
- int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
- uint32_t _dtmfTimestamp;
- uint8_t _dtmfKey;
- uint32_t _dtmfLengthSamples;
- uint8_t _dtmfLevel;
- int64_t _dtmfTimeLastSent;
- uint32_t _dtmfTimestampLastSent;
-
- int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
-
- // VAD detection, used for markerbit
- bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
- int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
-
- // Audio level indication
- // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
- uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
+class RTPSenderAudio : public DTMFqueue {
+ public:
+  RTPSenderAudio(Clock* clock,
+                 RTPSender* rtpSender,
+                 RtpAudioFeedback* audio_feedback);
+  virtual ~RTPSenderAudio();
+
+  int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+                               int8_t payloadType,
+                               uint32_t frequency,
+                               size_t channels,
+                               uint32_t rate,
+                               RtpUtility::Payload** payload);
+
+  int32_t SendAudio(FrameType frameType,
+                    int8_t payloadType,
+                    uint32_t captureTimeStamp,
+                    const uint8_t* payloadData,
+                    size_t payloadSize,
+                    const RTPFragmentationHeader* fragmentation);
+
+  // set audio packet size, used to determine when it's time to send a DTMF
+  // packet in silence (CNG)
+  int32_t SetAudioPacketSize(uint16_t packetSizeSamples);
+
+  // Store the audio level in dBov for
+  // header-extension-for-audio-level-indication.
+  // Valid range is [0,100]. Actual value is negative.
+  int32_t SetAudioLevel(uint8_t level_dBov);
+
+  // Send a DTMF tone using RFC 2833 (4733)
+  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
+
+  int AudioFrequency() const;
+
+  // Set payload type for Redundant Audio Data RFC 2198
+  int32_t SetRED(int8_t payloadType);
+
+  // Get payload type for Redundant Audio Data RFC 2198
+  int32_t RED(int8_t* payloadType) const;
+
+ protected:
+  int32_t SendTelephoneEventPacket(
+      bool ended,
+      int8_t dtmf_payload_type,
+      uint32_t dtmfTimeStamp,
+      uint16_t duration,
+      bool markerBit);  // set on first packet in talk burst
+
+  bool MarkerBit(const FrameType frameType, const int8_t payloadType);
+
+ private:
+  Clock* const _clock;
+  RTPSender* const _rtpSender;
+  RtpAudioFeedback* const _audioFeedback;
+
+  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+
+  uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
+
+  // DTMF
+  bool _dtmfEventIsOn;
+  bool _dtmfEventFirstPacketSent;
+  int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
+  uint32_t _dtmfTimestamp;
+  uint8_t _dtmfKey;
+  uint32_t _dtmfLengthSamples;
+  uint8_t _dtmfLevel;
+  int64_t _dtmfTimeLastSent;
+  uint32_t _dtmfTimestampLastSent;
+
+  int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
+
+  // VAD detection, used for markerbit
+  bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
+
+  // Audio level indication
+  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
+  uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index fde6d47ceb..6bc122201a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -12,20 +12,22 @@
  * This file includes unit tests for the RTPSender.
  */
 
+#include <list>
+#include <vector>
+
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
-#include "webrtc/system_wrappers/include/stl_util.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/include/stl_util.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/typedefs.h"
 
@@ -90,9 +92,7 @@ class LoopbackTransportTest : public webrtc::Transport {
     sent_packets_.push_back(buffer);
     return true;
   }
-  bool SendRtcp(const uint8_t* data, size_t len) override {
-    return false;
-  }
+  bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
   int packets_sent_;
   size_t last_sent_packet_len_;
   size_t total_bytes_sent_;
@@ -163,11 +163,8 @@ class RtpSenderTest : public ::testing::Test {
 
   void SendPacket(int64_t capture_time_ms, int payload_length) {
     uint32_t timestamp = capture_time_ms * 90;
-    int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
-                                                     kPayload,
-                                                     kMarkerBit,
-                                                     timestamp,
-                                                     capture_time_ms);
+    int32_t rtp_length = rtp_sender_->BuildRTPheader(
+        packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
     ASSERT_GE(rtp_length, 0);
 
     // Packet should be stored in a send bucket.
@@ -186,7 +183,7 @@ class RtpSenderTestWithoutPacer : public RtpSenderTest {
 
 class RtpSenderVideoTest : public RtpSenderTest {
  protected:
-  virtual void SetUp() override {
+  void SetUp() override {
     // TODO(pbos): Set up to use pacer.
     SetUpRtpSender(false);
     rtp_sender_video_.reset(
@@ -211,7 +208,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
     } else {
       ASSERT_EQ(kRtpHeaderSize, length);
     }
-    ASSERT_TRUE(rtp_parser.Parse(rtp_header, map));
+    ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map));
     ASSERT_FALSE(rtp_parser.RTCP());
     EXPECT_EQ(payload_, rtp_header.payloadType);
     EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -228,53 +225,57 @@ TEST_F(RtpSenderTestWithoutPacer,
        RegisterRtpTransmissionTimeOffsetHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength,
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset));
+                   kRtpExtensionTransmissionTimeOffset));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAbsoluteSendTimeHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kAbsoluteSendTimeLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime));
+                   kRtpExtensionAbsoluteSendTime));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAudioLevelHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength),
       rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel));
+  EXPECT_EQ(0,
+            rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kTransmissionTimeOffsetLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kTransmissionTimeOffsetLength +
                                     kAbsoluteSendTimeLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(
                 kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength +
                 kAbsoluteSendTimeLength + kAudioLevelLength),
@@ -290,18 +291,18 @@ TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
 
   // Deregister starts.
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset));
+                   kRtpExtensionTransmissionTimeOffset));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kAbsoluteSendTimeLength +
                                     kAudioLevelLength + kVideoRotationLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime));
+                   kRtpExtensionAbsoluteSendTime));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kAudioLevelLength + kVideoRotationLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel));
+  EXPECT_EQ(0,
+            rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
       rtp_sender_->RtpHeaderExtensionTotalLength());
@@ -334,7 +335,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, nullptr);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, nullptr);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -354,7 +355,8 @@ TEST_F(RtpSenderTestWithoutPacer,
        BuildRTPPacketWithTransmissionOffsetExtension) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -368,7 +370,7 @@ TEST_F(RtpSenderTestWithoutPacer,
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -379,7 +381,7 @@ TEST_F(RtpSenderTestWithoutPacer,
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -393,7 +395,8 @@ TEST_F(RtpSenderTestWithoutPacer,
   const int kNegTimeOffset = -500;
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -407,7 +410,7 @@ TEST_F(RtpSenderTestWithoutPacer,
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -419,8 +422,9 @@ TEST_F(RtpSenderTestWithoutPacer,
 
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -433,7 +437,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -444,7 +448,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -472,7 +476,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
@@ -500,7 +504,7 @@ TEST_F(RtpSenderTestWithoutPacer,
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header, false);
   EXPECT_EQ(length, rtp_header.headerLength);
@@ -508,8 +512,8 @@ TEST_F(RtpSenderTestWithoutPacer,
 }
 
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -521,12 +525,12 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -538,7 +542,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -554,11 +558,13 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   EXPECT_EQ(0,
             rtp_sender_->SetTransportSequenceNumber(kTransportSequenceNumber));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
@@ -573,7 +579,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
 
   RtpHeaderExtensionMap map;
@@ -583,7 +589,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
   map.Register(kRtpExtensionTransportSequenceNumber,
                kTransportSequenceNumberExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -602,7 +608,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -626,9 +632,11 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
@@ -659,7 +667,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
   ASSERT_TRUE(valid_rtp_header);
 
   // Verify transmission time offset.
@@ -676,9 +684,11 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
@@ -717,7 +727,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
   ASSERT_TRUE(valid_rtp_header);
 
   // Verify transmission time offset.
@@ -740,10 +750,12 @@ TEST_F(RtpSenderTest, SendPadding) {
   rtp_sender_->SetStorePacketsStatus(true, 10);
   size_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
@@ -815,8 +827,8 @@ TEST_F(RtpSenderTest, SendPadding) {
 
   // Send a regular video packet again.
   capture_time_ms = fake_clock_.TimeInMilliseconds();
-  rtp_length_int = rtp_sender_->BuildRTPheader(
-      packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
+  rtp_length_int = rtp_sender_->BuildRTPheader(packet_, kPayload, kMarkerBit,
+                                               timestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   rtp_length = static_cast<size_t>(rtp_length_int);
 
@@ -830,8 +842,8 @@ TEST_F(RtpSenderTest, SendPadding) {
   EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
   EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
   // Parse sent packet.
-  ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, rtp_length,
-                                &rtp_header));
+  ASSERT_TRUE(
+      rtp_parser->Parse(transport_.last_sent_packet_, rtp_length, &rtp_header));
 
   // Verify sequence number and timestamp.
   EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -858,8 +870,9 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
   uint16_t seq_num = kSeqNum;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   int32_t rtp_header_len = kRtpHeaderSize;
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
@@ -876,8 +889,8 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
                                          kAbsoluteSendTimeExtensionId);
   rtp_sender_->SetTargetBitrate(300000);
   const size_t kNumPayloadSizes = 10;
-  const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750,
-      800, 850, 900, 950};
+  const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
+                                                  750, 800, 850, 900, 950};
   // Send 10 packets of increasing size.
   for (size_t i = 0; i < kNumPayloadSizes; ++i) {
     int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
@@ -921,10 +934,10 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
-  const uint8_t* payload_data = GetPayloadData(rtp_header,
-      transport_.last_sent_packet_);
+  const uint8_t* payload_data =
+      GetPayloadData(rtp_header, transport_.last_sent_packet_);
   uint8_t generic_header = *payload_data++;
 
   ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
@@ -946,7 +959,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
 
   RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
                                           transport_.last_sent_packet_len_);
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_);
   generic_header = *payload_data++;
@@ -1043,9 +1056,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
-  ASSERT_EQ(
-      0,
-      rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 1500));
+  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
+                                            0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
@@ -1057,13 +1069,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
   // Send a few frames.
   for (uint32_t i = 0; i < kNumPackets; ++i) {
     ASSERT_EQ(0,
-              rtp_sender_->SendOutgoingData(kVideoFrameKey,
-                                            payload_type,
-                                            1234,
-                                            4321,
-                                            payload,
-                                            sizeof(payload),
-                                            0));
+              rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                            4321, payload, sizeof(payload), 0));
     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
   }
 
@@ -1100,8 +1107,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
 TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
-    TestCallback()
-      : StreamDataCountersCallback(), ssrc_(0), counters_() {}
+    TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
     virtual ~TestCallback() {}
 
     void DataCountersUpdated(const StreamDataCounters& counters,
@@ -1127,7 +1133,6 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
       MatchPacketCounter(counters.retransmitted, counters_.retransmitted);
       EXPECT_EQ(counters.fec.packets, counters_.fec.packets);
     }
-
   } callback;
 
   const uint8_t kRedPayloadType = 96;
@@ -1212,10 +1217,10 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
-  const uint8_t* payload_data = GetPayloadData(rtp_header,
-      transport_.last_sent_packet_);
+  const uint8_t* payload_data =
+      GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
@@ -1225,8 +1230,8 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
 
 TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
@@ -1241,21 +1246,22 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
-  const uint8_t* payload_data = GetPayloadData(rtp_header,
-                                               transport_.last_sent_packet_);
+  const uint8_t* payload_data =
+      GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
 
   EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
 
-  uint8_t extension[] = { 0xbe, 0xde, 0x00, 0x01,
-                          (kAudioLevelExtensionId << 4) + 0, // ID + length.
-                          kAudioLevel,                       // Data.
-                          0x00, 0x00                         // Padding.
-                        };
+  uint8_t extension[] = {
+      0xbe, 0xde, 0x00, 0x01,
+      (kAudioLevelExtensionId << 4) + 0,  // ID + length.
+      kAudioLevel,                        // Data.
+      0x00, 0x00                          // Padding.
+  };
 
   EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension),
                       sizeof(extension)));
@@ -1270,14 +1276,14 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
 TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event";
   uint8_t payload_type = 126;
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 0,
-                                            0, 0));
+  ASSERT_EQ(0,
+            rtp_sender_->RegisterPayload(payload_name, payload_type, 0, 0, 0));
   // For Telephone events, payload is not added to the registered payload list,
   // it will register only the payload used for audio stream.
   // Registering the payload again for audio stream with different payload name.
-  strcpy(payload_name, "payload_name");
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000,
-                                            1, 0));
+  const char kPayloadName[] = "payload_name";
+  ASSERT_EQ(
+      0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);
@@ -1298,8 +1304,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
-                                transport_.last_sent_packet_len_,
-                                &rtp_header));
+                                transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 1 for first packet.
   EXPECT_TRUE(rtp_header.markerBit);
 
@@ -1307,8 +1312,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
                                              capture_time_ms + 4000, 0, nullptr,
                                              0, nullptr));
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
-                                transport_.last_sent_packet_len_,
-                                &rtp_header));
+                                transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 0 for rest of the packets.
   EXPECT_FALSE(rtp_header.markerBit);
 }
@@ -1321,19 +1325,13 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
 
-  ASSERT_EQ(
-      0,
-      rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000, 0, 1500));
+  ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
+                                            0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(0,
-            rtp_sender_->SendOutgoingData(kVideoFrameKey,
-                                          kPayloadType,
-                                          1234,
-                                          4321,
-                                          payload,
-                                          sizeof(payload),
-                                          0));
+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321,
+                                       payload, sizeof(payload), 0));
 
   // Will send 2 full-size padding packets.
   rtp_sender_->TimeToSendPadding(1);
@@ -1353,17 +1351,17 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   EXPECT_EQ(rtx_stats.transmitted.padding_bytes, 2 * kMaxPaddingSize);
 
   EXPECT_EQ(rtp_stats.transmitted.TotalBytes(),
-      rtp_stats.transmitted.payload_bytes +
-      rtp_stats.transmitted.header_bytes +
-      rtp_stats.transmitted.padding_bytes);
+            rtp_stats.transmitted.payload_bytes +
+                rtp_stats.transmitted.header_bytes +
+                rtp_stats.transmitted.padding_bytes);
   EXPECT_EQ(rtx_stats.transmitted.TotalBytes(),
-      rtx_stats.transmitted.payload_bytes +
-      rtx_stats.transmitted.header_bytes +
-      rtx_stats.transmitted.padding_bytes);
+            rtx_stats.transmitted.payload_bytes +
+                rtx_stats.transmitted.header_bytes +
+                rtx_stats.transmitted.padding_bytes);
 
-  EXPECT_EQ(transport_.total_bytes_sent_,
-            rtp_stats.transmitted.TotalBytes() +
-            rtx_stats.transmitted.TotalBytes());
+  EXPECT_EQ(
+      transport_.total_bytes_sent_,
+      rtp_stats.transmitted.TotalBytes() + rtx_stats.transmitted.TotalBytes());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 66062771de..5a565dfa99 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -18,7 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
@@ -42,13 +42,13 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
       _retransmissionSettings(kRetransmitBaseLayer),
 
       // Generic FEC
-      _fec(),
-      _fecEnabled(false),
-      _payloadTypeRED(-1),
-      _payloadTypeFEC(-1),
+      fec_(),
+      fec_enabled_(false),
+      red_payload_type_(-1),
+      fec_payload_type_(-1),
       delta_fec_params_(),
       key_fec_params_(),
-      producer_fec_(&_fec),
+      producer_fec_(&fec_),
       _fecOverheadRate(clock, NULL),
       _videoBitrate(clock, NULL) {
   memset(&delta_fec_params_, 0, sizeof(delta_fec_params_));
@@ -104,7 +104,7 @@ void RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
                                      StorageType storage) {
   if (_rtpSender.SendToNetwork(data_buffer, payload_length, rtp_header_length,
                                capture_time_ms, storage,
-                               RtpPacketSender::kNormalPriority) == 0) {
+                               RtpPacketSender::kLowPriority) == 0) {
     _videoBitrate.Update(payload_length + rtp_header_length);
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                          "Video::PacketNormal", "timestamp", capture_timestamp,
@@ -130,7 +130,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
     // Only protect while creating RED and FEC packets, not when sending.
     CriticalSectionScoped cs(crit_.get());
     red_packet.reset(producer_fec_.BuildRedPacket(
-        data_buffer, payload_length, rtp_header_length, _payloadTypeRED));
+        data_buffer, payload_length, rtp_header_length, red_payload_type_));
     if (protect) {
       producer_fec_.AddRtpPacketAndGenerateFec(data_buffer, payload_length,
                                                rtp_header_length);
@@ -140,7 +140,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
       next_fec_sequence_number =
           _rtpSender.AllocateSequenceNumber(num_fec_packets);
       fec_packets = producer_fec_.GetFecPackets(
-          _payloadTypeRED, _payloadTypeFEC, next_fec_sequence_number,
+          red_payload_type_, fec_payload_type_, next_fec_sequence_number,
           rtp_header_length);
       RTC_DCHECK_EQ(num_fec_packets, fec_packets.size());
       if (_retransmissionSettings & kRetransmitFECPackets)
@@ -150,7 +150,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
   if (_rtpSender.SendToNetwork(
           red_packet->data(), red_packet->length() - rtp_header_length,
           rtp_header_length, capture_time_ms, media_packet_storage,
-          RtpPacketSender::kNormalPriority) == 0) {
+          RtpPacketSender::kLowPriority) == 0) {
     _videoBitrate.Update(red_packet->length());
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                          "Video::PacketRed", "timestamp", capture_timestamp,
@@ -162,7 +162,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
     if (_rtpSender.SendToNetwork(
             fec_packet->data(), fec_packet->length() - rtp_header_length,
             rtp_header_length, capture_time_ms, fec_storage,
-            RtpPacketSender::kNormalPriority) == 0) {
+            RtpPacketSender::kLowPriority) == 0) {
       _fecOverheadRate.Update(fec_packet->length());
       TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                            "Video::PacketFec", "timestamp", capture_timestamp,
@@ -180,9 +180,9 @@ void RTPSenderVideo::SetGenericFECStatus(const bool enable,
                                          const uint8_t payloadTypeRED,
                                          const uint8_t payloadTypeFEC) {
   CriticalSectionScoped cs(crit_.get());
-  _fecEnabled = enable;
-  _payloadTypeRED = payloadTypeRED;
-  _payloadTypeFEC = payloadTypeFEC;
+  fec_enabled_ = enable;
+  red_payload_type_ = payloadTypeRED;
+  fec_payload_type_ = payloadTypeFEC;
   memset(&delta_fec_params_, 0, sizeof(delta_fec_params_));
   memset(&key_fec_params_, 0, sizeof(key_fec_params_));
   delta_fec_params_.max_fec_frames = key_fec_params_.max_fec_frames = 1;
@@ -190,18 +190,18 @@ void RTPSenderVideo::SetGenericFECStatus(const bool enable,
       kFecMaskRandom;
 }
 
-void RTPSenderVideo::GenericFECStatus(bool& enable,
-                                      uint8_t& payloadTypeRED,
-                                      uint8_t& payloadTypeFEC) const {
+void RTPSenderVideo::GenericFECStatus(bool* enable,
+                                      uint8_t* payloadTypeRED,
+                                      uint8_t* payloadTypeFEC) const {
   CriticalSectionScoped cs(crit_.get());
-  enable = _fecEnabled;
-  payloadTypeRED = _payloadTypeRED;
-  payloadTypeFEC = _payloadTypeFEC;
+  *enable = fec_enabled_;
+  *payloadTypeRED = red_payload_type_;
+  *payloadTypeFEC = fec_payload_type_;
 }
 
 size_t RTPSenderVideo::FECPacketOverhead() const {
   CriticalSectionScoped cs(crit_.get());
-  if (_fecEnabled) {
+  if (fec_enabled_) {
     // Overhead is FEC headers plus RED for FEC header plus anything in RTP
     // header beyond the 12 bytes base header (CSRC list, extensions...)
     // This reason for the header extensions to be included here is that
@@ -247,7 +247,7 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
         frameType == kVideoFrameKey ? &key_fec_params_ : &delta_fec_params_;
     producer_fec_.SetFecParameters(fec_params, 0);
     storage = packetizer->GetStorageType(_retransmissionSettings);
-    fec_enabled = _fecEnabled;
+    fec_enabled = fec_enabled_;
   }
 
   // Register CVO rtp header extension at the first time when we receive a frame
@@ -304,7 +304,7 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
       size_t packetSize = payloadSize + rtp_header_length;
       RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
       RTPHeader rtp_header;
-      rtp_parser.Parse(rtp_header);
+      rtp_parser.Parse(&rtp_header);
       _rtpSender.UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
                                      rtpHdr->rotation);
     }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index f412542d86..e59321ab93 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -16,7 +16,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
@@ -67,9 +67,9 @@ class RTPSenderVideo {
                            const uint8_t payloadTypeRED,
                            const uint8_t payloadTypeFEC);
 
-  void GenericFECStatus(bool& enable,
-                        uint8_t& payloadTypeRED,
-                        uint8_t& payloadTypeFEC) const;
+  void GenericFECStatus(bool* enable,
+                        uint8_t* payloadTypeRED,
+                        uint8_t* payloadTypeFEC) const;
 
   void SetFecParameters(const FecProtectionParams* delta_params,
                         const FecProtectionParams* key_params);
@@ -82,7 +82,7 @@ class RTPSenderVideo {
   int SelectiveRetransmissions() const;
   void SetSelectiveRetransmissions(uint8_t settings);
 
-private:
+ private:
   void SendVideoPacket(uint8_t* dataBuffer,
                        const size_t payloadLength,
                        const size_t rtpHeaderLength,
@@ -110,10 +110,10 @@ private:
   int32_t _retransmissionSettings GUARDED_BY(crit_);
 
   // FEC
-  ForwardErrorCorrection _fec;
-  bool _fecEnabled GUARDED_BY(crit_);
-  int8_t _payloadTypeRED GUARDED_BY(crit_);
-  int8_t _payloadTypeFEC GUARDED_BY(crit_);
+  ForwardErrorCorrection fec_;
+  bool fec_enabled_ GUARDED_BY(crit_);
+  int8_t red_payload_type_ GUARDED_BY(crit_);
+  int8_t fec_payload_type_ GUARDED_BY(crit_);
   FecProtectionParams delta_fec_params_ GUARDED_BY(crit_);
   FecProtectionParams key_fec_params_ GUARDED_BY(crit_);
   ProducerFec producer_fec_ GUARDED_BY(crit_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index bf0b30a064..0f0ad835b1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -10,38 +10,10 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
-#include <assert.h>
-#include <math.h>  // ceil
-#include <string.h>  // memcpy
-
-#if defined(_WIN32)
-// Order for these headers are important
-#include <winsock2.h>  // timeval
-#include <windows.h>  // FILETIME
-#include <MMSystem.h>  // timeGetTime
-#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
-#include <sys/time.h>  // gettimeofday
-#include <time.h>
-#endif
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#include <stdio.h>
-#endif
+#include <string.h>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#define DEBUG_PRINT(...)           \
-  {                                \
-    char msg[256];                 \
-    sprintf(msg, __VA_ARGS__);     \
-    OutputDebugString(msg);        \
-  }
-#else
-// special fix for visual 2003
-#define DEBUG_PRINT(exp)        ((void)0)
-#endif  // defined(_DEBUG) && defined(_WIN32)
 
 namespace webrtc {
 
@@ -77,50 +49,18 @@ enum {
 };
 
 /*
- * Time routines.
- */
-
-uint32_t GetCurrentRTP(Clock* clock, uint32_t freq) {
-  const bool use_global_clock = (clock == NULL);
-  Clock* local_clock = clock;
-  if (use_global_clock) {
-    local_clock = Clock::GetRealTimeClock();
-  }
-  uint32_t secs = 0, frac = 0;
-  local_clock->CurrentNtp(secs, frac);
-  if (use_global_clock) {
-    delete local_clock;
-  }
-  return ConvertNTPTimeToRTP(secs, frac, freq);
-}
-
-uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec, uint32_t NTPfrac, uint32_t freq) {
-  float ftemp = (float)NTPfrac / (float)NTP_FRAC;
-  uint32_t tmp = (uint32_t)(ftemp * freq);
-  return NTPsec * freq + tmp;
-}
-
-uint32_t ConvertNTPTimeToMS(uint32_t NTPsec, uint32_t NTPfrac) {
-  int freq = 1000;
-  float ftemp = (float)NTPfrac / (float)NTP_FRAC;
-  uint32_t tmp = (uint32_t)(ftemp * freq);
-  uint32_t MStime = NTPsec * freq + tmp;
-  return MStime;
-}
-
-/*
  * Misc utility routines
  */
 
 #if defined(_WIN32)
 bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
-  return (_strnicmp(str1, str2, length) == 0) ? true : false;
+  return _strnicmp(str1, str2, length) == 0;
 }
 #elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
 bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
-  return (strncasecmp(str1, str2, length) == 0) ? true : false;
+  return strncasecmp(str1, str2, length) == 0;
 }
 #endif
 
@@ -131,10 +71,6 @@ size_t Word32Align(size_t size) {
   return size;
 }
 
-uint32_t pow2(uint8_t exp) {
-  return 1 << exp;
-}
-
 RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData,
                                  const size_t rtpDataLength)
     : _ptrRTPDataBegin(rtpData),
@@ -244,7 +180,7 @@ bool RtpHeaderParser::ParseRtcp(RTPHeader* header) const {
   return true;
 }
 
-bool RtpHeaderParser::Parse(RTPHeader& header,
+bool RtpHeaderParser::Parse(RTPHeader* header,
                             RtpHeaderExtensionMap* ptrExtensionMap) const {
   const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
   if (length < kRtpMinParseLength) {
@@ -283,39 +219,39 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
     return false;
   }
 
-  header.markerBit      = M;
-  header.payloadType    = PT;
-  header.sequenceNumber = sequenceNumber;
-  header.timestamp      = RTPTimestamp;
-  header.ssrc           = SSRC;
-  header.numCSRCs       = CC;
-  header.paddingLength  = P ? *(_ptrRTPDataEnd - 1) : 0;
+  header->markerBit      = M;
+  header->payloadType    = PT;
+  header->sequenceNumber = sequenceNumber;
+  header->timestamp      = RTPTimestamp;
+  header->ssrc           = SSRC;
+  header->numCSRCs       = CC;
+  header->paddingLength  = P ? *(_ptrRTPDataEnd - 1) : 0;
 
   for (uint8_t i = 0; i < CC; ++i) {
     uint32_t CSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
     ptr += 4;
-    header.arrOfCSRCs[i] = CSRC;
+    header->arrOfCSRCs[i] = CSRC;
   }
 
-  header.headerLength   = 12 + CSRCocts;
+  header->headerLength   = 12 + CSRCocts;
 
   // If in effect, MAY be omitted for those packets for which the offset
   // is zero.
-  header.extension.hasTransmissionTimeOffset = false;
-  header.extension.transmissionTimeOffset = 0;
+  header->extension.hasTransmissionTimeOffset = false;
+  header->extension.transmissionTimeOffset = 0;
 
   // May not be present in packet.
-  header.extension.hasAbsoluteSendTime = false;
-  header.extension.absoluteSendTime = 0;
+  header->extension.hasAbsoluteSendTime = false;
+  header->extension.absoluteSendTime = 0;
 
   // May not be present in packet.
-  header.extension.hasAudioLevel = false;
-  header.extension.voiceActivity = false;
-  header.extension.audioLevel = 0;
+  header->extension.hasAudioLevel = false;
+  header->extension.voiceActivity = false;
+  header->extension.audioLevel = 0;
 
   // May not be present in packet.
-  header.extension.hasVideoRotation = false;
-  header.extension.videoRotation = 0;
+  header->extension.hasVideoRotation = false;
+  header->extension.videoRotation = 0;
 
   if (X) {
     /* RTP header extension, RFC 3550.
@@ -332,7 +268,7 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
       return false;
     }
 
-    header.headerLength += 4;
+    header->headerLength += 4;
 
     uint16_t definedByProfile = ByteReader<uint16_t>::ReadBigEndian(ptr);
     ptr += 2;
@@ -352,15 +288,16 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
                                   ptrRTPDataExtensionEnd,
                                   ptr);
     }
-    header.headerLength += XLen;
+    header->headerLength += XLen;
   }
-  if (header.headerLength + header.paddingLength > static_cast<size_t>(length))
+  if (header->headerLength + header->paddingLength >
+      static_cast<size_t>(length))
     return false;
   return true;
 }
 
 void RtpHeaderParser::ParseOneByteExtensionHeader(
-    RTPHeader& header,
+    RTPHeader* header,
     const RtpHeaderExtensionMap* ptrExtensionMap,
     const uint8_t* ptrRTPDataExtensionEnd,
     const uint8_t* ptr) const {
@@ -377,8 +314,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
 
     // Note that 'len' is the header extension element length, which is the
     // number of bytes - 1.
-    const uint8_t id = (*ptr & 0xf0) >> 4;
-    const uint8_t len = (*ptr & 0x0f);
+    const int id = (*ptr & 0xf0) >> 4;
+    const int len = (*ptr & 0x0f);
     ptr++;
 
     if (id == 15) {
@@ -390,8 +327,7 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
     RTPExtensionType type;
     if (ptrExtensionMap->GetType(id, &type) != 0) {
       // If we encounter an unknown extension, just skip over it.
-      LOG(LS_WARNING) << "Failed to find extension id: "
-                      << static_cast<int>(id);
+      LOG(LS_WARNING) << "Failed to find extension id: " << id;
     } else {
       switch (type) {
         case kRtpExtensionTransmissionTimeOffset: {
@@ -406,9 +342,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=2 |              transmission offset              |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-          header.extension.transmissionTimeOffset =
+          header->extension.transmissionTimeOffset =
               ByteReader<int32_t, 3>::ReadBigEndian(ptr);
-          header.extension.hasTransmissionTimeOffset = true;
+          header->extension.hasTransmissionTimeOffset = true;
           break;
         }
         case kRtpExtensionAudioLevel: {
@@ -422,9 +358,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=0 |V|   level     |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           //
-          header.extension.audioLevel = ptr[0] & 0x7f;
-          header.extension.voiceActivity = (ptr[0] & 0x80) != 0;
-          header.extension.hasAudioLevel = true;
+          header->extension.audioLevel = ptr[0] & 0x7f;
+          header->extension.voiceActivity = (ptr[0] & 0x80) != 0;
+          header->extension.hasAudioLevel = true;
           break;
         }
         case kRtpExtensionAbsoluteSendTime: {
@@ -438,9 +374,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=2 |              absolute send time               |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-          header.extension.absoluteSendTime =
+          header->extension.absoluteSendTime =
               ByteReader<uint32_t, 3>::ReadBigEndian(ptr);
-          header.extension.hasAbsoluteSendTime = true;
+          header->extension.hasAbsoluteSendTime = true;
           break;
         }
         case kRtpExtensionVideoRotation: {
@@ -454,14 +390,14 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           // |  ID   | len=0 |0 0 0 0 C F R R|
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-          header.extension.hasVideoRotation = true;
-          header.extension.videoRotation = ptr[0];
+          header->extension.hasVideoRotation = true;
+          header->extension.videoRotation = ptr[0];
           break;
         }
         case kRtpExtensionTransportSequenceNumber: {
           if (len != 1) {
-            LOG(LS_WARNING)
-                << "Incorrect peer connection sequence number len: " << len;
+            LOG(LS_WARNING) << "Incorrect transport sequence number len: "
+                            << len;
             return;
           }
           //   0                   1                   2
@@ -472,8 +408,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
 
           uint16_t sequence_number = ptr[0] << 8;
           sequence_number += ptr[1];
-          header.extension.transportSequenceNumber = sequence_number;
-          header.extension.hasTransportSequenceNumber = true;
+          header->extension.transportSequenceNumber = sequence_number;
+          header->extension.hasTransportSequenceNumber = true;
           break;
         }
         default: {
@@ -502,5 +438,4 @@ uint8_t RtpHeaderParser::ParsePaddingBytes(
   return num_zero_bytes;
 }
 }  // namespace RtpUtility
-
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index af20f97e82..23c175356a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -11,10 +11,11 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 
-#include <stddef.h> // size_t, ptrdiff_t
+#include <map>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/base/deprecation.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
@@ -29,71 +30,48 @@ RtpAudioFeedback* NullObjectRtpAudioFeedback();
 ReceiveStatistics* NullObjectReceiveStatistics();
 
 namespace RtpUtility {
-    // January 1970, in NTP seconds.
-    const uint32_t NTP_JAN_1970 = 2208988800UL;
-
-    // Magic NTP fractional unit.
-    const double NTP_FRAC = 4.294967296E+9;
-
-    struct Payload
-    {
-        char name[RTP_PAYLOAD_NAME_SIZE];
-        bool audio;
-        PayloadUnion typeSpecific;
-    };
-
-    typedef std::map<int8_t, Payload*> PayloadTypeMap;
-
-    // Return the current RTP timestamp from the NTP timestamp
-    // returned by the specified clock.
-    uint32_t GetCurrentRTP(Clock* clock, uint32_t freq);
-
-    // Return the current RTP absolute timestamp.
-    uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec,
-                                 uint32_t NTPfrac,
-                                 uint32_t freq);
-
-    uint32_t pow2(uint8_t exp);
-
-    // Returns true if |newTimestamp| is older than |existingTimestamp|.
-    // |wrapped| will be set to true if there has been a wraparound between the
-    // two timestamps.
-    bool OldTimestamp(uint32_t newTimestamp,
-                      uint32_t existingTimestamp,
-                      bool* wrapped);
-
-    bool StringCompare(const char* str1,
-                       const char* str2,
-                       const uint32_t length);
-
-    // Round up to the nearest size that is a multiple of 4.
-    size_t Word32Align(size_t size);
-
-    class RtpHeaderParser {
-    public:
-     RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
-     ~RtpHeaderParser();
-
-        bool RTCP() const;
-        bool ParseRtcp(RTPHeader* header) const;
-        bool Parse(RTPHeader& parsedPacket,
-                   RtpHeaderExtensionMap* ptrExtensionMap = NULL) const;
-
-    private:
-        void ParseOneByteExtensionHeader(
-            RTPHeader& parsedPacket,
-            const RtpHeaderExtensionMap* ptrExtensionMap,
-            const uint8_t* ptrRTPDataExtensionEnd,
-            const uint8_t* ptr) const;
-
-        uint8_t ParsePaddingBytes(
-            const uint8_t* ptrRTPDataExtensionEnd,
-            const uint8_t* ptr) const;
-
-        const uint8_t* const _ptrRTPDataBegin;
-        const uint8_t* const _ptrRTPDataEnd;
-    };
+
+struct Payload {
+  char name[RTP_PAYLOAD_NAME_SIZE];
+  bool audio;
+  PayloadUnion typeSpecific;
+};
+
+typedef std::map<int8_t, Payload*> PayloadTypeMap;
+
+bool StringCompare(const char* str1, const char* str2, const uint32_t length);
+
+// Round up to the nearest size that is a multiple of 4.
+size_t Word32Align(size_t size);
+
+class RtpHeaderParser {
+ public:
+  RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
+  ~RtpHeaderParser();
+
+  bool RTCP() const;
+  bool ParseRtcp(RTPHeader* header) const;
+  bool Parse(RTPHeader* parsedPacket,
+             RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
+  RTC_DEPRECATED bool Parse(
+      RTPHeader& parsedPacket,  // NOLINT(runtime/references)
+      RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const {
+    return Parse(&parsedPacket, ptrExtensionMap);
+  }
+
+ private:
+  void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
+                                   const RtpHeaderExtensionMap* ptrExtensionMap,
+                                   const uint8_t* ptrRTPDataExtensionEnd,
+                                   const uint8_t* ptr) const;
+
+  uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd,
+                            const uint8_t* ptr) const;
+
+  const uint8_t* const _ptrRTPDataBegin;
+  const uint8_t* const _ptrRTPDataEnd;
+};
 }  // namespace RtpUtility
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
diff --git a/webrtc/modules/rtp_rtcp/source/ssrc_database.cc b/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
index 6fb7c4701a..fb02b7ef12 100644
--- a/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
+++ b/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
@@ -10,110 +10,51 @@
 
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 
-#include <assert.h>
-#include <stdlib.h>
-
+#include "webrtc/base/checks.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
-#ifdef _WIN32
-    #include <windows.h>
-    #include <MMSystem.h> //timeGetTime
-
-// TODO(hellner): investigate if it is necessary to disable these warnings.
-    #pragma warning(disable:4311)
-    #pragma warning(disable:4312)
-#else
-    #include <stdio.h>
-    #include <string.h>
-    #include <time.h>
-    #include <sys/time.h>
-#endif
-
 namespace webrtc {
-SSRCDatabase*
-SSRCDatabase::StaticInstance(CountOperation count_operation)
-{
-  SSRCDatabase* impl =
-      GetStaticInstance<SSRCDatabase>(count_operation);
-  return impl;
+namespace {
+uint64_t Seed() {
+  return Clock::GetRealTimeClock()->TimeInMicroseconds();
 }
+}  // namespace
 
-SSRCDatabase*
-SSRCDatabase::GetSSRCDatabase()
-{
-    return StaticInstance(kAddRef);
+SSRCDatabase* SSRCDatabase::GetSSRCDatabase() {
+  return GetStaticInstance<SSRCDatabase>(kAddRef);
 }
 
-void
-SSRCDatabase::ReturnSSRCDatabase()
-{
-    StaticInstance(kRelease);
+void SSRCDatabase::ReturnSSRCDatabase() {
+  GetStaticInstance<SSRCDatabase>(kRelease);
 }
 
-uint32_t
-SSRCDatabase::CreateSSRC()
-{
-    CriticalSectionScoped lock(_critSect);
+uint32_t SSRCDatabase::CreateSSRC() {
+  CriticalSectionScoped lock(crit_.get());
 
-    uint32_t ssrc = GenerateRandom();
-
-    while(_ssrcMap.find(ssrc) != _ssrcMap.end())
-    {
-        ssrc = GenerateRandom();
+  while (true) {  // Try until get a new ssrc.
+    // 0 and 0xffffffff are invalid values for SSRC.
+    uint32_t ssrc = random_.Rand(1u, 0xfffffffe);
+    if (ssrcs_.insert(ssrc).second) {
+      return ssrc;
     }
-    _ssrcMap[ssrc] = 0;
-
-    return ssrc;
+  }
 }
 
-int32_t
-SSRCDatabase::RegisterSSRC(const uint32_t ssrc)
-{
-    CriticalSectionScoped lock(_critSect);
-    _ssrcMap[ssrc] = 0;
-    return 0;
+void SSRCDatabase::RegisterSSRC(uint32_t ssrc) {
+  CriticalSectionScoped lock(crit_.get());
+  ssrcs_.insert(ssrc);
 }
 
-int32_t
-SSRCDatabase::ReturnSSRC(const uint32_t ssrc)
-{
-    CriticalSectionScoped lock(_critSect);
-    _ssrcMap.erase(ssrc);
-    return 0;
+void SSRCDatabase::ReturnSSRC(uint32_t ssrc) {
+  CriticalSectionScoped lock(crit_.get());
+  ssrcs_.erase(ssrc);
 }
 
 SSRCDatabase::SSRCDatabase()
-{
-    // we need to seed the random generator, otherwise we get 26500 each time, hardly a random value :)
-#ifdef _WIN32
-    srand(timeGetTime());
-#else
-    struct timeval tv;
-    struct timezone tz;
-    gettimeofday(&tv, &tz);
-    srand(tv.tv_usec);
-#endif
+    : crit_(CriticalSectionWrapper::CreateCriticalSection()), random_(Seed()) {}
 
-    _critSect = CriticalSectionWrapper::CreateCriticalSection();
+SSRCDatabase::~SSRCDatabase() {
 }
 
-SSRCDatabase::~SSRCDatabase()
-{
-    _ssrcMap.clear();
-    delete _critSect;
-}
-
-uint32_t SSRCDatabase::GenerateRandom()
-{
-    uint32_t ssrc = 0;
-    do
-    {
-        ssrc = rand();
-        ssrc = ssrc <<16;
-        ssrc += rand();
-
-    } while (ssrc == 0 || ssrc == 0xffffffff);
-
-    return ssrc;
-}
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/ssrc_database.h b/webrtc/modules/rtp_rtcp/source/ssrc_database.h
index 7129d0de76..7a3133638d 100644
--- a/webrtc/modules/rtp_rtcp/source/ssrc_database.h
+++ b/webrtc/modules/rtp_rtcp/source/ssrc_database.h
@@ -11,43 +11,41 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
 
-#include <map>
+#include <set>
 
+#include "webrtc/base/random.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/static_instance.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
 
-class SSRCDatabase
-{
-public:
-    static SSRCDatabase* GetSSRCDatabase();
-    static void ReturnSSRCDatabase();
+class SSRCDatabase {
+ public:
+  static SSRCDatabase* GetSSRCDatabase();
+  static void ReturnSSRCDatabase();
 
-    uint32_t CreateSSRC();
-    int32_t RegisterSSRC(const uint32_t ssrc);
-    int32_t ReturnSSRC(const uint32_t ssrc);
+  uint32_t CreateSSRC();
+  void RegisterSSRC(uint32_t ssrc);
+  void ReturnSSRC(uint32_t ssrc);
 
-protected:
-    SSRCDatabase();
-    virtual ~SSRCDatabase();
+ protected:
+  SSRCDatabase();
+  virtual ~SSRCDatabase();
 
-    static SSRCDatabase* CreateInstance() { return new SSRCDatabase(); }
+  static SSRCDatabase* CreateInstance() { return new SSRCDatabase(); }
 
-private:
-    // Friend function to allow the SSRC destructor to be accessed from the
-    // template class.
-    friend SSRCDatabase* GetStaticInstance<SSRCDatabase>(
-        CountOperation count_operation);
-    static SSRCDatabase* StaticInstance(CountOperation count_operation);
+ private:
+  // Friend function to allow the SSRC destructor to be accessed from the
+  // template class.
+  friend SSRCDatabase* GetStaticInstance<SSRCDatabase>(
+      CountOperation count_operation);
 
-    uint32_t GenerateRandom();
-
-    std::map<uint32_t, uint32_t>    _ssrcMap;
-
-    CriticalSectionWrapper* _critSect;
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  Random random_ GUARDED_BY(crit_);
+  std::set<uint32_t> ssrcs_ GUARDED_BY(crit_);
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
diff --git a/webrtc/modules/rtp_rtcp/source/time_util.h b/webrtc/modules/rtp_rtcp/source/time_util.h
new file mode 100644
index 0000000000..5b544ddf9a
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/time_util.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+// Converts NTP timestamp to RTP timestamp.
+inline uint32_t NtpToRtp(NtpTime ntp, uint32_t freq) {
+  uint32_t tmp = (static_cast<uint64_t>(ntp.fractions()) * freq) >> 32;
+  return ntp.seconds() * freq + tmp;
+}
+// Return the current RTP timestamp from the NTP timestamp
+// returned by the specified clock.
+inline uint32_t CurrentRtp(const Clock& clock, uint32_t freq) {
+  return NtpToRtp(NtpTime(clock), freq);
+}
+
+// Helper function for compact ntp representation:
+// RFC 3550, Section 4. Time Format.
+// Wallclock time is represented using the timestamp format of
+// the Network Time Protocol (NTP).
+// ...
+// In some fields where a more compact representation is
+// appropriate, only the middle 32 bits are used; that is, the low 16
+// bits of the integer part and the high 16 bits of the fractional part.
+inline uint32_t CompactNtp(NtpTime ntp) {
+  return (ntp.seconds() << 16) | (ntp.fractions() >> 16);
+}
+// Converts interval between compact ntp timestamps to milliseconds.
+// This interval can be upto ~18.2 hours (2^16 seconds).
+inline uint32_t CompactNtpIntervalToMs(uint32_t compact_ntp_interval) {
+  return static_cast<uint64_t>(compact_ntp_interval) * 1000 / (1 << 16);
+}
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
diff --git a/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc b/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
new file mode 100644
index 0000000000..7efb83ccad
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+TEST(TimeUtilTest, CompactNtp) {
+  const uint32_t kNtpSec = 0x12345678;
+  const uint32_t kNtpFrac = 0x23456789;
+  const NtpTime kNtp(kNtpSec, kNtpFrac);
+  const uint32_t kNtpMid = 0x56782345;
+  EXPECT_EQ(kNtpMid, CompactNtp(kNtp));
+}
+
+TEST(TimeUtilTest, CompactNtpToMs) {
+  const NtpTime ntp1(0x12345, 0x23456);
+  const NtpTime ntp2(0x12654, 0x64335);
+  uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+
+  uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+  EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpToMsWithWrap) {
+  const NtpTime ntp1(0x1ffff, 0x23456);
+  const NtpTime ntp2(0x20000, 0x64335);
+  uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+
+  // While ntp2 > ntp1, there compact ntp presentation happen to be opposite.
+  // That shouldn't be a problem as long as unsigned arithmetic is used.
+  ASSERT_GT(ntp2.ToMs(), ntp1.ToMs());
+  ASSERT_LT(CompactNtp(ntp2), CompactNtp(ntp1));
+
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+  uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+  EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpToMsLarge) {
+  const NtpTime ntp1(0x10000, 0x23456);
+  const NtpTime ntp2(0x1ffff, 0x64335);
+  uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+  // Ntp difference close to maximum of ~18 hours should convert correctly too.
+  ASSERT_GT(ms_diff, 18u * 3600 * 1000);
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+  uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+  EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc b/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
index fb1ed625ed..f994ff7049 100644
--- a/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
+++ b/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -11,8 +11,10 @@
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 
 #include <assert.h>
-#include <limits>
 #include <string.h>
+
+#include <limits>
+
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/video_codec_information.h b/webrtc/modules/rtp_rtcp/source/video_codec_information.h
index 456b3bb934..7b819d060f 100644
--- a/webrtc/modules/rtp_rtcp/source/video_codec_information.h
+++ b/webrtc/modules/rtp_rtcp/source/video_codec_information.h
@@ -15,14 +15,13 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
-class VideoCodecInformation
-{
-public:
-    virtual void Reset() = 0;
+class VideoCodecInformation {
+ public:
+  virtual void Reset() = 0;
 
-    virtual RtpVideoCodecTypes Type() = 0;
-    virtual ~VideoCodecInformation(){};
+  virtual RtpVideoCodecTypes Type() = 0;
+  virtual ~VideoCodecInformation() {}
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
diff --git a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
index feed784839..9721a7e9ac 100644
--- a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
+++ b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
@@ -37,9 +37,8 @@ PartitionTreeNode::PartitionTreeNode(PartitionTreeNode* parent,
 
 PartitionTreeNode* PartitionTreeNode::CreateRootNode(const size_t* size_vector,
                                                      size_t num_partitions) {
-  PartitionTreeNode* root_node =
-      new PartitionTreeNode(NULL, &size_vector[1], num_partitions - 1,
-                            size_vector[0]);
+  PartitionTreeNode* root_node = new PartitionTreeNode(
+      NULL, &size_vector[1], num_partitions - 1, size_vector[0]);
   root_node->set_packet_start(true);
   return root_node;
 }
@@ -54,7 +53,7 @@ int PartitionTreeNode::Cost(size_t penalty) {
   if (num_partitions_ == 0) {
     // This is a solution node.
     cost = std::max(max_parent_size_, this_size_int()) -
-        std::min(min_parent_size_, this_size_int());
+           std::min(min_parent_size_, this_size_int());
   } else {
     cost = std::max(max_parent_size_, this_size_int()) - min_parent_size_;
   }
@@ -68,9 +67,7 @@ bool PartitionTreeNode::CreateChildren(size_t max_size) {
     if (this_size_ + size_vector_[0] <= max_size) {
       assert(!children_[kLeftChild]);
       children_[kLeftChild] =
-          new PartitionTreeNode(this,
-                                &size_vector_[1],
-                                num_partitions_ - 1,
+          new PartitionTreeNode(this, &size_vector_[1], num_partitions_ - 1,
                                 this_size_ + size_vector_[0]);
       children_[kLeftChild]->set_max_parent_size(max_parent_size_);
       children_[kLeftChild]->set_min_parent_size(min_parent_size_);
@@ -80,10 +77,8 @@ bool PartitionTreeNode::CreateChildren(size_t max_size) {
     }
     if (this_size_ > 0) {
       assert(!children_[kRightChild]);
-      children_[kRightChild] = new PartitionTreeNode(this,
-                                                     &size_vector_[1],
-                                                     num_partitions_ - 1,
-                                                     size_vector_[0]);
+      children_[kRightChild] = new PartitionTreeNode(
+          this, &size_vector_[1], num_partitions_ - 1, size_vector_[0]);
       children_[kRightChild]->set_max_parent_size(
           std::max(max_parent_size_, this_size_int()));
       children_[kRightChild]->set_min_parent_size(
@@ -148,7 +143,8 @@ PartitionTreeNode* PartitionTreeNode::GetOptimalNode(size_t max_size,
 
 Vp8PartitionAggregator::Vp8PartitionAggregator(
     const RTPFragmentationHeader& fragmentation,
-    size_t first_partition_idx, size_t last_partition_idx)
+    size_t first_partition_idx,
+    size_t last_partition_idx)
     : root_(NULL),
       num_partitions_(last_partition_idx - first_partition_idx + 1),
       size_vector_(new size_t[num_partitions_]),
@@ -158,14 +154,14 @@ Vp8PartitionAggregator::Vp8PartitionAggregator(
   for (size_t i = 0; i < num_partitions_; ++i) {
     size_vector_[i] =
         fragmentation.fragmentationLength[i + first_partition_idx];
-    largest_partition_size_ = std::max(largest_partition_size_,
-                                       size_vector_[i]);
+    largest_partition_size_ =
+        std::max(largest_partition_size_, size_vector_[i]);
   }
   root_ = PartitionTreeNode::CreateRootNode(size_vector_, num_partitions_);
 }
 
 Vp8PartitionAggregator::~Vp8PartitionAggregator() {
-  delete [] size_vector_;
+  delete[] size_vector_;
   delete root_;
 }
 
@@ -190,14 +186,16 @@ Vp8PartitionAggregator::FindOptimalConfiguration(size_t max_size,
     assert(packet_index > 0);
     assert(temp_node != NULL);
     config_vector[i - 1] = packet_index - 1;
-    if (temp_node->packet_start()) --packet_index;
+    if (temp_node->packet_start())
+      --packet_index;
     temp_node = temp_node->parent();
   }
   return config_vector;
 }
 
 void Vp8PartitionAggregator::CalcMinMax(const ConfigVec& config,
-                                        int* min_size, int* max_size) const {
+                                        int* min_size,
+                                        int* max_size) const {
   if (*min_size < 0) {
     *min_size = std::numeric_limits<int>::max();
   }
@@ -263,8 +261,8 @@ size_t Vp8PartitionAggregator::CalcNumberOfFragments(
   }
   assert(num_fragments > 0);
   // TODO(mflodman) Assert disabled since it's falsely triggered, see issue 293.
-  //assert(large_partition_size / num_fragments + 1 <= max_payload_size);
+  // assert(large_partition_size / num_fragments + 1 <= max_payload_size);
   return num_fragments;
 }
 
-}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
index 53b678f3b9..ccd22e5be2 100644
--- a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
+++ b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -132,6 +132,6 @@ class Vp8PartitionAggregator {
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Vp8PartitionAggregator);
 };
-}  // namespace
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
index 4650c94047..726d83ec50 100644
--- a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
@@ -209,4 +209,4 @@ TEST(Vp8PartitionAggregator, TestCalcNumberOfFragments) {
                 1600, kMTU, 1, 900, 1000));
 }
 
-}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
deleted file mode 100644
index 711be4a623..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// BWEStandAlone.cpp : Defines the entry point for the console application.
-//
-
-#include <stdio.h>
-#include <string>
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/channel_transport/udp_transport.h"
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
-
-//#include "vld.h"
-
-class myTransportCB: public UdpTransportData
-{
-public:
-    myTransportCB (RtpRtcp *rtpMod) : _rtpMod(rtpMod) {};
-protected:
-    // Inherited from UdpTransportData
- void IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                        const size_t rtpPacketLength,
-                        const int8_t* fromIP,
-                        const uint16_t fromPort) override;
-
- void IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                         const size_t rtcpPacketLength,
-                         const int8_t* fromIP,
-                         const uint16_t fromPort) override;
-
-private:
-    RtpRtcp *_rtpMod;
-};
-
-void myTransportCB::IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                                      const size_t rtpPacketLength,
-                                      const int8_t* fromIP,
-                                      const uint16_t fromPort)
-{
-    printf("Receiving RTP from IP %s, port %u\n", fromIP, fromPort);
-    _rtpMod->IncomingPacket((uint8_t *) incomingRtpPacket, rtpPacketLength);
-}
-
-void myTransportCB::IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                                       const size_t rtcpPacketLength,
-                                       const int8_t* fromIP,
-                                       const uint16_t fromPort)
-{
-    printf("Receiving RTCP from IP %s, port %u\n", fromIP, fromPort);
-    _rtpMod->IncomingPacket((uint8_t *) incomingRtcpPacket, rtcpPacketLength);
-}
-
-
-int main(int argc, char* argv[])
-{
-    bool isSender = false;
-    bool isReceiver = false;
-    uint16_t port;
-    std::string ip;
-    TestSenderReceiver *sendrec = new TestSenderReceiver();
-    TestLoadGenerator *gen;
-
-    if (argc == 2)
-    {
-        // receiver only
-        isReceiver = true;
-
-        // read port
-        port = atoi(argv[1]);
-    }
-    else if (argc == 3)
-    {
-        // sender and receiver
-        isSender = true;
-        isReceiver = true;
-
-        // read IP
-        ip = argv[1];
-
-        // read port
-        port = atoi(argv[2]);
-    }
-
-    Trace::CreateTrace();
-    Trace::SetTraceFile("BWEStandAloneTrace.txt");
-    Trace::set_level_filter(webrtc::kTraceAll);
-
-    sendrec->InitReceiver(port);
-
-    sendrec->Start();
-
-    if (isSender)
-    {
-        const uint32_t startRateKbps = 1000;
-        //gen = new CBRGenerator(sendrec, 1000, 500);
-        gen = new CBRFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2);
-        //gen = new PeriodicKeyFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2, 7, 300);
-        //const uint16_t numFrameRates = 5;
-        //const uint8_t frameRates[numFrameRates] = {30, 15, 20, 23, 25};
-        //gen = new CBRVarFRGenerator(sendrec, 1000, frameRates, numFrameRates, 90000, 4.0, 0.1, 0.2);
-        //gen = new CBRFrameDropGenerator(sendrec, startRateKbps, 90000, 0.2);
-        sendrec->SetLoadGenerator(gen);
-        sendrec->InitSender(startRateKbps, ip.c_str(), port);
-        gen->Start();
-    }
-
-    while (1)
-    {
-    }
-
-    if (isSender)
-    {
-        gen->Stop();
-        delete gen;
-    }
-
-    delete sendrec;
-
-    //uint8_t numberOfSocketThreads = 1;
-    //UdpTransport* transport = UdpTransport::Create(0, numberOfSocketThreads);
-
-    //RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(1, false);
-    //if (rtp->InitSender() != 0)
-    //{
-    //    exit(1);
-    //}
-    //if (rtp->RegisterSendTransport(transport) != 0)
-    //{
-    //    exit(1);
-    //}
-
-//    transport->InitializeSendSockets("192.168.200.39", 8000);
-    //transport->InitializeSendSockets("127.0.0.1", 10000);
-    //transport->InitializeSourcePorts(8000);
-
-
-    return(0);
- //   myTransportCB *tp = new myTransportCB(rtp);
- //   transport->InitializeReceiveSockets(tp, 10000, "0.0.0.0");
- //   transport->StartReceiving(500);
-
- //   int8_t data[100];
- //   for (int i = 0; i < 100; data[i] = i++);
-
- //   for (int i = 0; i < 100; i++)
- //   {
- //       transport->SendRaw(data, 100, false);
- //   }
-
-
-
- //   int32_t totTime = 0;
- //   while (totTime < 10000)
- //   {
- //       transport->Process();
- //       int32_t wTime = transport->TimeUntilNextProcess();
- //       totTime += wTime;
- //       Sleep(wTime);
- //   }
-
-
-    //if (transport)
-    //{
-    //    // Destroy the Socket Transport module
-    //    transport->StopReceiving();
-    //    transport->InitializeReceiveSockets(NULL,0);// deregister callback
- //       UdpTransport::Destroy(transport);
-    //    transport = NULL;
- //   }
-
- //   if (tp)
- //   {
- //       delete tp;
- //       tp = NULL;
- //   }
-
- //   if (rtp)
- //   {
- //       RtpRtcp::DestroyRtpRtcp(rtp);
- //       rtp = NULL;
- //   }
-
-
-    //return 0;
-}
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
deleted file mode 100644
index fe54d67ee7..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
+++ /dev/null
@@ -1,1055 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
-
-#include <math.h>
-#include <stdio.h>
-
-#include <algorithm>
-#include <sstream>
-
-#ifdef MATLAB
-#include "engine.h"
-#endif
-
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-using namespace webrtc;
-
-#ifdef MATLAB
-MatlabEngine eng;
-
-MatlabLine::MatlabLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
-:
-_xArray(NULL),
-_yArray(NULL),
-_maxLen(maxLen),
-_plotAttribute(),
-_name()
-{
-    if (_maxLen > 0)
-    {
-        _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-        _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-    }
-
-    if (plotAttrib)
-    {
-        _plotAttribute = plotAttrib;
-    }
-
-    if (name)
-    {
-        _name = name;
-    }
-}
-
-MatlabLine::~MatlabLine()
-{
-    if (_xArray != NULL)
-    {
-        mxDestroyArray(_xArray);
-    }
-    if (_yArray != NULL)
-    {
-        mxDestroyArray(_yArray);
-    }
-}
-
-void MatlabLine::Append(double x, double y)
-{
-    if (_maxLen > 0 && _xData.size() > static_cast<uint32_t>(_maxLen))
-    {
-        _xData.resize(_maxLen);
-        _yData.resize(_maxLen);
-    }
-
-    _xData.push_front(x);
-    _yData.push_front(y);
-}
-
-
-// append y-data with running integer index as x-data
-void MatlabLine::Append(double y)
-{
-    if (_xData.empty())
-    {
-        // first element is index 0
-        Append(0, y);
-    }
-    else
-    {
-        // take last x-value and increment
-        double temp = _xData.back(); // last x-value
-        Append(temp + 1, y);
-    }
-}
-
-
-void MatlabLine::SetMaxLen(int maxLen)
-{
-    if (maxLen <= 0)
-    {
-        // means no maxLen
-        _maxLen = -1;
-    }
-    else
-    {
-        _maxLen = maxLen;
-
-        if (_xArray != NULL)
-        {
-            mxDestroyArray(_xArray);
-            mxDestroyArray(_yArray);
-        }
-        _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-        _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-
-        maxLen = ((unsigned int)maxLen <= _xData.size()) ? maxLen : (int)_xData.size();
-        _xData.resize(maxLen);
-        _yData.resize(maxLen);
-
-        //// reserve the right amount of memory
-        //_xData.reserve(_maxLen);
-        //_yData.reserve(_maxLen);
-    }
-}
-
-void MatlabLine::SetAttribute(char *plotAttrib)
-{
-    _plotAttribute = plotAttrib;
-}
-
-void MatlabLine::SetName(char *name)
-{
-    _name = name;
-}
-
-void MatlabLine::GetPlotData(mxArray** xData, mxArray** yData)
-{
-    // Make sure we have enough Matlab allocated memory.
-    // Assuming both arrays (x and y) are of the same size.
-    if (_xData.empty())
-    {
-        return; // No data
-    }
-    unsigned int size = 0;
-    if (_xArray != NULL)
-    {
-        size = (unsigned int)mxGetNumberOfElements(_xArray);
-    }
-    if (size < _xData.size())
-    {
-        if (_xArray != NULL)
-        {
-            mxDestroyArray(_xArray);
-            mxDestroyArray(_yArray);
-        }
-        _xArray = mxCreateDoubleMatrix(1, _xData.size(), mxREAL);
-        _yArray = mxCreateDoubleMatrix(1, _yData.size(), mxREAL);
-    }
-
-    if (!_xData.empty())
-    {
-        double* x = mxGetPr(_xArray);
-
-        std::list<double>::iterator it = _xData.begin();
-
-        for (int i = 0; it != _xData.end(); it++, i++)
-        {
-            x[i] = *it;
-        }
-    }
-
-    if (!_yData.empty())
-    {
-        double* y = mxGetPr(_yArray);
-
-        std::list<double>::iterator it = _yData.begin();
-
-        for (int i = 0; it != _yData.end(); it++, i++)
-        {
-            y[i] = *it;
-        }
-    }
-    *xData = _xArray;
-    *yData = _yArray;
-}
-
-std::string MatlabLine::GetXName()
-{
-    std::ostringstream xString;
-    xString << "x_" << _name;
-    return xString.str();
-}
-
-std::string MatlabLine::GetYName()
-{
-    std::ostringstream yString;
-    yString << "y_" << _name;
-    return yString.str();
-}
-
-std::string MatlabLine::GetPlotString()
-{
-
-    std::ostringstream s;
-
-    if (_xData.size() == 0)
-    {
-        s << "[0 1], [0 1]"; // To get an empty plot
-    }
-    else
-    {
-        s << GetXName() << "(1:" << _xData.size() << "),";
-        s << GetYName() << "(1:" << _yData.size() << ")";
-    }
-
-    s << ", '";
-    s << _plotAttribute;
-    s << "'";
-
-    return s.str();
-}
-
-std::string MatlabLine::GetRefreshString()
-{
-    std::ostringstream s;
-
-    if (_xData.size() > 0)
-    {
-        s << "set(h,'xdata',"<< GetXName() <<"(1:" << _xData.size() << "),'ydata',"<< GetYName() << "(1:" << _yData.size() << "));";
-    }
-    else
-    {
-        s << "set(h,'xdata',[NaN],'ydata',[NaN]);";
-    }
-    return s.str();
-}
-
-std::string MatlabLine::GetLegendString()
-{
-    return ("'" + _name + "'");
-}
-
-bool MatlabLine::hasLegend()
-{
-    return (!_name.empty());
-}
-
-
-// remove data points, but keep attributes
-void MatlabLine::Reset()
-{
-    _xData.clear();
-    _yData.clear();
-}
-
-
-void MatlabLine::UpdateTrendLine(MatlabLine * sourceData, double slope, double offset)
-{
-    Reset(); // reset data, not attributes and name
-
-    double thexMin = sourceData->xMin();
-    double thexMax = sourceData->xMax();
-    Append(thexMin, thexMin * slope + offset);
-    Append(thexMax, thexMax * slope + offset);
-}
-
-double MatlabLine::xMin()
-{
-    if (!_xData.empty())
-    {
-        std::list<double>::iterator theStart = _xData.begin();
-        std::list<double>::iterator theEnd = _xData.end();
-        return(*min_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-double MatlabLine::xMax()
-{
-    if (!_xData.empty())
-    {
-        std::list<double>::iterator theStart = _xData.begin();
-        std::list<double>::iterator theEnd = _xData.end();
-        return(*max_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-double MatlabLine::yMin()
-{
-    if (!_yData.empty())
-    {
-        std::list<double>::iterator theStart = _yData.begin();
-        std::list<double>::iterator theEnd = _yData.end();
-        return(*min_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-double MatlabLine::yMax()
-{
-    if (!_yData.empty())
-    {
-        std::list<double>::iterator theStart = _yData.begin();
-        std::list<double>::iterator theEnd = _yData.end();
-        return(*max_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-
-
-MatlabTimeLine::MatlabTimeLine(int horizonSeconds /*= -1*/, const char *plotAttrib /*= NULL*/,
-                               const char *name /*= NULL*/,
-                               int64_t refTimeMs /* = -1*/)
-                               :
-_timeHorizon(horizonSeconds),
-MatlabLine(-1, plotAttrib, name) // infinite number of elements
-{
-    if (refTimeMs < 0)
-        _refTimeMs = TickTime::MillisecondTimestamp();
-    else
-        _refTimeMs = refTimeMs;
-}
-
-void MatlabTimeLine::Append(double y)
-{
-    MatlabLine::Append(static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0, y);
-
-    PurgeOldData();
-}
-
-
-void MatlabTimeLine::PurgeOldData()
-{
-    if (_timeHorizon > 0)
-    {
-        // remove old data
-        double historyLimit = static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0
-            - _timeHorizon; // remove data points older than this
-
-        std::list<double>::reverse_iterator ritx = _xData.rbegin();
-        uint32_t removeCount = 0;
-        while (ritx != _xData.rend())
-        {
-            if (*ritx >= historyLimit)
-            {
-                break;
-            }
-            ritx++;
-            removeCount++;
-        }
-        if (removeCount == 0)
-        {
-            return;
-        }
-
-        // remove the range [begin, it).
-        //if (removeCount > 10)
-        //{
-        //    printf("Removing %lu elements\n", removeCount);
-        //}
-        _xData.resize(_xData.size() - removeCount);
-        _yData.resize(_yData.size() - removeCount);
-    }
-}
-
-
-int64_t MatlabTimeLine::GetRefTime()
-{
-    return(_refTimeMs);
-}
-
-
-
-
-MatlabPlot::MatlabPlot()
-:
-_figHandle(-1),
-_smartAxis(false),
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_timeToPlot(false),
-_plotting(false),
-_enabled(true),
-_firstPlot(true),
-_legendEnabled(true),
-_donePlottingEvent(EventWrapper::Create())
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _xlim[0] = 0;
-    _xlim[1] = 0;
-    _ylim[0] = 0;
-    _ylim[1] = 0;
-
-#ifdef PLOT_TESTING
-    _plotStartTime = -1;
-    _plotDelay = 0;
-#endif
-
-}
-
-
-MatlabPlot::~MatlabPlot()
-{
-    _critSect->Enter();
-
-    // delete all line objects
-    while (!_line.empty())
-    {
-        delete *(_line.end() - 1);
-        _line.pop_back();
-    }
-
-    delete _critSect;
-    delete _donePlottingEvent;
-}
-
-
-int MatlabPlot::AddLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
-{
-    CriticalSectionScoped cs(_critSect);
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    MatlabLine *newLine = new MatlabLine(maxLen, plotAttrib, name);
-    _line.push_back(newLine);
-
-    return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
-}
-
-
-int MatlabPlot::AddTimeLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/,
-                            int64_t refTimeMs /*= -1*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    MatlabTimeLine *newLine = new MatlabTimeLine(maxLen, plotAttrib, name, refTimeMs);
-    _line.push_back(newLine);
-
-    return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
-}
-
-
-int MatlabPlot::GetLineIx(const char *name)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    // search the list for a matching line name
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    bool matchFound = false;
-    int lineIx = 0;
-
-    for (; it != _line.end(); it++, lineIx++)
-    {
-        if ((*it)->_name == name)
-        {
-            matchFound = true;
-            break;
-        }
-    }
-
-    if (matchFound)
-    {
-        return (lineIx);
-    }
-    else
-    {
-        return (-1);
-    }
-}
-
-
-void MatlabPlot::Append(int lineIndex, double x, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return;
-    }
-
-    // sanity for index
-    if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
-    {
-        throw "Line index out of range";
-        exit(1);
-    }
-
-    return (_line[lineIndex]->Append(x, y));
-}
-
-
-void MatlabPlot::Append(int lineIndex, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return;
-    }
-
-    // sanity for index
-    if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
-    {
-        throw "Line index out of range";
-        exit(1);
-    }
-
-    return (_line[lineIndex]->Append(y));
-}
-
-
-int MatlabPlot::Append(const char *name, double x, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    // search the list for a matching line name
-    int lineIx = GetLineIx(name);
-
-    if (lineIx < 0) //(!matchFound)
-    {
-        // no match; append new line
-        lineIx = AddLine(-1, NULL, name);
-    }
-
-    // append data to line
-    Append(lineIx, x, y);
-    return (lineIx);
-}
-
-int MatlabPlot::Append(const char *name, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    // search the list for a matching line name
-    int lineIx = GetLineIx(name);
-
-    if (lineIx < 0) //(!matchFound)
-    {
-        // no match; append new line
-        lineIx = AddLine(-1, NULL, name);
-    }
-
-    // append data to line
-    Append(lineIx, y);
-    return (lineIx);
-}
-
-int MatlabPlot::Length(char *name)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    int ix = GetLineIx(name);
-    if (ix >= 0)
-    {
-        return (static_cast<int>(_line[ix]->_xData.size()));
-    }
-    else
-    {
-        return (-1);
-    }
-}
-
-
-void MatlabPlot::SetPlotAttribute(char *name, char *plotAttrib)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return;
-    }
-
-    int lineIx = GetLineIx(name);
-
-    if (lineIx >= 0)
-    {
-        _line[lineIx]->SetAttribute(plotAttrib);
-    }
-}
-
-// Must be called under critical section _critSect
-void MatlabPlot::UpdateData(Engine* ep)
-{
-    if (!_enabled)
-    {
-        return;
-    }
-
-    for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-    {
-        mxArray* xData = NULL;
-        mxArray* yData = NULL;
-        (*it)->GetPlotData(&xData, &yData);
-        if (xData != NULL)
-        {
-            std::string xName = (*it)->GetXName();
-            std::string yName = (*it)->GetYName();
-            _critSect->Leave();
-#ifdef MATLAB6
-            mxSetName(xData, xName.c_str());
-            mxSetName(yData, yName.c_str());
-            engPutArray(ep, xData);
-            engPutArray(ep, yData);
-#else
-            int ret = engPutVariable(ep, xName.c_str(), xData);
-            assert(ret == 0);
-            ret = engPutVariable(ep, yName.c_str(), yData);
-            assert(ret == 0);
-#endif
-            _critSect->Enter();
-        }
-    }
-}
-
-bool MatlabPlot::GetPlotCmd(std::ostringstream & cmd, Engine* ep)
-{
-    _critSect->Enter();
-
-    if (!DataAvailable())
-    {
-        return false;
-    }
-
-    if (_firstPlot)
-    {
-        GetPlotCmd(cmd);
-        _firstPlot = false;
-    }
-    else
-    {
-        GetRefreshCmd(cmd);
-    }
-
-    UpdateData(ep);
-
-    _critSect->Leave();
-
-    return true;
-}
-
-// Call inside critsect
-void MatlabPlot::GetPlotCmd(std::ostringstream & cmd)
-{
-    // we have something to plot
-    // empty the stream
-    cmd.str(""); // (this seems to be the only way)
-
-    cmd << "figure; h" << _figHandle << "= plot(";
-
-    // first line
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    cmd << (*it)->GetPlotString();
-
-    it++;
-
-    // remaining lines
-    for (; it != _line.end(); it++)
-    {
-        cmd << ", ";
-        cmd << (*it)->GetPlotString();
-    }
-
-    cmd << "); ";
-
-    if (_legendEnabled)
-    {
-        GetLegendCmd(cmd);
-    }
-
-    if (_smartAxis)
-    {
-        double xMin = _xlim[0];
-        double xMax = _xlim[1];
-        double yMax = _ylim[1];
-        for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-        {
-            xMax = std::max(xMax, (*it)->xMax());
-            xMin = std::min(xMin, (*it)->xMin());
-
-            yMax = std::max(yMax, (*it)->yMax());
-            yMax = std::max(yMax, fabs((*it)->yMin()));
-        }
-        _xlim[0] = xMin;
-        _xlim[1] = xMax;
-        _ylim[0] = -yMax;
-        _ylim[1] = yMax;
-
-        cmd << "axis([" << _xlim[0] << ", " << _xlim[1] << ", " << _ylim[0] << ", " << _ylim[1] << "]);";
-    }
-
-    int i=1;
-    for (it = _line.begin(); it != _line.end(); i++, it++)
-    {
-        cmd << "set(h" << _figHandle << "(" << i << "), 'Tag', " << (*it)->GetLegendString() << ");";
-    }
-}
-
-// Call inside critsect
-void MatlabPlot::GetRefreshCmd(std::ostringstream & cmd)
-{
-    cmd.str(""); // (this seems to be the only way)
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    for (it = _line.begin(); it != _line.end(); it++)
-    {
-        cmd << "h = findobj(0, 'Tag', " << (*it)->GetLegendString() << ");";
-        cmd << (*it)->GetRefreshString();
-    }
-    //if (_legendEnabled)
-    //{
-    //    GetLegendCmd(cmd);
-    //}
-}
-
-void MatlabPlot::GetLegendCmd(std::ostringstream & cmd)
-{
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    bool anyLegend = false;
-    for (; it != _line.end(); it++)
-    {
-        anyLegend = anyLegend || (*it)->hasLegend();
-    }
-    if (anyLegend)
-    {
-        // create the legend
-
-        cmd << "legend(h" << _figHandle << ",{";
-
-
-        // iterate lines
-        int i = 0;
-        for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-        {
-            if (i > 0)
-            {
-                cmd << ", ";
-            }
-            cmd << (*it)->GetLegendString();
-            i++;
-        }
-
-        cmd << "}, 2); "; // place legend in upper-left corner
-    }
-}
-
-// Call inside critsect
-bool MatlabPlot::DataAvailable()
-{
-    if (!_enabled)
-    {
-        return false;
-    }
-
-    for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-    {
-        (*it)->PurgeOldData();
-    }
-
-    return true;
-}
-
-void MatlabPlot::Plot()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _timeToPlot = true;
-
-#ifdef PLOT_TESTING
-    _plotStartTime = TickTime::MillisecondTimestamp();
-#endif
-}
-
-
-void MatlabPlot::Reset()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _enabled = true;
-
-    for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-    {
-        (*it)->Reset();
-    }
-
-}
-
-void MatlabPlot::SetFigHandle(int handle)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (handle > 0)
-        _figHandle = handle;
-}
-
-bool
-MatlabPlot::TimeToPlot()
-{
-    CriticalSectionScoped cs(_critSect);
-    return _enabled && _timeToPlot;
-}
-
-void
-MatlabPlot::Plotting()
-{
-    CriticalSectionScoped cs(_critSect);
-    _plotting = true;
-}
-
-void
-MatlabPlot::DonePlotting()
-{
-    CriticalSectionScoped cs(_critSect);
-    _timeToPlot = false;
-    _plotting = false;
-    _donePlottingEvent->Set();
-}
-
-void
-MatlabPlot::DisablePlot()
-{
-    _critSect->Enter();
-    while (_plotting)
-    {
-        _critSect->Leave();
-        _donePlottingEvent->Wait(WEBRTC_EVENT_INFINITE);
-        _critSect->Enter();
-    }
-    _enabled = false;
-}
-
-int MatlabPlot::MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    int sourceIx;
-    int trendIx;
-
-    sourceIx = GetLineIx(sourceName);
-    if (sourceIx < 0)
-    {
-        // could not find source
-        return (-1);
-    }
-
-    trendIx = GetLineIx(trendName);
-    if (trendIx < 0)
-    {
-        // no trend found; add new line
-        trendIx = AddLine(2 /*maxLen*/, plotAttrib, trendName);
-    }
-
-    _line[trendIx]->UpdateTrendLine(_line[sourceIx], slope, offset);
-
-    return (trendIx);
-
-}
-
-
-MatlabEngine::MatlabEngine()
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_running(false),
-_numPlots(0)
-{
-    _eventPtr = EventWrapper::Create();
-
-    _plotThread = ThreadWrapper::CreateThread(MatlabEngine::PlotThread, this,
-                                              kLowPriority, "MatlabPlot");
-    _running = true;
-    _plotThread->Start();
-}
-
-MatlabEngine::~MatlabEngine()
-{
-    _critSect->Enter();
-
-    if (_plotThread)
-    {
-        _running = false;
-        _eventPtr->Set();
-
-        _plotThread->Stop();
-    }
-
-    _plots.clear();
-
-    delete _eventPtr;
-    _eventPtr = NULL;
-
-    _critSect->Leave();
-    delete _critSect;
-
-}
-
-MatlabPlot * MatlabEngine::NewPlot(MatlabPlot *newPlot)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    //MatlabPlot *newPlot = new MatlabPlot();
-
-    if (newPlot)
-    {
-        newPlot->SetFigHandle(++_numPlots); // first plot is number 1
-        _plots.push_back(newPlot);
-    }
-
-    return (newPlot);
-
-}
-
-
-void MatlabEngine::DeletePlot(MatlabPlot *plot)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (plot == NULL)
-    {
-        return;
-    }
-
-    std::vector<MatlabPlot *>::iterator it;
-    for (it = _plots.begin(); it < _plots.end(); it++)
-    {
-        if (plot == *it)
-        {
-            break;
-        }
-    }
-
-    assert (plot == *it);
-
-    (*it)->DisablePlot();
-
-    _plots.erase(it);
-    --_numPlots;
-
-    delete plot;
-}
-
-
-bool MatlabEngine::PlotThread(void *obj)
-{
-    if (!obj)
-    {
-        return (false);
-    }
-
-    MatlabEngine *eng = (MatlabEngine *) obj;
-
-    Engine *ep = engOpen(NULL);
-    if (!ep)
-    {
-        throw "Cannot open Matlab engine";
-        return (false);
-    }
-
-    engSetVisible(ep, true);
-    engEvalString(ep, "close all;");
-
-    while (eng->_running)
-    {
-        eng->_critSect->Enter();
-
-        // iterate through all plots
-        for (unsigned int ix = 0; ix < eng->_plots.size(); ix++)
-        {
-            MatlabPlot *plot = eng->_plots[ix];
-            if (plot->TimeToPlot())
-            {
-                plot->Plotting();
-                eng->_critSect->Leave();
-                std::ostringstream cmd;
-
-                if (engEvalString(ep, cmd.str().c_str()))
-                {
-                    // engine dead
-                    return (false);
-                }
-
-                // empty the stream
-                cmd.str(""); // (this seems to be the only way)
-                if (plot->GetPlotCmd(cmd, ep))
-                {
-                    // things to plot, we have already accessed what we need in the plot
-                    plot->DonePlotting();
-
-                    int64_t start = TickTime::MillisecondTimestamp();
-                    // plot it
-                    int ret = engEvalString(ep, cmd.str().c_str());
-                    printf("time=%I64i\n", TickTime::MillisecondTimestamp() - start);
-                    if (ret)
-                    {
-                        // engine dead
-                        return (false);
-                    }
-
-#ifdef PLOT_TESTING
-                    if(plot->_plotStartTime >= 0)
-                    {
-                        plot->_plotDelay = TickTime::MillisecondTimestamp() - plot->_plotStartTime;
-                        plot->_plotStartTime = -1;
-                    }
-#endif
-                }
-                eng->_critSect->Enter();
-            }
-        }
-
-        eng->_critSect->Leave();
-        // wait a while
-        eng->_eventPtr->Wait(66); // 33 ms
-    }
-
-    if (ep)
-    {
-        engClose(ep);
-        ep = NULL;
-    }
-
-    return (true);
-
-}
-
-#endif // MATLAB
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
deleted file mode 100644
index 3ed89f8f91..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
+++ /dev/null
@@ -1,170 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
-
-#include <list>
-#include <string>
-#include <vector>
-
-#include "webrtc/typedefs.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-}
-
-//#define PLOT_TESTING
-
-#ifdef MATLAB
-
-typedef struct engine Engine;
-typedef struct mxArray_tag mxArray;
-
-class MatlabLine
-{
-    friend class MatlabPlot;
-
-public:
-    MatlabLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
-    ~MatlabLine();
-    virtual void Append(double x, double y);
-    virtual void Append(double y);
-    void SetMaxLen(int maxLen);
-    void SetAttribute(char *plotAttrib);
-    void SetName(char *name);
-    void Reset();
-    virtual void PurgeOldData() {};
-
-    void UpdateTrendLine(MatlabLine * sourceData, double slope, double offset);
-
-    double xMin();
-    double xMax();
-    double yMin();
-    double yMax();
-
-protected:
-    void GetPlotData(mxArray** xData, mxArray** yData);
-    std::string GetXName();
-    std::string GetYName();
-    std::string GetPlotString();
-    std::string GetRefreshString();
-    std::string GetLegendString();
-    bool hasLegend();
-    std::list<double> _xData;
-    std::list<double> _yData;
-    mxArray* _xArray;
-    mxArray* _yArray;
-    int _maxLen;
-    std::string _plotAttribute;
-    std::string _name;
-};
-
-
-class MatlabTimeLine : public MatlabLine
-{
-public:
-    MatlabTimeLine(int horizonSeconds = -1, const char *plotAttrib = NULL, const char *name = NULL,
-        int64_t refTimeMs = -1);
-    ~MatlabTimeLine() {};
-    void Append(double y);
-    void PurgeOldData();
-    int64_t GetRefTime();
-
-private:
-    int64_t _refTimeMs;
-    int _timeHorizon;
-};
-
-
-class MatlabPlot
-{
-    friend class MatlabEngine;
-
-public:
-    MatlabPlot();
-    ~MatlabPlot();
-
-    int AddLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
-    int AddTimeLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL,
-        int64_t refTimeMs = -1);
-    int GetLineIx(const char *name);
-    void Append(int lineIndex, double x, double y);
-    void Append(int lineIndex, double y);
-    int Append(const char *name, double x, double y);
-    int Append(const char *name, double y);
-    int Length(char *name);
-    void SetPlotAttribute(char *name, char *plotAttrib);
-    void Plot();
-    void Reset();
-    void SmartAxis(bool status = true) { _smartAxis = status; };
-    void SetFigHandle(int handle);
-    void EnableLegend(bool enable) { _legendEnabled = enable; };
-
-    bool TimeToPlot();
-    void Plotting();
-    void DonePlotting();
-    void DisablePlot();
-
-    int MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib = NULL);
-
-#ifdef PLOT_TESTING
-    int64_t _plotStartTime;
-    int64_t _plotDelay;
-#endif
-
-private:
-    void UpdateData(Engine* ep);
-    bool GetPlotCmd(std::ostringstream & cmd, Engine* ep);
-    void GetPlotCmd(std::ostringstream & cmd); // call inside crit sect
-    void GetRefreshCmd(std::ostringstream & cmd); // call inside crit sect
-    void GetLegendCmd(std::ostringstream & cmd);
-    bool DataAvailable();
-
-    std::vector<MatlabLine *> _line;
-    int _figHandle;
-    bool _smartAxis;
-    double _xlim[2];
-    double _ylim[2];
-    webrtc::CriticalSectionWrapper *_critSect;
-    bool _timeToPlot;
-    bool _plotting;
-    bool _enabled;
-    bool _firstPlot;
-    bool _legendEnabled;
-    webrtc::EventWrapper* _donePlottingEvent;
-};
-
-
-class MatlabEngine
-{
-public:
-    MatlabEngine();
-    ~MatlabEngine();
-
-    MatlabPlot * NewPlot(MatlabPlot *newPlot);
-    void DeletePlot(MatlabPlot *plot);
-
-private:
-    static bool PlotThread(void *obj);
-
-    std::vector<MatlabPlot *> _plots;
-    webrtc::CriticalSectionWrapper *_critSect;
-    webrtc::EventWrapper *_eventPtr;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _plotThread;
-    bool _running;
-    int _numPlots;
-};
-
-#endif //MATLAB
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
deleted file mode 100644
index 0e3e87976c..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
+++ /dev/null
@@ -1,432 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
-
-#include <stdio.h>
-
-#include <algorithm>
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-
-bool SenderThreadFunction(void *obj)
-{
-    if (obj == NULL)
-    {
-        return false;
-    }
-    TestLoadGenerator *_genObj = static_cast<TestLoadGenerator *>(obj);
-
-    return _genObj->GeneratorLoop();
-}
-
-
-TestLoadGenerator::TestLoadGenerator(TestSenderReceiver *sender, int32_t rtpSampleRate)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_bitrateKbps(0),
-_sender(sender),
-_running(false),
-_rtpSampleRate(rtpSampleRate)
-{
-}
-
-TestLoadGenerator::~TestLoadGenerator ()
-{
-    if (_running)
-    {
-        Stop();
-    }
-
-    delete _critSect;
-}
-
-int32_t TestLoadGenerator::SetBitrate (int32_t newBitrateKbps)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (newBitrateKbps < 0)
-    {
-        return -1;
-    }
-
-    _bitrateKbps = newBitrateKbps;
-
-    printf("New bitrate = %i kbps\n", _bitrateKbps);
-
-    return _bitrateKbps;
-}
-
-
-int32_t TestLoadGenerator::Start (const char *threadName)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _eventPtr = EventWrapper::Create();
-
-    _genThread = ThreadWrapper::CreateThread(SenderThreadFunction, this,
-                                             threadName);
-    _running = true;
-
-    _genThread->Start();
-    _genThread->SetPriority(kRealtimePriority);
-
-    return 0;
-}
-
-
-int32_t TestLoadGenerator::Stop ()
-{
-    _critSect.Enter();
-
-    if (_genThread)
-    {
-        _running = false;
-        _eventPtr->Set();
-
-        _genThread->Stop();
-        _genThread.reset();
-
-        delete _eventPtr;
-        _eventPtr = NULL;
-    }
-
-    _critSect.Leave();
-    return (0);
-}
-
-
-int TestLoadGenerator::generatePayload ()
-{
-    return(generatePayload( static_cast<uint32_t>( TickTime::MillisecondTimestamp() * _rtpSampleRate / 1000 )));
-}
-
-
-int TestLoadGenerator::sendPayload (const uint32_t timeStamp,
-                                    const uint8_t* payloadData,
-                                    const size_t payloadSize,
-                                    const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
-{
-
-    return (_sender->SendOutgoingData(timeStamp, payloadData, payloadSize, frameType));
-}
-
-
-CBRGenerator::CBRGenerator (TestSenderReceiver *sender,
-                            size_t payloadSizeBytes,
-                            int32_t bitrateKbps,
-                            int32_t rtpSampleRate)
-:
-//_eventPtr(NULL),
-_payloadSizeBytes(payloadSizeBytes),
-_payload(new uint8_t[payloadSizeBytes]),
-TestLoadGenerator(sender, rtpSampleRate)
-{
-    SetBitrate (bitrateKbps);
-}
-
-CBRGenerator::~CBRGenerator ()
-{
-    if (_running)
-    {
-        Stop();
-    }
-
-    if (_payload)
-    {
-        delete [] _payload;
-    }
-
-}
-
-bool CBRGenerator::GeneratorLoop ()
-{
-    double periodMs;
-    int64_t nextSendTime = TickTime::MillisecondTimestamp();
-
-
-    // no critSect
-    while (_running)
-    {
-        // send data (critSect inside)
-        generatePayload( static_cast<uint32_t>(nextSendTime * _rtpSampleRate / 1000) );
-
-        // calculate wait time
-        periodMs = 8.0 * _payloadSizeBytes / ( _bitrateKbps );
-
-        nextSendTime = static_cast<int64_t>(nextSendTime + periodMs);
-
-        int32_t waitTime = static_cast<int32_t>(nextSendTime - TickTime::MillisecondTimestamp());
-        if (waitTime < 0)
-        {
-            waitTime = 0;
-        }
-        // wait
-        _eventPtr->Wait(static_cast<int32_t>(waitTime));
-    }
-
-    return true;
-}
-
-int CBRGenerator::generatePayload ( uint32_t timestamp )
-{
-    CriticalSectionScoped cs(_critSect);
-
-    //uint8_t *payload = new uint8_t[_payloadSizeBytes];
-
-    int ret = sendPayload(timestamp, _payload, _payloadSizeBytes);
-
-    //delete [] payload;
-    return ret;
-}
-
-
-
-
-/////////////////////
-
-CBRFixFRGenerator::CBRFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps,
-                                      int32_t rtpSampleRate, int32_t frameRateFps /*= 30*/,
-                                      double spread /*= 0.0*/)
-:
-//_eventPtr(NULL),
-_payloadSizeBytes(0),
-_payload(NULL),
-_payloadAllocLen(0),
-_frameRateFps(frameRateFps),
-_spreadFactor(spread),
-TestLoadGenerator(sender, rtpSampleRate)
-{
-    SetBitrate (bitrateKbps);
-}
-
-CBRFixFRGenerator::~CBRFixFRGenerator ()
-{
-    if (_running)
-    {
-        Stop();
-    }
-
-    if (_payload)
-    {
-        delete [] _payload;
-        _payloadAllocLen = 0;
-    }
-
-}
-
-bool CBRFixFRGenerator::GeneratorLoop ()
-{
-    double periodMs;
-    int64_t nextSendTime = TickTime::MillisecondTimestamp();
-
-    _critSect.Enter();
-
-    if (_frameRateFps <= 0)
-    {
-        return false;
-    }
-
-    _critSect.Leave();
-
-    // no critSect
-    while (_running)
-    {
-        _critSect.Enter();
-
-        // calculate payload size
-        _payloadSizeBytes = nextPayloadSize();
-
-        if (_payloadSizeBytes > 0)
-        {
-
-            if (_payloadAllocLen < _payloadSizeBytes * (1 + _spreadFactor))
-            {
-                // re-allocate _payload
-                if (_payload)
-                {
-                    delete [] _payload;
-                    _payload = NULL;
-                }
-
-                _payloadAllocLen = static_cast<int32_t>((_payloadSizeBytes * (1 + _spreadFactor) * 3) / 2 + .5); // 50% extra to avoid frequent re-alloc
-                _payload = new uint8_t[_payloadAllocLen];
-            }
-
-
-            // send data (critSect inside)
-            generatePayload( static_cast<uint32_t>(nextSendTime * _rtpSampleRate / 1000) );
-        }
-
-        _critSect.Leave();
-
-        // calculate wait time
-        periodMs = 1000.0 / _frameRateFps;
-        nextSendTime = static_cast<int64_t>(nextSendTime + periodMs + 0.5);
-
-        int32_t waitTime = static_cast<int32_t>(nextSendTime - TickTime::MillisecondTimestamp());
-        if (waitTime < 0)
-        {
-            waitTime = 0;
-        }
-        // wait
-        _eventPtr->Wait(waitTime);
-    }
-
-    return true;
-}
-
-size_t CBRFixFRGenerator::nextPayloadSize()
-{
-    const double periodMs = 1000.0 / _frameRateFps;
-    return static_cast<size_t>(_bitrateKbps * periodMs / 8 + 0.5);
-}
-
-int CBRFixFRGenerator::generatePayload ( uint32_t timestamp )
-{
-    CriticalSectionScoped cs(_critSect);
-
-    double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
-    factor = 1 + 2 * _spreadFactor * factor; // [1 - _spreadFactor ; 1 + _spreadFactor]
-
-    size_t thisPayloadBytes = static_cast<size_t>(_payloadSizeBytes * factor);
-    // sanity
-    if (thisPayloadBytes > _payloadAllocLen)
-    {
-        thisPayloadBytes = _payloadAllocLen;
-    }
-
-    int ret = sendPayload(timestamp, _payload, thisPayloadBytes);
-    return ret;
-}
-
-
-/////////////////////
-
-PeriodicKeyFixFRGenerator::PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps,
-                                                      int32_t rtpSampleRate, int32_t frameRateFps /*= 30*/,
-                                                      double spread /*= 0.0*/, double keyFactor /*= 4.0*/, uint32_t keyPeriod /*= 300*/)
-:
-_keyFactor(keyFactor),
-_keyPeriod(keyPeriod),
-_frameCount(0),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRateFps, spread)
-{
-}
-
-size_t PeriodicKeyFixFRGenerator::nextPayloadSize()
-{
-    // calculate payload size for a delta frame
-    size_t payloadSizeBytes = static_cast<size_t>(1000 * _bitrateKbps /
-        (8.0 * _frameRateFps * (1.0 + (_keyFactor - 1.0) / _keyPeriod)) + 0.5);
-
-    if (_frameCount % _keyPeriod == 0)
-    {
-        // this is a key frame, scale the payload size
-        payloadSizeBytes =
-            static_cast<size_t>(_keyFactor * _payloadSizeBytes + 0.5);
-    }
-    _frameCount++;
-
-    return payloadSizeBytes;
-}
-
-////////////////////
-
-CBRVarFRGenerator::CBRVarFRGenerator(TestSenderReceiver *sender, int32_t bitrateKbps, const uint8_t* frameRates,
-                                     uint16_t numFrameRates, int32_t rtpSampleRate, double avgFrPeriodMs,
-                                     double frSpreadFactor, double spreadFactor)
-:
-_avgFrPeriodMs(avgFrPeriodMs),
-_frSpreadFactor(frSpreadFactor),
-_frameRates(NULL),
-_numFrameRates(numFrameRates),
-_frChangeTimeMs(TickTime::MillisecondTimestamp() + _avgFrPeriodMs),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRates[0], spreadFactor)
-{
-    _frameRates = new uint8_t[_numFrameRates];
-    memcpy(_frameRates, frameRates, _numFrameRates);
-}
-
-CBRVarFRGenerator::~CBRVarFRGenerator()
-{
-    delete [] _frameRates;
-}
-
-void CBRVarFRGenerator::ChangeFrameRate()
-{
-    const int64_t nowMs = TickTime::MillisecondTimestamp();
-    if (nowMs < _frChangeTimeMs)
-    {
-        return;
-    }
-    // Time to change frame rate
-    uint16_t frIndex = static_cast<uint16_t>(static_cast<double>(rand()) / RAND_MAX
-                                            * (_numFrameRates - 1) + 0.5) ;
-    assert(frIndex < _numFrameRates);
-    _frameRateFps = _frameRates[frIndex];
-    // Update the next frame rate change time
-    double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
-    factor = 1 + 2 * _frSpreadFactor * factor; // [1 - _frSpreadFactor ; 1 + _frSpreadFactor]
-    _frChangeTimeMs = nowMs + static_cast<int64_t>(1000.0 * factor *
-                                                   _avgFrPeriodMs + 0.5);
-
-    printf("New frame rate: %d\n", _frameRateFps);
-}
-
-size_t CBRVarFRGenerator::nextPayloadSize()
-{
-    ChangeFrameRate();
-    return CBRFixFRGenerator::nextPayloadSize();
-}
-
-////////////////////
-
-CBRFrameDropGenerator::CBRFrameDropGenerator(TestSenderReceiver *sender, int32_t bitrateKbps,
-                                         int32_t rtpSampleRate, double spreadFactor)
-:
-_accBits(0),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, 30, spreadFactor)
-{
-}
-
-CBRFrameDropGenerator::~CBRFrameDropGenerator()
-{
-}
-
-size_t CBRFrameDropGenerator::nextPayloadSize()
-{
-    _accBits -= 1000 * _bitrateKbps / _frameRateFps;
-    if (_accBits < 0)
-    {
-        _accBits = 0;
-    }
-    if (_accBits > 0.3 * _bitrateKbps * 1000)
-    {
-        //printf("drop\n");
-        return 0;
-    }
-    else
-    {
-        //printf("keep\n");
-        const double periodMs = 1000.0 / _frameRateFps;
-        size_t frameSize =
-            static_cast<size_t>(_bitrateKbps * periodMs / 8 + 0.5);
-        frameSize =
-            std::max(frameSize, static_cast<size_t>(300 * periodMs / 8 + 0.5));
-        _accBits += frameSize * 8;
-        return frameSize;
-    }
-}
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
deleted file mode 100644
index bd83962fa3..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
-
-#include <stdlib.h>
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
-#include "webrtc/typedefs.h"
-
-class TestSenderReceiver;
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-}
-
-class TestLoadGenerator
-{
-public:
-    TestLoadGenerator (TestSenderReceiver *sender, int32_t rtpSampleRate = 90000);
-    virtual ~TestLoadGenerator ();
-
-    int32_t SetBitrate (int32_t newBitrateKbps);
-    virtual int32_t Start (const char *threadName = NULL);
-    virtual int32_t Stop ();
-    virtual bool GeneratorLoop () = 0;
-
-protected:
-    virtual int generatePayload ( uint32_t timestamp ) = 0;
-    int generatePayload ();
-    int sendPayload (const uint32_t timeStamp,
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
-
-    webrtc::CriticalSectionWrapper* _critSect;
-    webrtc::EventWrapper *_eventPtr;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _genThread;
-    int32_t _bitrateKbps;
-    TestSenderReceiver *_sender;
-    bool _running;
-    int32_t _rtpSampleRate;
-};
-
-
-class CBRGenerator : public TestLoadGenerator
-{
-public:
-    CBRGenerator (TestSenderReceiver *sender,
-                  size_t payloadSizeBytes,
-                  int32_t bitrateKbps,
-                  int32_t rtpSampleRate = 90000);
-    virtual ~CBRGenerator ();
-
-    virtual int32_t Start () {return (TestLoadGenerator::Start("CBRGenerator"));};
-
-    virtual bool GeneratorLoop ();
-
-protected:
-    virtual int generatePayload ( uint32_t timestamp );
-
-    size_t _payloadSizeBytes;
-    uint8_t *_payload;
-};
-
-
-class CBRFixFRGenerator : public TestLoadGenerator // constant bitrate and fixed frame rate
-{
-public:
-    CBRFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps, int32_t rtpSampleRate = 90000,
-        int32_t frameRateFps = 30, double spread = 0.0);
-    virtual ~CBRFixFRGenerator ();
-
-    virtual int32_t Start () {return (TestLoadGenerator::Start("CBRFixFRGenerator"));};
-
-    virtual bool GeneratorLoop ();
-
-protected:
-    virtual size_t nextPayloadSize ();
-    virtual int generatePayload ( uint32_t timestamp );
-
-    size_t _payloadSizeBytes;
-    uint8_t *_payload;
-    size_t _payloadAllocLen;
-    int32_t _frameRateFps;
-    double      _spreadFactor;
-};
-
-class PeriodicKeyFixFRGenerator : public CBRFixFRGenerator // constant bitrate and fixed frame rate with periodically large frames
-{
-public:
-    PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps, int32_t rtpSampleRate = 90000,
-        int32_t frameRateFps = 30, double spread = 0.0, double keyFactor = 4.0, uint32_t keyPeriod = 300);
-    virtual ~PeriodicKeyFixFRGenerator () {}
-
-protected:
-    virtual size_t nextPayloadSize ();
-
-    double          _keyFactor;
-    uint32_t    _keyPeriod;
-    uint32_t    _frameCount;
-};
-
-// Probably better to inherit CBRFixFRGenerator from CBRVarFRGenerator, but since
-// the fix FR version already existed this was easier.
-class CBRVarFRGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
-{
-public:
-    CBRVarFRGenerator(TestSenderReceiver *sender, int32_t bitrateKbps, const uint8_t* frameRates,
-        uint16_t numFrameRates, int32_t rtpSampleRate = 90000, double avgFrPeriodMs = 5.0,
-        double frSpreadFactor = 0.05, double spreadFactor = 0.0);
-
-    ~CBRVarFRGenerator();
-
-protected:
-    virtual void ChangeFrameRate();
-    virtual size_t nextPayloadSize ();
-
-    double       _avgFrPeriodMs;
-    double       _frSpreadFactor;
-    uint8_t* _frameRates;
-    uint16_t _numFrameRates;
-    int64_t  _frChangeTimeMs;
-};
-
-class CBRFrameDropGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
-{
-public:
-    CBRFrameDropGenerator(TestSenderReceiver *sender, int32_t bitrateKbps,
-                    int32_t rtpSampleRate = 90000, double spreadFactor = 0.0);
-
-    ~CBRFrameDropGenerator();
-
-protected:
-    virtual size_t nextPayloadSize();
-
-    double       _accBits;
-};
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
deleted file mode 100644
index 47f2880638..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
+++ /dev/null
@@ -1,411 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/test/channel_transport/udp_transport.h"
-
-#define NR_OF_SOCKET_BUFFERS 500
-
-
-bool ProcThreadFunction(void *obj)
-{
-    if (obj == NULL)
-    {
-        return false;
-    }
-    TestSenderReceiver *theObj = static_cast<TestSenderReceiver *>(obj);
-
-    return theObj->ProcLoop();
-}
-
-
-TestSenderReceiver::TestSenderReceiver (void)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_running(false),
-_payloadType(0),
-_loadGenerator(NULL),
-_isSender(false),
-_isReceiver(false),
-_sendRecCB(NULL),
-_lastBytesReceived(0),
-_lastTime(-1)
-{
-    // RTP/RTCP module
-    _rtp = RtpRtcp::CreateRtpRtcp(0, false);
-    if (!_rtp)
-    {
-        throw "Could not create RTP/RTCP module";
-        exit(1);
-    }
-
-    if (_rtp->InitReceiver() != 0)
-    {
-        throw "_rtp->InitReceiver()";
-        exit(1);
-    }
-
-    if (_rtp->InitSender() != 0)
-    {
-        throw "_rtp->InitSender()";
-        exit(1);
-    }
-
-    // SocketTransport module
-    uint8_t numberOfThreads = 1;
-    _transport = UdpTransport::Create(0, numberOfThreads);
-    if (!_transport)
-    {
-        throw "Could not create transport module";
-        exit(1);
-    }
-}
-
-TestSenderReceiver::~TestSenderReceiver (void)
-{
-
-    Stop(); // N.B. without critSect
-
-    _critSect->Enter();
-
-    if (_rtp)
-    {
-        RtpRtcp::DestroyRtpRtcp(_rtp);
-        _rtp = NULL;
-    }
-
-    if (_transport)
-    {
-        UdpTransport::Destroy(_transport);
-        _transport = NULL;
-    }
-
-    delete _critSect;
-
-}
-
-
-int32_t TestSenderReceiver::InitReceiver (const uint16_t rtpPort,
-                                          const uint16_t rtcpPort,
-                                          const int8_t payloadType /*= 127*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    // init transport
-    if (_transport->InitializeReceiveSockets(this, rtpPort/*, 0, NULL, 0, true*/) != 0)
-    {
-        throw "_transport->InitializeReceiveSockets";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingRTPCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingRTPCallback";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingDataCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingRTPCallback";
-        exit(1);
-    }
-
-    if (_rtp->SetRTCPStatus(RtcpMode::kReducedSize) != 0) {
-        throw "_rtp->SetRTCPStatus";
-        exit(1);
-    }
-
-    if (_rtp->SetTMMBRStatus(true) != 0)
-    {
-        throw "_rtp->SetTMMBRStatus";
-        exit(1);
-    }
-
-    if (_rtp->RegisterReceivePayload("I420", payloadType, 90000) != 0)
-    {
-        throw "_rtp->RegisterReceivePayload";
-        exit(1);
-    }
-
-    _isReceiver = true;
-
-    return (0);
-}
-
-
-int32_t TestSenderReceiver::Start()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _eventPtr = EventWrapper::Create();
-
-    if (_rtp->SetSendingStatus(true) != 0)
-    {
-        throw "_rtp->SetSendingStatus";
-        exit(1);
-    }
-
-    _procThread = ThreadWrapper::CreateThread(ProcThreadFunction, this,
-                                              "TestSenderReceiver");
-
-    _running = true;
-
-    if (_isReceiver)
-    {
-        if (_transport->StartReceiving(NR_OF_SOCKET_BUFFERS) != 0)
-        {
-            throw "_transport->StartReceiving";
-            exit(1);
-        }
-    }
-
-    _procThread->Start();
-    _procThread->SetPriority(kRealtimePriority);
-
-    return 0;
-
-}
-
-
-int32_t TestSenderReceiver::Stop ()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _transport->StopReceiving();
-
-    if (_procThread)
-    {
-        _running = false;
-        _eventPtr->Set();
-
-        _procThread->Stop();
-        _procThread.reset();
-
-        delete _eventPtr;
-    }
-
-    return (0);
-}
-
-
-bool TestSenderReceiver::ProcLoop(void)
-{
-
-    // process RTP/RTCP module
-    _rtp->Process();
-
-    // process SocketTransport module
-    _transport->Process();
-
-    // no critSect
-    while (_running)
-    {
-        // ask RTP/RTCP module for wait time
-        int32_t rtpWait = _rtp->TimeUntilNextProcess();
-
-        // ask SocketTransport module for wait time
-        int32_t tpWait = _transport->TimeUntilNextProcess();
-
-        int32_t minWait = (rtpWait < tpWait) ? rtpWait: tpWait;
-        minWait = (minWait > 0) ? minWait : 0;
-        // wait
-        _eventPtr->Wait(minWait);
-
-        // process RTP/RTCP module
-        _rtp->Process();
-
-        // process SocketTransport module
-        _transport->Process();
-
-    }
-
-    return true;
-}
-
-
-int32_t TestSenderReceiver::ReceiveBitrateKbps ()
-{
-    size_t bytesSent;
-    uint32_t packetsSent;
-    size_t bytesReceived;
-    uint32_t packetsReceived;
-
-    if (_rtp->DataCountersRTP(&bytesSent, &packetsSent, &bytesReceived, &packetsReceived) == 0)
-    {
-        int64_t now = TickTime::MillisecondTimestamp();
-        int32_t kbps = 0;
-        if (now > _lastTime)
-        {
-            if (_lastTime > 0)
-            {
-                // 8 * bytes / ms = kbps
-                kbps = static_cast<int32_t>(
-                    (8 * (bytesReceived - _lastBytesReceived)) / (now - _lastTime));
-            }
-            _lastTime = now;
-            _lastBytesReceived = bytesReceived;
-        }
-        return (kbps);
-    }
-
-    return (-1);
-}
-
-
-int32_t TestSenderReceiver::SetPacketTimeout(const uint32_t timeoutMS)
-{
-    return (_rtp->SetPacketTimeout(timeoutMS, 0 /* RTCP timeout */));
-}
-
-
-int32_t TestSenderReceiver::OnReceivedPayloadData(const uint8_t* payloadData,
-                                                  const size_t payloadSize,
-                                                  const webrtc::WebRtcRTPHeader* rtpHeader)
-{
-    //printf("OnReceivedPayloadData\n");
-    return (0);
-}
-
-
-void TestSenderReceiver::IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                                           const size_t rtpPacketLength,
-                                           const int8_t* fromIP,
-                                           const uint16_t fromPort)
-{
-    _rtp->IncomingPacket((uint8_t *) incomingRtpPacket, rtpPacketLength);
-}
-
-
-
-void TestSenderReceiver::IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                                            const size_t rtcpPacketLength,
-                                            const int8_t* fromIP,
-                                            const uint16_t fromPort)
-{
-    _rtp->IncomingPacket((uint8_t *) incomingRtcpPacket, rtcpPacketLength);
-}
-
-
-
-
-
-///////////////////
-
-
-int32_t TestSenderReceiver::InitSender (const uint32_t startBitrateKbps,
-                                        const int8_t* ipAddr,
-                                        const uint16_t rtpPort,
-                                        const uint16_t rtcpPort /*= 0*/,
-                                        const int8_t payloadType /*= 127*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _payloadType = payloadType;
-
-    // check load generator valid
-    if (_loadGenerator)
-    {
-        _loadGenerator->SetBitrate(startBitrateKbps);
-    }
-
-    if (_rtp->RegisterSendTransport(_transport) != 0)
-    {
-        throw "_rtp->RegisterSendTransport";
-        exit(1);
-    }
-    if (_rtp->RegisterSendPayload("I420", _payloadType, 90000) != 0)
-    {
-        throw "_rtp->RegisterSendPayload";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingVideoCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingVideoCallback";
-        exit(1);
-    }
-
-    if (_rtp->SetRTCPStatus(RtcpMode::kReducedSize) != 0) {
-        throw "_rtp->SetRTCPStatus";
-        exit(1);
-    }
-
-    if (_rtp->SetSendBitrate(startBitrateKbps*1000, 0, MAX_BITRATE_KBPS) != 0)
-    {
-        throw "_rtp->SetSendBitrate";
-        exit(1);
-    }
-
-
-    // SocketTransport
-    if (_transport->InitializeSendSockets(ipAddr, rtpPort, rtcpPort))
-    {
-        throw "_transport->InitializeSendSockets";
-        exit(1);
-    }
-
-    _isSender = true;
-
-    return (0);
-}
-
-
-
-int32_t
-TestSenderReceiver::SendOutgoingData(const uint32_t timeStamp,
-                                     const uint8_t* payloadData,
-                                     const size_t payloadSize,
-                                     const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
-{
-    return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize));
-}
-
-
-int32_t TestSenderReceiver::SetLoadGenerator(TestLoadGenerator *generator)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _loadGenerator = generator;
-    return(0);
-
-}
-
-void TestSenderReceiver::OnNetworkChanged(const int32_t id,
-                                  const uint32_t minBitrateBps,
-                                  const uint32_t maxBitrateBps,
-                                  const uint8_t fractionLost,
-                                  const uint16_t roundTripTimeMs,
-                                  const uint16_t bwEstimateKbitMin,
-                                  const uint16_t bwEstimateKbitMax)
-{
-    if (_loadGenerator)
-    {
-        _loadGenerator->SetBitrate(maxBitrateBps/1000);
-    }
-
-    if (_sendRecCB)
-    {
-        _sendRecCB->OnOnNetworkChanged(maxBitrateBps,
-            fractionLost,
-            roundTripTimeMs,
-            bwEstimateKbitMin,
-            bwEstimateKbitMax);
-    }
-}
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
deleted file mode 100644
index f394412a73..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
+++ /dev/null
@@ -1,153 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
-#include "webrtc/test/channel_transport/udp_transport.h"
-#include "webrtc/typedefs.h"
-
-class TestLoadGenerator;
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-}
-
-using namespace webrtc;
-
-#define MAX_BITRATE_KBPS 50000
-
-
-class SendRecCB
-{
-public:
-    virtual void OnOnNetworkChanged(const uint32_t bitrateTarget,
-        const uint8_t fractionLost,
-        const uint16_t roundTripTimeMs,
-        const uint16_t bwEstimateKbitMin,
-        const uint16_t bwEstimateKbitMax) = 0;
-
-    virtual ~SendRecCB() {};
-};
-
-
-class TestSenderReceiver : public RtpFeedback, public RtpData, public UdpTransportData, public RtpVideoFeedback
-{
-
-public:
-    TestSenderReceiver (void);
-
-    ~TestSenderReceiver (void);
-
-    void SetCallback (SendRecCB *cb) { _sendRecCB = cb; };
-
-    int32_t Start();
-
-    int32_t Stop();
-
-    bool ProcLoop();
-
-    /////////////////////////////////////////////
-    // Receiver methods
-
-    int32_t InitReceiver (const uint16_t rtpPort,
-                          const uint16_t rtcpPort = 0,
-                          const int8_t payloadType = 127);
-
-    int32_t ReceiveBitrateKbps ();
-
-    int32_t SetPacketTimeout(const uint32_t timeoutMS);
-
-    // Inherited from RtpFeedback
-    int32_t OnInitializeDecoder(const int32_t id,
-                                const int8_t payloadType,
-                                const int8_t payloadName[RTP_PAYLOAD_NAME_SIZE],
-                                const uint32_t frequency,
-                                const uint8_t channels,
-                                const uint32_t rate) override {
-      return 0;
-    }
-
-    void OnIncomingSSRCChanged(const int32_t id, const uint32_t SSRC) override {
-    }
-
-    void OnIncomingCSRCChanged(const int32_t id,
-                               const uint32_t CSRC,
-                               const bool added) override {}
-
-    // Inherited from RtpData
-    int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const webrtc::WebRtcRTPHeader* rtpHeader) override;
-
-    // Inherited from UdpTransportData
-    void IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                           const size_t rtpPacketLength,
-                           const int8_t* fromIP,
-                           const uint16_t fromPort) override;
-
-    void IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                            const size_t rtcpPacketLength,
-                            const int8_t* fromIP,
-                            const uint16_t fromPort) override;
-
-    /////////////////////////////////
-    // Sender methods
-
-    int32_t InitSender (const uint32_t startBitrateKbps,
-                        const int8_t* ipAddr,
-                        const uint16_t rtpPort,
-                        const uint16_t rtcpPort = 0,
-                        const int8_t payloadType = 127);
-
-    int32_t SendOutgoingData(const uint32_t timeStamp,
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
-
-    int32_t SetLoadGenerator(TestLoadGenerator *generator);
-
-    uint32_t BitrateSent() { return (_rtp->BitrateSent()); };
-
-
-    // Inherited from RtpVideoFeedback
-    virtual void OnReceivedIntraFrameRequest(const int32_t id,
-        const uint8_t message = 0) {};
-
-    virtual void OnNetworkChanged(const int32_t id,
-                                  const uint32_t minBitrateBps,
-                                  const uint32_t maxBitrateBps,
-                                  const uint8_t fractionLost,
-                                  const uint16_t roundTripTimeMs,
-                                  const uint16_t bwEstimateKbitMin,
-                                  const uint16_t bwEstimateKbitMax);
-
-private:
-    RtpRtcp* _rtp;
-    UdpTransport* _transport;
-    webrtc::CriticalSectionWrapper* _critSect;
-    webrtc::EventWrapper *_eventPtr;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _procThread;
-    bool _running;
-    int8_t _payloadType;
-    TestLoadGenerator* _loadGenerator;
-    bool _isSender;
-    bool _isReceiver;
-    SendRecCB * _sendRecCB;
-    size_t _lastBytesReceived;
-    int64_t _lastTime;
-
-};
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/test/bwe_standalone.gypi b/webrtc/modules/rtp_rtcp/test/bwe_standalone.gypi
deleted file mode 100644
index e45daec77d..0000000000
--- a/webrtc/modules/rtp_rtcp/test/bwe_standalone.gypi
+++ /dev/null
@@ -1,85 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'bwe_standalone',
-      'type': 'executable',
-      'dependencies': [
-        'matlab_plotting',
-        'rtp_rtcp',
-        'udp_transport',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
-      'sources': [
-        'BWEStandAlone/BWEStandAlone.cc',
-        'BWEStandAlone/TestLoadGenerator.cc',
-        'BWEStandAlone/TestLoadGenerator.h',
-        'BWEStandAlone/TestSenderReceiver.cc',
-        'BWEStandAlone/TestSenderReceiver.h',
-      ], # source
-      'conditions': [
-          ['OS=="linux"', {
-              'cflags': [
-                  '-fexceptions', # enable exceptions
-                  ],
-              },
-           ],
-          ],
-    },
-
-    {
-      'target_name': 'matlab_plotting',
-      'type': 'static_library',
-      'dependencies': [
-        'matlab_plotting_include',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
-      'include_dirs': [
-          '/opt/matlab2010a/extern/include',
-          ],
-      'export_dependent_settings': [
-          'matlab_plotting_include',
-          ],
-      'sources': [
-          'BWEStandAlone/MatlabPlot.cc',
-          'BWEStandAlone/MatlabPlot.h',
-          ],
-      'link_settings': {
-          'ldflags' : [
-              '-L/opt/matlab2010a/bin/glnxa64',
-              '-leng',
-              '-lmx',
-              '-Wl,-rpath,/opt/matlab2010a/bin/glnxa64',
-              ],
-          },
-      'defines': [
-          'MATLAB',
-          ],
-      'conditions': [
-          ['OS=="linux"', {
-              'cflags': [
-                  '-fexceptions', # enable exceptions
-                  ],
-              },
-           ],
-          ],
-      },
-
-    {
-      'target_name': 'matlab_plotting_include',
-      'type': 'none',
-      'direct_dependent_settings': {
-          'include_dirs': [
-              'BWEStandAlone',
-              ],
-          },
-      },
-  ],
-}
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 0270e55802..1d4d6d04a5 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -9,14 +9,14 @@
  */
 
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
-#include "webrtc/test/null_transport.h"
 
 #include <algorithm>
 #include <vector>
 
-using namespace webrtc;
+#include "webrtc/test/null_transport.h"
 
 namespace webrtc {
+
 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
                                       RTPPayloadRegistry* payload_registry,
                                       RtpReceiver* receiver,
@@ -76,7 +76,6 @@ int32_t TestRtpReceiver::OnReceivedPayloadData(
   payload_size_ = payload_size;
   return 0;
 }
-}  // namespace webrtc
 
 class RtpRtcpAPITest : public ::testing::Test {
  protected:
@@ -187,3 +186,5 @@ TEST_F(RtpRtcpAPITest, RtxReceiver) {
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }
+
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index 73334a8b26..d8040f7902 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -7,16 +7,18 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
@@ -68,3 +70,4 @@ class TestRtpReceiver : public NullRtpData {
 };
 
 }  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 745386d485..634969b311 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -15,12 +15,12 @@
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
-using namespace webrtc;
-
+namespace webrtc {
+namespace {
 #define test_rate 64000u
 
 class VerifyingAudioReceiver : public NullRtpData {
@@ -64,7 +64,7 @@ class RTPCallback : public NullRtpFeedback {
   int32_t OnInitializeDecoder(const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
-                              const uint8_t channels,
+                              const size_t channels,
                               const uint32_t rate) override {
     if (payloadType == 96) {
       EXPECT_EQ(test_rate, rate) <<
@@ -165,7 +165,7 @@ TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetStartTimestamp(test_timestamp);
 
   // Test detection at the end of a DTMF tone.
-  //EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
+  // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
 
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
@@ -241,7 +241,7 @@ TEST_F(RtpRtcpAudioTest, RED) {
 
   EXPECT_EQ(0, module1->SetSendREDPayloadType(voice_codec.pltype));
   int8_t red = 0;
-  EXPECT_EQ(0, module1->SendREDPayloadType(red));
+  EXPECT_EQ(0, module1->SendREDPayloadType(&red));
   EXPECT_EQ(voice_codec.pltype, red);
   EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
       voice_codec.plname,
@@ -278,7 +278,7 @@ TEST_F(RtpRtcpAudioTest, RED) {
                                          &fragmentation));
 
   EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
-  EXPECT_EQ(-1, module1->SendREDPayloadType(red));
+  EXPECT_EQ(-1, module1->SendREDPayloadType(&red));
 }
 
 TEST_F(RtpRtcpAudioTest, DTMF) {
@@ -334,7 +334,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
 
   // Send RTP packets for 16 tones a 160 ms  100ms
   // pause between = 2560ms + 1600ms = 4160ms
-  for (;timeStamp <= 250 * 160; timeStamp += 160) {
+  for (; timeStamp <= 250 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
@@ -342,10 +342,13 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
   }
   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
 
-  for (;timeStamp <= 740 * 160; timeStamp += 160) {
+  for (; timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index e9d81122b1..6c60bf1f6d 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -14,13 +14,14 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
-using namespace webrtc;
+namespace webrtc {
+namespace {
 
 const uint64_t kTestPictureId = 12345678;
 const uint8_t kSliPictureId = 156;
@@ -29,30 +30,29 @@ class RtcpCallback : public RtcpIntraFrameObserver {
  public:
   void SetModule(RtpRtcp* module) {
     _rtpRtcpModule = module;
-  };
+  }
   virtual void OnRTCPPacketTimeout(const int32_t id) {
   }
   virtual void OnLipSyncUpdate(const int32_t id,
-                               const int32_t audioVideoOffset) {
-  };
-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
-  };
+                               const int32_t audioVideoOffset) {}
+  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
   virtual void OnReceivedSLI(uint32_t ssrc,
                              uint8_t pictureId) {
     EXPECT_EQ(kSliPictureId & 0x3f, pictureId);
-  };
+  }
   virtual void OnReceivedRPSI(uint32_t ssrc,
                               uint64_t pictureId) {
     EXPECT_EQ(kTestPictureId, pictureId);
-  };
-  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {};
+  }
+  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {}
+
  private:
   RtpRtcp* _rtpRtcpModule;
 };
 
 class TestRtpFeedback : public NullRtpFeedback {
  public:
-  TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
+  explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
   void OnIncomingSSRCChanged(const uint32_t ssrc) override {
@@ -266,3 +266,6 @@ TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
   EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
   EXPECT_EQ(0u, report_blocks[0].fractionLost);
 }
+
+}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index 30a6a1c303..16ea540bd5 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -15,9 +15,9 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
diff --git a/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h b/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
index 2e8d676e47..6c233bba17 100644
--- a/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
+++ b/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
@@ -7,8 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
 
-namespace {
+namespace webrtc {
 
 // Maximum number of media packets allowed in this test. The burst mask types
 // are currently defined up to (kMaxMediaPacketsTest, kMaxMediaPacketsTest).
@@ -185,4 +187,5 @@ const float kMaxResidualLossBurstyMask[kNumberCodes] = {
   0.009657f
 };
 
-}  // namespace
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
diff --git a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
index a8eafdd27e..b164b7e04c 100644
--- a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -22,43 +22,49 @@
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+#include "webrtc/base/random.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
-
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
-//#define VERBOSE_OUTPUT
+// #define VERBOSE_OUTPUT
 
 namespace webrtc {
+namespace fec_private_tables {
+extern const uint8_t** kPacketMaskBurstyTbl[12];
+}
 namespace test {
+using fec_private_tables::kPacketMaskBurstyTbl;
 
 void ReceivePackets(
     ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
     ForwardErrorCorrection::ReceivedPacketList* receivedPacketList,
-    uint32_t numPacketsToDecode, float reorderRate, float duplicateRate) {
+    size_t numPacketsToDecode,
+    float reorderRate,
+    float duplicateRate,
+    Random* random) {
   assert(toDecodeList->empty());
   assert(numPacketsToDecode <= receivedPacketList->size());
 
   ForwardErrorCorrection::ReceivedPacketList::iterator it;
-  for (uint32_t i = 0; i < numPacketsToDecode; i++) {
+  for (size_t i = 0; i < numPacketsToDecode; i++) {
     it = receivedPacketList->begin();
     // Reorder packets.
-    float randomVariable = static_cast<float>(rand()) / RAND_MAX;
+    float randomVariable = random->Rand<float>();
     while (randomVariable < reorderRate) {
       ++it;
       if (it == receivedPacketList->end()) {
         --it;
         break;
       }
-      randomVariable = static_cast<float>(rand()) / RAND_MAX;
+      randomVariable = random->Rand<float>();
     }
     ForwardErrorCorrection::ReceivedPacket* receivedPacket = *it;
     toDecodeList->push_back(receivedPacket);
 
     // Duplicate packets.
-    randomVariable = static_cast<float>(rand()) / RAND_MAX;
+    randomVariable = random->Rand<float>();
     while (randomVariable < duplicateRate) {
       ForwardErrorCorrection::ReceivedPacket* duplicatePacket =
           new ForwardErrorCorrection::ReceivedPacket;
@@ -69,7 +75,7 @@ void ReceivePackets(
       duplicatePacket->pkt->length = receivedPacket->pkt->length;
 
       toDecodeList->push_back(duplicatePacket);
-      randomVariable = static_cast<float>(rand()) / RAND_MAX;
+      randomVariable = random->Rand<float>();
     }
     receivedPacketList->erase(it);
   }
@@ -77,12 +83,8 @@ void ReceivePackets(
 
 TEST(FecTest, FecTest) {
   // TODO(marpan): Split this function into subroutines/helper functions.
-  enum {
-    kMaxNumberMediaPackets = 48
-  };
-  enum {
-    kMaxNumberFecPackets = 48
-  };
+  enum { kMaxNumberMediaPackets = 48 };
+  enum { kMaxNumberFecPackets = 48 };
 
   const uint32_t kNumMaskBytesL0 = 2;
   const uint32_t kNumMaskBytesL1 = 6;
@@ -91,15 +93,12 @@ TEST(FecTest, FecTest) {
   const bool kUseUnequalProtection = true;
 
   // FEC mask types.
-  const FecMaskType kMaskTypes[] = { kFecMaskRandom, kFecMaskBursty };
+  const FecMaskType kMaskTypes[] = {kFecMaskRandom, kFecMaskBursty};
   const int kNumFecMaskTypes = sizeof(kMaskTypes) / sizeof(*kMaskTypes);
 
-  // TODO(pbos): Fix this. Hack to prevent a warning
-  // ('-Wunneeded-internal-declaration') from clang.
-  (void) kPacketMaskBurstyTbl;
-
   // Maximum number of media packets allowed for the mask type.
-  const uint16_t kMaxMediaPackets[] = {kMaxNumberMediaPackets,
+  const uint16_t kMaxMediaPackets[] = {
+      kMaxNumberMediaPackets,
       sizeof(kPacketMaskBurstyTbl) / sizeof(*kPacketMaskBurstyTbl)};
 
   ASSERT_EQ(12, kMaxMediaPackets[1]) << "Max media packets for bursty mode not "
@@ -115,7 +114,7 @@ TEST(FecTest, FecTest) {
 
   ForwardErrorCorrection::Packet* mediaPacket = NULL;
   // Running over only one loss rate to limit execution time.
-  const float lossRate[] = { 0.5f };
+  const float lossRate[] = {0.5f};
   const uint32_t lossRateSize = sizeof(lossRate) / sizeof(*lossRate);
   const float reorderRate = 0.1f;
   const float duplicateRate = 0.1f;
@@ -127,7 +126,7 @@ TEST(FecTest, FecTest) {
   // Seed the random number generator, storing the seed to file in order to
   // reproduce past results.
   const unsigned int randomSeed = static_cast<unsigned int>(time(NULL));
-  srand(randomSeed);
+  Random random(randomSeed);
   std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt";
   FILE* randomSeedFile = fopen(filename.c_str(), "a");
   fprintf(randomSeedFile, "%u\n", randomSeed);
@@ -135,15 +134,13 @@ TEST(FecTest, FecTest) {
   randomSeedFile = NULL;
 
   uint16_t seqNum = 0;
-  uint32_t timeStamp = static_cast<uint32_t>(rand());
-  const uint32_t ssrc = static_cast<uint32_t>(rand());
+  uint32_t timeStamp = random.Rand<uint32_t>();
+  const uint32_t ssrc = random.Rand(1u, 0xfffffffe);
 
   // Loop over the mask types: random and bursty.
   for (int mask_type_idx = 0; mask_type_idx < kNumFecMaskTypes;
        ++mask_type_idx) {
-
     for (uint32_t lossRateIdx = 0; lossRateIdx < lossRateSize; ++lossRateIdx) {
-
       printf("Loss rate: %.2f, Mask type %d \n", lossRate[lossRateIdx],
              mask_type_idx);
 
@@ -159,14 +156,12 @@ TEST(FecTest, FecTest) {
         for (uint32_t numFecPackets = 1;
              numFecPackets <= numMediaPackets && numFecPackets <= packetMaskMax;
              numFecPackets++) {
-
           // Loop over numImpPackets: usually <= (0.3*numMediaPackets).
           // For this test we check up to ~ (numMediaPackets / 4).
           uint32_t maxNumImpPackets = numMediaPackets / 4 + 1;
           for (uint32_t numImpPackets = 0; numImpPackets <= maxNumImpPackets &&
-                                               numImpPackets <= packetMaskMax;
+                                           numImpPackets <= packetMaskMax;
                numImpPackets++) {
-
             uint8_t protectionFactor =
                 static_cast<uint8_t>(numFecPackets * 255 / numMediaPackets);
 
@@ -181,10 +176,11 @@ TEST(FecTest, FecTest) {
                                           mask_table, packetMask);
 
 #ifdef VERBOSE_OUTPUT
-            printf("%u media packets, %u FEC packets, %u numImpPackets, "
-                   "loss rate = %.2f \n",
-                   numMediaPackets, numFecPackets, numImpPackets,
-                   lossRate[lossRateIdx]);
+            printf(
+                "%u media packets, %u FEC packets, %u numImpPackets, "
+                "loss rate = %.2f \n",
+                numMediaPackets, numFecPackets, numImpPackets,
+                lossRate[lossRateIdx]);
             printf("Packet mask matrix \n");
 #endif
 
@@ -232,16 +228,15 @@ TEST(FecTest, FecTest) {
             for (uint32_t i = 0; i < numMediaPackets; ++i) {
               mediaPacket = new ForwardErrorCorrection::Packet;
               mediaPacketList.push_back(mediaPacket);
-              mediaPacket->length = static_cast<size_t>(
-                  (static_cast<float>(rand()) / RAND_MAX) *
-                  (IP_PACKET_SIZE - 12 - 28 -
-                   ForwardErrorCorrection::PacketOverhead()));
-              if (mediaPacket->length < 12) {
-                mediaPacket->length = 12;
-              }
+              const uint32_t kMinPacketSize = 12;
+              const uint32_t kMaxPacketSize = static_cast<uint32_t>(
+                  IP_PACKET_SIZE - 12 - 28 -
+                  ForwardErrorCorrection::PacketOverhead());
+              mediaPacket->length = random.Rand(kMinPacketSize, kMaxPacketSize);
+
               // Generate random values for the first 2 bytes.
-              mediaPacket->data[0] = static_cast<uint8_t>(rand() % 256);
-              mediaPacket->data[1] = static_cast<uint8_t>(rand() % 256);
+              mediaPacket->data[0] = random.Rand<uint8_t>();
+              mediaPacket->data[1] = random.Rand<uint8_t>();
 
               // The first two bits are assumed to be 10 by the
               // FEC encoder. In fact the FEC decoder will set the
@@ -266,7 +261,7 @@ TEST(FecTest, FecTest) {
               ByteWriter<uint32_t>::WriteBigEndian(&mediaPacket->data[8], ssrc);
               // Generate random values for payload
               for (size_t j = 12; j < mediaPacket->length; ++j) {
-                mediaPacket->data[j] = static_cast<uint8_t>(rand() % 256);
+                mediaPacket->data[j] = random.Rand<uint8_t>();
               }
               seqNum++;
             }
@@ -289,8 +284,7 @@ TEST(FecTest, FecTest) {
             while (mediaPacketListItem != mediaPacketList.end()) {
               mediaPacket = *mediaPacketListItem;
               // We want a value between 0 and 1.
-              const float lossRandomVariable =
-                  (static_cast<float>(rand()) / (RAND_MAX));
+              const float lossRandomVariable = random.Rand<float>();
 
               if (lossRandomVariable >= lossRate[lossRateIdx]) {
                 mediaLossMask[mediaPacketIdx] = 1;
@@ -315,8 +309,7 @@ TEST(FecTest, FecTest) {
             uint32_t fecPacketIdx = 0;
             while (fecPacketListItem != fecPacketList.end()) {
               fecPacket = *fecPacketListItem;
-              const float lossRandomVariable =
-                  (static_cast<float>(rand()) / (RAND_MAX));
+              const float lossRandomVariable = random.Rand<float>();
               if (lossRandomVariable >= lossRate[lossRateIdx]) {
                 fecLossMask[fecPacketIdx] = 1;
                 receivedPacket = new ForwardErrorCorrection::ReceivedPacket;
@@ -387,18 +380,15 @@ TEST(FecTest, FecTest) {
             // For error-checking frame completion.
             bool fecPacketReceived = false;
             while (!receivedPacketList.empty()) {
-              uint32_t numPacketsToDecode = static_cast<uint32_t>(
-                  (static_cast<float>(rand()) / RAND_MAX) *
-                      receivedPacketList.size() + 0.5);
-              if (numPacketsToDecode < 1) {
-                numPacketsToDecode = 1;
-              }
+              size_t numPacketsToDecode = random.Rand(
+                  1u, static_cast<uint32_t>(receivedPacketList.size()));
               ReceivePackets(&toDecodeList, &receivedPacketList,
-                             numPacketsToDecode, reorderRate, duplicateRate);
+                             numPacketsToDecode, reorderRate, duplicateRate,
+                             &random);
 
               if (fecPacketReceived == false) {
                 ForwardErrorCorrection::ReceivedPacketList::iterator
-                toDecodeIt = toDecodeList.begin();
+                    toDecodeIt = toDecodeList.begin();
                 while (toDecodeIt != toDecodeList.end()) {
                   receivedPacket = *toDecodeIt;
                   if (receivedPacket->is_fec) {
@@ -418,11 +408,11 @@ TEST(FecTest, FecTest) {
               if (mediaLossMask[mediaPacketIdx] == 1) {
                 // Should have recovered this packet.
                 ForwardErrorCorrection::RecoveredPacketList::iterator
-                recoveredPacketListItem = recoveredPacketList.begin();
+                    recoveredPacketListItem = recoveredPacketList.begin();
 
-                ASSERT_FALSE(
-                    recoveredPacketListItem == recoveredPacketList.end())
-                  << "Insufficient number of recovered packets.";
+                ASSERT_FALSE(recoveredPacketListItem ==
+                             recoveredPacketList.end())
+                    << "Insufficient number of recovered packets.";
                 mediaPacket = *mediaPacketListItem;
                 ForwardErrorCorrection::RecoveredPacket* recoveredPacket =
                     *recoveredPacketListItem;
@@ -462,7 +452,7 @@ TEST(FecTest, FecTest) {
             // Delete received packets we didn't pass to DecodeFEC(), due to
             // early frame completion.
             ForwardErrorCorrection::ReceivedPacketList::iterator
-            receivedPacketIt = receivedPacketList.begin();
+                receivedPacketIt = receivedPacketList.begin();
             while (receivedPacketIt != receivedPacketList.end()) {
               receivedPacket = *receivedPacketIt;
               delete receivedPacket;
@@ -476,11 +466,11 @@ TEST(FecTest, FecTest) {
             }
             timeStamp += 90000 / 30;
           }  // loop over numImpPackets
-        }  // loop over FecPackets
-      }  // loop over numMediaPackets
+        }    // loop over FecPackets
+      }      // loop over numMediaPackets
       delete[] packetMask;
     }  // loop over loss rates
-  }  // loop over mask types
+  }    // loop over mask types
 
   // Have DecodeFEC free allocated memory.
   fec.ResetState(&recoveredPacketList);
diff --git a/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
index 843a7f77f5..466214c740 100644
--- a/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
+++ b/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -59,13 +59,6 @@ enum { kMaxNumberMediaPackets = 48 };
 // Maximum number of media packets allowed for each mask type.
 const uint16_t kMaxMediaPackets[] = {kMaxNumberMediaPackets, 12};
 
-// Maximum number of media packets allowed in this test. The burst mask types
-// are currently defined up to (k=12,m=12).
-const int kMaxMediaPacketsTest = 12;
-
-// Maximum number of FEC codes considered in this test.
-const int kNumberCodes = kMaxMediaPacketsTest * (kMaxMediaPacketsTest + 1) / 2;
-
 // Maximum gap size for characterizing the consecutiveness of the loss.
 const int kMaxGapSize = 2 * kMaxMediaPacketsTest;
 
@@ -407,7 +400,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
     // Loop over all loss configurations for the symbol sequence of length
     // |tot_num_packets|. In this version we process up to (k=12, m=12) codes,
     // and get exact expressions for the residual loss.
-    // TODO (marpan): For larger codes, loop over some random sample of loss
+    // TODO(marpan): For larger codes, loop over some random sample of loss
     // configurations, sampling driven by the underlying statistical loss model
     // (importance sampling).
 
@@ -427,7 +420,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
 
       // Map configuration number to a loss state.
       for (int j = 0; j < tot_num_packets; j++) {
-        state[j]=0;  // Received state.
+        state[j] = 0;  // Received state.
         int bit_value = i >> (tot_num_packets - j - 1) & 1;
         if (bit_value == 1) {
           state[j] = 1;  // Lost state.
@@ -860,9 +853,9 @@ TEST_F(FecPacketMaskMetricsTest, FecXorVsRS) {
         EXPECT_GE(kMetricsXorBursty[code_index].average_residual_loss[k],
                   kMetricsReedSolomon[code_index].average_residual_loss[k]);
        }
-      // TODO (marpan): There are some cases (for high loss rates and/or
-      // burst loss models) where XOR is better than RS. Is there some pattern
-      // we can identify and enforce as a constraint?
+       // TODO(marpan): There are some cases (for high loss rates and/or
+       // burst loss models) where XOR is better than RS. Is there some pattern
+       // we can identify and enforce as a constraint?
     }
   }
 }
@@ -874,7 +867,7 @@ TEST_F(FecPacketMaskMetricsTest, FecXorVsRS) {
 TEST_F(FecPacketMaskMetricsTest, FecTrendXorVsRsLossRate) {
   SetLossModels();
   SetCodeParams();
-  // TODO (marpan): Examine this further to see if the condition can be strictly
+  // TODO(marpan): Examine this further to see if the condition can be strictly
   // satisfied (i.e., scale = 1.0) for all codes with different/better masks.
   double scale = 0.90;
   int num_loss_rates = sizeof(kAverageLossRate) /
@@ -898,7 +891,7 @@ TEST_F(FecPacketMaskMetricsTest, FecTrendXorVsRsLossRate) {
                kMetricsXorRandom[code_index].average_residual_loss[k+1];
           EXPECT_GE(diff_rs_xor_random_loss1, scale * diff_rs_xor_random_loss2);
         }
-        // TODO (marpan): Investigate the cases for the bursty mask where
+        // TODO(marpan): Investigate the cases for the bursty mask where
         // this trend is not strictly satisfied.
       }
     }
@@ -937,7 +930,7 @@ TEST_F(FecPacketMaskMetricsTest, FecBehaviorViaProtectionLevelAndLength) {
             EXPECT_LT(
                 kMetricsReedSolomon[code_index2].average_residual_loss[k],
                 kMetricsReedSolomon[code_index1].average_residual_loss[k]);
-            // TODO (marpan): There are some corner cases where this is not
+            // TODO(marpan): There are some corner cases where this is not
             // satisfied with the current packet masks. Look into updating
             // these cases to see if this behavior should/can be satisfied,
             // with overall lower residual loss for those XOR codes.
@@ -963,7 +956,7 @@ TEST_F(FecPacketMaskMetricsTest, FecVarianceBehaviorXorVsRs) {
   SetCodeParams();
   // The condition is not strictly satisfied with the current masks,
   // i.e., for some codes, the variance of XOR may be slightly higher than RS.
-  // TODO (marpan): Examine this further to see if the condition can be strictly
+  // TODO(marpan): Examine this further to see if the condition can be strictly
   // satisfied (i.e., scale = 1.0) for all codes with different/better masks.
   double scale = 0.95;
   for (int code_index = 0; code_index < max_num_codes_; code_index++) {
@@ -998,7 +991,7 @@ TEST_F(FecPacketMaskMetricsTest, FecXorBurstyPerfectRecoveryConsecutiveLoss) {
 // bursty mask type, for random loss models at low loss rates.
 // The XOR codes with bursty mask types are generally better than the one with
 // random mask type, for bursty loss models and/or high loss rates.
-// TODO (marpan): Enable this test when some of the packet masks are updated.
+// TODO(marpan): Enable this test when some of the packet masks are updated.
 // Some isolated cases of the codes don't pass this currently.
 /*
 TEST_F(FecPacketMaskMetricsTest, FecXorRandomVsBursty) {
diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn
index 163515c466..6704cd6d9a 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -10,12 +10,12 @@ import("../../build/webrtc.gni")
 
 source_set("utility") {
   sources = [
-    "interface/audio_frame_operations.h",
-    "interface/file_player.h",
-    "interface/file_recorder.h",
-    "interface/helpers_android.h",
-    "interface/jvm_android.h",
-    "interface/process_thread.h",
+    "include/audio_frame_operations.h",
+    "include/file_player.h",
+    "include/file_recorder.h",
+    "include/helpers_android.h",
+    "include/jvm_android.h",
+    "include/process_thread.h",
     "source/audio_frame_operations.cc",
     "source/coder.cc",
     "source/coder.h",
diff --git a/webrtc/modules/utility/OWNERS b/webrtc/modules/utility/OWNERS
index 347d278614..65cb70c9b9 100644
--- a/webrtc/modules/utility/OWNERS
+++ b/webrtc/modules/utility/OWNERS
@@ -1,4 +1,9 @@
 asapersson@webrtc.org
 perkj@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/utility/interface/audio_frame_operations.h b/webrtc/modules/utility/include/audio_frame_operations.h
index c2af68ab1b..1551d86894 100644
--- a/webrtc/modules/utility/interface/audio_frame_operations.h
+++ b/webrtc/modules/utility/include/audio_frame_operations.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
-#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
 
 #include "webrtc/typedefs.h"
 
@@ -55,4 +55,4 @@ class AudioFrameOperations {
 
 }  // namespace webrtc
 
-#endif  // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
diff --git a/webrtc/modules/utility/interface/file_player.h b/webrtc/modules/utility/include/file_player.h
index 44f03e475a..4ca134a669 100644
--- a/webrtc/modules/utility/interface/file_player.h
+++ b/webrtc/modules/utility/include/file_player.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
@@ -108,4 +108,4 @@ protected:
 
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/webrtc/modules/utility/interface/file_recorder.h b/webrtc/modules/utility/include/file_recorder.h
index f2ce785368..09ed8ae350 100644
--- a/webrtc/modules/utility/interface/file_recorder.h
+++ b/webrtc/modules/utility/include/file_recorder.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
@@ -81,4 +81,4 @@ protected:
 
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/webrtc/modules/utility/interface/helpers_android.h b/webrtc/modules/utility/include/helpers_android.h
index 5c73fe4566..2840ca965e 100644
--- a/webrtc/modules/utility/interface/helpers_android.h
+++ b/webrtc/modules/utility/include/helpers_android.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
 
 #include <jni.h>
 #include <string>
@@ -84,4 +84,4 @@ class ScopedGlobalRef {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/helpers_ios.h b/webrtc/modules/utility/include/helpers_ios.h
index a5edee0279..a5a07ace17 100644
--- a/webrtc/modules/utility/interface/helpers_ios.h
+++ b/webrtc/modules/utility/include/helpers_ios.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
 
 #if defined(WEBRTC_IOS)
 
@@ -56,4 +56,4 @@ std::string GetDeviceName();
 
 #endif  // defined(WEBRTC_IOS)
 
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
diff --git a/webrtc/modules/utility/interface/jvm_android.h b/webrtc/modules/utility/include/jvm_android.h
index 0744fdbf12..f527dff632 100644
--- a/webrtc/modules/utility/interface/jvm_android.h
+++ b/webrtc/modules/utility/include/jvm_android.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
 
 #include <jni.h>
 #include <string>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 namespace webrtc {
 
@@ -182,4 +182,4 @@ class JVM {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/mock/mock_process_thread.h b/webrtc/modules/utility/include/mock/mock_process_thread.h
index fd108a8354..56d92f4527 100644
--- a/webrtc/modules/utility/interface/mock/mock_process_thread.h
+++ b/webrtc/modules/utility/include/mock/mock_process_thread.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
 
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 
@@ -35,4 +35,4 @@ class MockProcessThread : public ProcessThread {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/interface/process_thread.h b/webrtc/modules/utility/include/process_thread.h
index 451a5a301b..285a5ea587 100644
--- a/webrtc/modules/utility/interface/process_thread.h
+++ b/webrtc/modules/utility/include/process_thread.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
 
 #include "webrtc/typedefs.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -63,4 +63,4 @@ class ProcessThread {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/source/audio_frame_operations.cc b/webrtc/modules/utility/source/audio_frame_operations.cc
index c07ca1fdf6..fe09d7972f 100644
--- a/webrtc/modules/utility/source/audio_frame_operations.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations.cc
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
index c278cdddcd..fff8f4407b 100644
--- a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
@@ -10,8 +10,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index 4ec5f9b4e2..18b690dc67 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/source/coder.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h
index 4270e9b380..abfa87efe1 100644
--- a/webrtc/modules/utility/source/coder.h
+++ b/webrtc/modules/utility/source/coder.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/utility/source/file_player_impl.h b/webrtc/modules/utility/source/file_player_impl.h
index f411db9151..beb6379ff0 100644
--- a/webrtc/modules/utility/source/file_player_impl.h
+++ b/webrtc/modules/utility/source/file_player_impl.h
@@ -14,9 +14,9 @@
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc
index 4b65acdeef..58471e5e8d 100644
--- a/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/webrtc/modules/utility/source/file_player_unittests.cc
@@ -10,7 +10,7 @@
 
 // Unit tests for FilePlayer.
 
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/utility/include/file_player.h"
 
 #include <stdio.h>
 #include <string>
@@ -20,7 +20,6 @@
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 DEFINE_bool(file_player_output, false, "Generate reference files.");
 
@@ -82,7 +81,12 @@ class FilePlayerTest : public ::testing::Test {
   FILE* output_file_;
 };
 
-TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_PlayWavPcmuFile DISABLED_PlayWavPcmuFile
+#else
+#define MAYBE_PlayWavPcmuFile PlayWavPcmuFile
+#endif
+TEST_F(FilePlayerTest, MAYBE_PlayWavPcmuFile) {
   const std::string kFileName =
       test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
   // The file is longer than this, but keeping the output shorter limits the
@@ -93,7 +97,12 @@ TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
   PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
 }
 
-TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcm16File)) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_PlayWavPcm16File DISABLED_PlayWavPcm16File
+#else
+#define MAYBE_PlayWavPcm16File PlayWavPcm16File
+#endif
+TEST_F(FilePlayerTest, MAYBE_PlayWavPcm16File) {
   const std::string kFileName =
       test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
   // The file is longer than this, but keeping the output shorter limits the
diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc
index 13926deb4a..88b20eeac2 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/utility/source/file_recorder_impl.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h
index 8ea96bdad4..697d759375 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.h
+++ b/webrtc/modules/utility/source/file_recorder_impl.h
@@ -17,16 +17,16 @@
 
 #include <list>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
diff --git a/webrtc/modules/utility/source/helpers_android.cc b/webrtc/modules/utility/source/helpers_android.cc
index 25652f237e..aea35f8d5a 100644
--- a/webrtc/modules/utility/source/helpers_android.cc
+++ b/webrtc/modules/utility/source/helpers_android.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 #include <android/log.h>
 #include <assert.h>
diff --git a/webrtc/modules/utility/source/helpers_ios.mm b/webrtc/modules/utility/source/helpers_ios.mm
index 90b7c8f605..2d0ac098c1 100644
--- a/webrtc/modules/utility/source/helpers_ios.mm
+++ b/webrtc/modules/utility/source/helpers_ios.mm
@@ -18,7 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/utility/interface/helpers_ios.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
 
 namespace webrtc {
 namespace ios {
diff --git a/webrtc/modules/utility/source/jvm_android.cc b/webrtc/modules/utility/source/jvm_android.cc
index 648c1685ea..eb37fda040 100644
--- a/webrtc/modules/utility/source/jvm_android.cc
+++ b/webrtc/modules/utility/source/jvm_android.cc
@@ -10,7 +10,7 @@
 
 #include <android/log.h>
 
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 #include "webrtc/base/checks.h"
 
diff --git a/webrtc/modules/utility/source/process_thread_impl.cc b/webrtc/modules/utility/source/process_thread_impl.cc
index 04fa88739f..8cdf01634c 100644
--- a/webrtc/modules/utility/source/process_thread_impl.cc
+++ b/webrtc/modules/utility/source/process_thread_impl.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/utility/source/process_thread_impl.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
@@ -38,8 +38,7 @@ ProcessThread::~ProcessThread() {}
 // static
 rtc::scoped_ptr<ProcessThread> ProcessThread::Create(
     const char* thread_name) {
-  return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl(thread_name))
-      .Pass();
+  return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl(thread_name));
 }
 
 ProcessThreadImpl::ProcessThreadImpl(const char* thread_name)
@@ -76,9 +75,9 @@ void ProcessThreadImpl::Start() {
       m.module->ProcessThreadAttached(this);
   }
 
-  thread_ = ThreadWrapper::CreateThread(&ProcessThreadImpl::Run, this,
-                                        thread_name_);
-  RTC_CHECK(thread_->Start());
+  thread_.reset(
+      new rtc::PlatformThread(&ProcessThreadImpl::Run, this, thread_name_));
+  thread_->Start();
 }
 
 void ProcessThreadImpl::Stop() {
@@ -93,7 +92,7 @@ void ProcessThreadImpl::Stop() {
 
   wake_up_->Set();
 
-  RTC_CHECK(thread_->Stop());
+  thread_->Stop();
   stop_ = false;
 
   // TODO(tommi): Since DeRegisterModule is currently being called from
diff --git a/webrtc/modules/utility/source/process_thread_impl.h b/webrtc/modules/utility/source/process_thread_impl.h
index 4e5861b41e..1c0a0cdfdd 100644
--- a/webrtc/modules/utility/source/process_thread_impl.h
+++ b/webrtc/modules/utility/source/process_thread_impl.h
@@ -15,10 +15,10 @@
 #include <queue>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -70,7 +70,8 @@ class ProcessThreadImpl : public ProcessThread {
 
   rtc::ThreadChecker thread_checker_;
   const rtc::scoped_ptr<EventWrapper> wake_up_;
-  rtc::scoped_ptr<ThreadWrapper> thread_;
+  // TODO(pbos): Remove scoped_ptr and stop recreating the thread.
+  rtc::scoped_ptr<rtc::PlatformThread> thread_;
 
   ModuleList modules_;
   // TODO(tommi): Support delayed tasks.
diff --git a/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
index e080545312..0b35fad7d2 100644
--- a/webrtc/modules/utility/source/process_thread_impl_unittest.cc
+++ b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
@@ -8,9 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <utility>
+
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/utility/source/process_thread_impl.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
@@ -251,8 +253,9 @@ TEST(ProcessThreadImpl, WakeUp) {
   rtc::scoped_ptr<EventWrapper> called(EventWrapper::Create());
 
   MockModule module;
-  int64_t start_time = 0;
-  int64_t called_time = 0;
+  int64_t start_time;
+  int64_t called_time;
+
   // Ask for a callback after 1000ms.
   // TimeUntilNextProcess will be called twice.
   // The first time we use it to get the thread into a waiting state.
@@ -281,8 +284,6 @@ TEST(ProcessThreadImpl, WakeUp) {
   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
   thread.Stop();
 
-  ASSERT_GT(start_time, 0);
-  ASSERT_GT(called_time, 0);
   EXPECT_GE(called_time, start_time);
   uint32_t diff = called_time - start_time;
   // We should have been called back much quicker than 1sec.
@@ -296,7 +297,7 @@ TEST(ProcessThreadImpl, PostTask) {
   rtc::scoped_ptr<EventWrapper> task_ran(EventWrapper::Create());
   rtc::scoped_ptr<RaiseEventTask> task(new RaiseEventTask(task_ran.get()));
   thread.Start();
-  thread.PostTask(task.Pass());
+  thread.PostTask(std::move(task));
   EXPECT_EQ(kEventSignaled, task_ran->Wait(100));
   thread.Stop();
 }
diff --git a/webrtc/modules/utility/utility.gypi b/webrtc/modules/utility/utility.gypi
index 38c9e3ebd9..e5b0a4d9c0 100644
--- a/webrtc/modules/utility/utility.gypi
+++ b/webrtc/modules/utility/utility.gypi
@@ -18,13 +18,13 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'interface/audio_frame_operations.h',
-        'interface/file_player.h',
-        'interface/file_recorder.h',
-        'interface/helpers_android.h',
-        'interface/helpers_ios.h',
-        'interface/jvm_android.h',
-        'interface/process_thread.h',
+        'include/audio_frame_operations.h',
+        'include/file_player.h',
+        'include/file_recorder.h',
+        'include/helpers_android.h',
+        'include/helpers_ios.h',
+        'include/jvm_android.h',
+        'include/process_thread.h',
         'source/audio_frame_operations.cc',
         'source/coder.cc',
         'source/coder.h',
diff --git a/webrtc/modules/video_capture/BUILD.gn b/webrtc/modules/video_capture/BUILD.gn
index b0ed6f4e6c..78f5212950 100644
--- a/webrtc/modules/video_capture/BUILD.gn
+++ b/webrtc/modules/video_capture/BUILD.gn
@@ -16,11 +16,11 @@ source_set("video_capture_module") {
   sources = [
     "device_info_impl.cc",
     "device_info_impl.h",
-    "include/video_capture.h",
-    "include/video_capture_defines.h",
-    "include/video_capture_factory.h",
+    "video_capture.h",
     "video_capture_config.h",
+    "video_capture_defines.h",
     "video_capture_delay.h",
+    "video_capture_factory.h",
     "video_capture_factory.cc",
     "video_capture_impl.cc",
     "video_capture_impl.h",
diff --git a/webrtc/modules/video_capture/device_info_impl.h b/webrtc/modules/video_capture/device_info_impl.h
index 420808bcee..44e7dd596b 100644
--- a/webrtc/modules/video_capture/device_info_impl.h
+++ b/webrtc/modules/video_capture/device_info_impl.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_delay.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
diff --git a/webrtc/modules/video_capture/ios/device_info_ios_objc.h b/webrtc/modules/video_capture/ios/device_info_ios_objc.h
index 65444bedcb..d67b559972 100644
--- a/webrtc/modules/video_capture/ios/device_info_ios_objc.h
+++ b/webrtc/modules/video_capture/ios/device_info_ios_objc.h
@@ -13,7 +13,7 @@
 
 #import <AVFoundation/AVFoundation.h>
 
-#include "webrtc/modules/video_capture/include/video_capture_defines.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
 
 @interface DeviceInfoIosObjC : NSObject
 + (int)captureDeviceCount;
diff --git a/webrtc/modules/video_capture/linux/video_capture_linux.cc b/webrtc/modules/video_capture/linux/video_capture_linux.cc
index fe99c7136f..401a69d8c1 100644
--- a/webrtc/modules/video_capture/linux/video_capture_linux.cc
+++ b/webrtc/modules/video_capture/linux/video_capture_linux.cc
@@ -280,10 +280,10 @@ int32_t VideoCaptureModuleV4L2::StartCapture(
     //start capture thread;
     if (!_captureThread)
     {
-        _captureThread = ThreadWrapper::CreateThread(
-            VideoCaptureModuleV4L2::CaptureThread, this, "CaptureThread");
+        _captureThread.reset(new rtc::PlatformThread(
+            VideoCaptureModuleV4L2::CaptureThread, this, "CaptureThread"));
         _captureThread->Start();
-        _captureThread->SetPriority(kHighPriority);
+        _captureThread->SetPriority(rtc::kHighPriority);
     }
 
     // Needed to start UVC camera - from the uvcview application
diff --git a/webrtc/modules/video_capture/linux/video_capture_linux.h b/webrtc/modules/video_capture/linux/video_capture_linux.h
index 996f8e10ca..8172eb8d2a 100644
--- a/webrtc/modules/video_capture/linux/video_capture_linux.h
+++ b/webrtc/modules/video_capture/linux/video_capture_linux.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
 #define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/video_capture/video_capture_impl.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 namespace webrtc
 {
@@ -39,7 +39,8 @@ private:
     bool AllocateVideoBuffers();
     bool DeAllocateVideoBuffers();
 
-    rtc::scoped_ptr<ThreadWrapper> _captureThread;
+    // TODO(pbos): Stop using scoped_ptr and resetting the thread.
+    rtc::scoped_ptr<rtc::PlatformThread> _captureThread;
     CriticalSectionWrapper* _captureCritSect;
 
     int32_t _deviceId;
diff --git a/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info.mm b/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info.mm
index 1251ecd830..0c0c6a1261 100644
--- a/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info.mm
+++ b/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info.mm
@@ -9,7 +9,7 @@
  */
 
 #import "webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info_objc.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
diff --git a/webrtc/modules/video_capture/test/video_capture_unittest.cc b/webrtc/modules/video_capture/test/video_capture_unittest.cc
index 2b8786b0fe..45d2d2f241 100644
--- a/webrtc/modules/video_capture/test/video_capture_unittest.cc
+++ b/webrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -17,13 +17,12 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_capture/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/video_frame.h"
 
 using rtc::scoped_ptr;
@@ -47,14 +46,14 @@ using webrtc::VideoCaptureModule;
       SleepMs(5); \
       res = (ex); \
     } \
-  } while (0);\
+  } while (0)
 
 #define EXPECT_TRUE_WAIT(ex, timeout) \
   do { \
     bool res; \
     WAIT_(ex, timeout, res); \
     if (!res) EXPECT_TRUE(ex); \
-  } while (0);
+  } while (0)
 
 
 static const int kTimeOut = 5000;
@@ -275,7 +274,14 @@ class VideoCaptureTest : public testing::Test {
   unsigned int number_of_devices_;
 };
 
-TEST_F(VideoCaptureTest, CreateDelete) {
+#ifdef WEBRTC_MAC
+// Currently fails on Mac 64-bit, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5406
+#define MAYBE_CreateDelete DISABLED_CreateDelete
+#else
+#define MAYBE_CreateDelete CreateDelete
+#endif
+TEST_F(VideoCaptureTest, MAYBE_CreateDelete) {
   for (int i = 0; i < 5; ++i) {
     int64_t start_time = TickTime::MillisecondTimestamp();
     TestVideoCaptureCallback capture_observer;
@@ -312,7 +318,14 @@ TEST_F(VideoCaptureTest, CreateDelete) {
   }
 }
 
-TEST_F(VideoCaptureTest, Capabilities) {
+#ifdef WEBRTC_MAC
+// Currently fails on Mac 64-bit, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5406
+#define MAYBE_Capabilities DISABLED_Capabilities
+#else
+#define MAYBE_Capabilities Capabilities
+#endif
+TEST_F(VideoCaptureTest, MAYBE_Capabilities) {
 #ifdef WEBRTC_MAC
   printf("Video capture capabilities are not supported on Mac.\n");
   return;
@@ -479,7 +492,12 @@ TEST_F(VideoCaptureExternalTest, TestExternalCapture) {
 
 // Test frame rate and no picture alarm.
 // Flaky on Win32, see webrtc:3270.
-TEST_F(VideoCaptureExternalTest, DISABLED_ON_WIN(FrameRate)) {
+#if defined(WEBRTC_WIN)
+#define MAYBE_FrameRate DISABLED_FrameRate
+#else
+#define MAYBE_FrameRate FrameRate
+#endif
+TEST_F(VideoCaptureExternalTest, MAYBE_FrameRate) {
   int64_t testTime = 3;
   TickTime startTime = TickTime::Now();
 
diff --git a/webrtc/modules/video_capture/video_capture.gypi b/webrtc/modules/video_capture/video_capture.gypi
index f552df7758..c80f2bf5b5 100644
--- a/webrtc/modules/video_capture/video_capture.gypi
+++ b/webrtc/modules/video_capture/video_capture.gypi
@@ -23,11 +23,11 @@
       'sources': [
         'device_info_impl.cc',
         'device_info_impl.h',
-        'include/video_capture.h',
-        'include/video_capture_defines.h',
-        'include/video_capture_factory.h',
+        'video_capture.h',
         'video_capture_config.h',
+        'video_capture_defines.h',
         'video_capture_delay.h',
+        'video_capture_factory.h',
         'video_capture_factory.cc',
         'video_capture_impl.cc',
         'video_capture_impl.h',
@@ -116,6 +116,23 @@
                 ],
               },
             }],  # win
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-comment',
+                    '-Wno-ignored-attributes',
+                    '-Wno-microsoft-extra-qualification',
+                    '-Wno-missing-braces',
+                    '-Wno-overloaded-virtual',
+                    '-Wno-reorder',
+                    '-Wno-writable-strings',
+                  ],
+                },
+              },
+            }],
             ['OS=="ios"', {
               'sources': [
                 'ios/device_info_ios.h',
diff --git a/webrtc/modules/video_capture/include/video_capture.h b/webrtc/modules/video_capture/video_capture.h
index 09b4502115..08d02211c7 100644
--- a/webrtc/modules/video_capture/include/video_capture.h
+++ b/webrtc/modules/video_capture/video_capture.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
 
 #include "webrtc/common_video/rotation.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/video_capture/include/video_capture_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
 
 namespace webrtc {
 
@@ -157,4 +157,4 @@ protected:
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
+#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
diff --git a/webrtc/modules/video_capture/include/video_capture_defines.h b/webrtc/modules/video_capture/video_capture_defines.h
index 1dee4fa814..ef97ecab9d 100644
--- a/webrtc/modules/video_capture/include/video_capture_defines.h
+++ b/webrtc/modules/video_capture/video_capture_defines.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
@@ -115,4 +115,4 @@ protected:
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
+#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
diff --git a/webrtc/modules/video_capture/video_capture_factory.cc b/webrtc/modules/video_capture/video_capture_factory.cc
index f88f916ba4..618c08bac6 100644
--- a/webrtc/modules/video_capture/video_capture_factory.cc
+++ b/webrtc/modules/video_capture/video_capture_factory.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
 
 #include "webrtc/modules/video_capture/video_capture_impl.h"
 
diff --git a/webrtc/modules/video_capture/include/video_capture_factory.h b/webrtc/modules/video_capture/video_capture_factory.h
index f78437d1a0..4765be1fde 100644
--- a/webrtc/modules/video_capture/include/video_capture_factory.h
+++ b/webrtc/modules/video_capture/video_capture_factory.h
@@ -11,10 +11,10 @@
 // This file contains interfaces used for creating the VideoCaptureModule
 // and DeviceInfo.
 
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_FACTORY_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_FACTORY_H_
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
 
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 
 namespace webrtc {
 
@@ -42,4 +42,4 @@ class VideoCaptureFactory {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_FACTORY_H_
+#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
diff --git a/webrtc/modules/video_capture/video_capture_impl.cc b/webrtc/modules/video_capture/video_capture_impl.cc
index 4046181505..90730cd984 100644
--- a/webrtc/modules/video_capture/video_capture_impl.cc
+++ b/webrtc/modules/video_capture/video_capture_impl.cc
@@ -14,7 +14,7 @@
 
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_capture/video_capture_impl.h b/webrtc/modules/video_capture/video_capture_impl.h
index deb989c251..164421776c 100644
--- a/webrtc/modules/video_capture/video_capture_impl.h
+++ b/webrtc/modules/video_capture/video_capture_impl.h
@@ -17,7 +17,7 @@
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/common_video/rotation.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_capture/windows/sink_filter_ds.h b/webrtc/modules/video_capture/windows/sink_filter_ds.h
index 064cd9d7d3..6be74f69f9 100644
--- a/webrtc/modules/video_capture/windows/sink_filter_ds.h
+++ b/webrtc/modules/video_capture/windows/sink_filter_ds.h
@@ -13,7 +13,7 @@
 
 #include <Streams.h> // Include base DS filter header files
 
-#include "webrtc/modules/video_capture/include/video_capture_defines.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
 
 namespace webrtc
 {
diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn
index 9e8cd47e53..32ac627ed2 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -10,57 +10,57 @@ import("../../build/webrtc.gni")
 
 source_set("video_coding") {
   sources = [
-    "main/interface/video_coding.h",
-    "main/interface/video_coding_defines.h",
-    "main/source/codec_database.cc",
-    "main/source/codec_database.h",
-    "main/source/codec_timer.cc",
-    "main/source/codec_timer.h",
-    "main/source/content_metrics_processing.cc",
-    "main/source/content_metrics_processing.h",
-    "main/source/decoding_state.cc",
-    "main/source/decoding_state.h",
-    "main/source/encoded_frame.cc",
-    "main/source/encoded_frame.h",
-    "main/source/fec_tables_xor.h",
-    "main/source/frame_buffer.cc",
-    "main/source/frame_buffer.h",
-    "main/source/generic_decoder.cc",
-    "main/source/generic_decoder.h",
-    "main/source/generic_encoder.cc",
-    "main/source/generic_encoder.h",
-    "main/source/inter_frame_delay.cc",
-    "main/source/inter_frame_delay.h",
-    "main/source/internal_defines.h",
-    "main/source/jitter_buffer.cc",
-    "main/source/jitter_buffer.h",
-    "main/source/jitter_buffer_common.h",
-    "main/source/jitter_estimator.cc",
-    "main/source/jitter_estimator.h",
-    "main/source/media_opt_util.cc",
-    "main/source/media_opt_util.h",
-    "main/source/media_optimization.cc",
-    "main/source/media_optimization.h",
-    "main/source/nack_fec_tables.h",
-    "main/source/packet.cc",
-    "main/source/packet.h",
-    "main/source/qm_select.cc",
-    "main/source/qm_select.h",
-    "main/source/qm_select_data.h",
-    "main/source/receiver.cc",
-    "main/source/receiver.h",
-    "main/source/rtt_filter.cc",
-    "main/source/rtt_filter.h",
-    "main/source/session_info.cc",
-    "main/source/session_info.h",
-    "main/source/timestamp_map.cc",
-    "main/source/timestamp_map.h",
-    "main/source/timing.cc",
-    "main/source/timing.h",
-    "main/source/video_coding_impl.cc",
-    "main/source/video_coding_impl.h",
-    "main/source/video_receiver.cc",
-    "main/source/video_sender.cc",
+    "codec_database.cc",
+    "codec_database.h",
+    "codec_timer.cc",
+    "codec_timer.h",
+    "content_metrics_processing.cc",
+    "content_metrics_processing.h",
+    "decoding_state.cc",
+    "decoding_state.h",
+    "encoded_frame.cc",
+    "encoded_frame.h",
+    "fec_tables_xor.h",
+    "frame_buffer.cc",
+    "frame_buffer.h",
+    "generic_decoder.cc",
+    "generic_decoder.h",
+    "generic_encoder.cc",
+    "generic_encoder.h",
+    "include/video_coding.h",
+    "include/video_coding_defines.h",
+    "inter_frame_delay.cc",
+    "inter_frame_delay.h",
+    "internal_defines.h",
+    "jitter_buffer.cc",
+    "jitter_buffer.h",
+    "jitter_buffer_common.h",
+    "jitter_estimator.cc",
+    "jitter_estimator.h",
+    "media_opt_util.cc",
+    "media_opt_util.h",
+    "media_optimization.cc",
+    "media_optimization.h",
+    "nack_fec_tables.h",
+    "packet.cc",
+    "packet.h",
+    "qm_select.cc",
+    "qm_select.h",
+    "qm_select_data.h",
+    "receiver.cc",
+    "receiver.h",
+    "rtt_filter.cc",
+    "rtt_filter.h",
+    "session_info.cc",
+    "session_info.h",
+    "timestamp_map.cc",
+    "timestamp_map.h",
+    "timing.cc",
+    "timing.h",
+    "video_coding_impl.cc",
+    "video_coding_impl.h",
+    "video_receiver.cc",
+    "video_sender.cc",
   ]
 
   configs += [ "../..:common_config" ]
@@ -94,14 +94,14 @@ source_set("video_coding") {
 source_set("video_coding_utility") {
   sources = [
     "utility/frame_dropper.cc",
-    "utility/include/frame_dropper.h",
-    "utility/include/moving_average.h",
-    "utility/include/qp_parser.h",
-    "utility/include/quality_scaler.h",
-    "utility/include/vp8_header_parser.h",
+    "utility/frame_dropper.h",
+    "utility/moving_average.h",
     "utility/qp_parser.cc",
+    "utility/qp_parser.h",
     "utility/quality_scaler.cc",
+    "utility/quality_scaler.h",
     "utility/vp8_header_parser.cc",
+    "utility/vp8_header_parser.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -136,6 +136,18 @@ source_set("webrtc_h264") {
   deps = [
     "../../system_wrappers",
   ]
+
+  if (use_third_party_h264) {
+    # Dependency added so that variables use_openh264 and ffmpeg_branding are
+    # recognized build arguments (avoid "Build argument has no effect" error).
+    # The variables and dependencies will be used for real as soon as
+    # https://codereview.webrtc.org/1306813009/ lands. In the meantime, the
+    # build arguments are to be used by waterfall/trybots.
+    deps += [
+      "//third_party/ffmpeg:ffmpeg",
+      "//third_party/openh264:encoder",
+    ]
+  }
 }
 
 # TODO(tkchin): Source set for webrtc_h264_video_toolbox. Currently not
@@ -209,19 +221,15 @@ source_set("webrtc_vp8") {
 }
 
 source_set("webrtc_vp9") {
-  if (rtc_build_vp9) {
-    sources = [
-      "codecs/vp9/include/vp9.h",
-      "codecs/vp9/vp9_frame_buffer_pool.cc",
-      "codecs/vp9/vp9_frame_buffer_pool.h",
-      "codecs/vp9/vp9_impl.cc",
-      "codecs/vp9/vp9_impl.h",
-    ]
-  } else {
-    sources = [
-      "codecs/vp9/vp9_dummy_impl.cc",
-    ]
-  }
+  sources = [
+    "codecs/vp9/include/vp9.h",
+    "codecs/vp9/screenshare_layers.cc",
+    "codecs/vp9/screenshare_layers.h",
+    "codecs/vp9/vp9_frame_buffer_pool.cc",
+    "codecs/vp9/vp9_frame_buffer_pool.h",
+    "codecs/vp9/vp9_impl.cc",
+    "codecs/vp9/vp9_impl.h",
+  ]
 
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
diff --git a/webrtc/modules/video_coding/OWNERS b/webrtc/modules/video_coding/OWNERS
index f452c9ed83..389d632dfd 100644
--- a/webrtc/modules/video_coding/OWNERS
+++ b/webrtc/modules/video_coding/OWNERS
@@ -1,4 +1,9 @@
 stefan@webrtc.org
 marpan@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/video_coding/main/source/codec_database.cc b/webrtc/modules/video_coding/codec_database.cc
index bfdc609e3c..1fae435bab 100644
--- a/webrtc/modules/video_coding/main/source/codec_database.cc
+++ b/webrtc/modules/video_coding/codec_database.cc
@@ -8,26 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/codec_database.h"
+#include "webrtc/modules/video_coding/codec_database.h"
 
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/engine_configurations.h"
-#ifdef VIDEOCODEC_H264
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
-#endif
-#ifdef VIDEOCODEC_I420
 #include "webrtc/modules/video_coding/codecs/i420/include/i420.h"
-#endif
-#ifdef VIDEOCODEC_VP8
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#endif
-#ifdef VIDEOCODEC_VP9
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
-#endif
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
 
 namespace {
 const size_t kDefaultPayloadSize = 1440;
@@ -74,9 +66,9 @@ VideoCodecH264 VideoEncoder::GetDefaultH264Settings() {
   h264_settings.profile = kProfileBase;
   h264_settings.frameDroppingOn = true;
   h264_settings.keyFrameInterval = 3000;
-  h264_settings.spsData = NULL;
+  h264_settings.spsData = nullptr;
   h264_settings.spsLen = 0;
-  h264_settings.ppsData = NULL;
+  h264_settings.ppsData = nullptr;
   h264_settings.ppsLen = 0;
 
   return h264_settings;
@@ -93,12 +85,9 @@ VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
 
 VCMExtDecoderMapItem::VCMExtDecoderMapItem(
     VideoDecoder* external_decoder_instance,
-    uint8_t payload_type,
-    bool internal_render_timing)
+    uint8_t payload_type)
     : payload_type(payload_type),
-      external_decoder_instance(external_decoder_instance),
-      internal_render_timing(internal_render_timing) {
-}
+      external_decoder_instance(external_decoder_instance) {}
 
 VCMCodecDataBase::VCMCodecDataBase(
     VideoEncoderRateObserver* encoder_rate_observer,
@@ -110,35 +99,27 @@ VCMCodecDataBase::VCMCodecDataBase(
       send_codec_(),
       receive_codec_(),
       encoder_payload_type_(0),
-      external_encoder_(NULL),
+      external_encoder_(nullptr),
       internal_source_(false),
       encoder_rate_observer_(encoder_rate_observer),
       encoded_frame_callback_(encoded_frame_callback),
-      ptr_decoder_(NULL),
+      ptr_decoder_(nullptr),
       dec_map_(),
       dec_external_map_() {}
 
 VCMCodecDataBase::~VCMCodecDataBase() {
-  ResetSender();
-  ResetReceiver();
-}
-
-int VCMCodecDataBase::NumberOfCodecs() {
-  return VCM_NUM_VIDEO_CODECS_AVAILABLE;
+  DeleteEncoder();
+  ReleaseDecoder(ptr_decoder_);
+  for (auto& kv : dec_map_)
+    delete kv.second;
+  for (auto& kv : dec_external_map_)
+    delete kv.second;
 }
 
-bool VCMCodecDataBase::Codec(int list_id,
-                             VideoCodec* settings) {
-  if (!settings) {
-    return false;
-  }
-  if (list_id >= VCM_NUM_VIDEO_CODECS_AVAILABLE) {
-    return false;
-  }
+void VCMCodecDataBase::Codec(VideoCodecType codec_type, VideoCodec* settings) {
   memset(settings, 0, sizeof(VideoCodec));
-  switch (list_id) {
-#ifdef VIDEOCODEC_VP8
-    case VCM_VP8_IDX: {
+  switch (codec_type) {
+    case kVideoCodecVP8:
       strncpy(settings->plName, "VP8", 4);
       settings->codecType = kVideoCodecVP8;
       // 96 to 127 dynamic payload types for video codecs.
@@ -152,11 +133,8 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->numberOfSimulcastStreams = 0;
       settings->qpMax = 56;
       settings->codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
-      return true;
-    }
-#endif
-#ifdef VIDEOCODEC_VP9
-    case VCM_VP9_IDX: {
+      return;
+    case kVideoCodecVP9:
       strncpy(settings->plName, "VP9", 4);
       settings->codecType = kVideoCodecVP9;
       // 96 to 127 dynamic payload types for video codecs.
@@ -170,11 +148,8 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->numberOfSimulcastStreams = 0;
       settings->qpMax = 56;
       settings->codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings();
-      return true;
-    }
-#endif
-#ifdef VIDEOCODEC_H264
-    case VCM_H264_IDX: {
+      return;
+    case kVideoCodecH264:
       strncpy(settings->plName, "H264", 5);
       settings->codecType = kVideoCodecH264;
       // 96 to 127 dynamic payload types for video codecs.
@@ -188,11 +163,8 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->numberOfSimulcastStreams = 0;
       settings->qpMax = 56;
       settings->codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
-      return true;
-    }
-#endif
-#ifdef VIDEOCODEC_I420
-    case VCM_I420_IDX: {
+      return;
+    case kVideoCodecI420:
       strncpy(settings->plName, "I420", 5);
       settings->codecType = kVideoCodecI420;
       // 96 to 127 dynamic payload types for video codecs.
@@ -207,32 +179,14 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->height = VCM_DEFAULT_CODEC_HEIGHT;
       settings->minBitrate = VCM_MIN_BITRATE;
       settings->numberOfSimulcastStreams = 0;
-      return true;
-    }
-#endif
-    default: {
-      return false;
-    }
-  }
-}
-
-bool VCMCodecDataBase::Codec(VideoCodecType codec_type,
-                             VideoCodec* settings) {
-  for (int i = 0; i < VCMCodecDataBase::NumberOfCodecs(); i++) {
-    const bool ret = VCMCodecDataBase::Codec(i, settings);
-    if (!ret) {
-      return false;
-    }
-    if (codec_type == settings->codecType) {
-      return true;
-    }
+      return;
+    case kVideoCodecRED:
+    case kVideoCodecULPFEC:
+    case kVideoCodecGeneric:
+    case kVideoCodecUnknown:
+      RTC_NOTREACHED();
+      return;
   }
-  return false;
-}
-
-void VCMCodecDataBase::ResetSender() {
-  DeleteEncoder();
-  periodic_key_frames_ = false;
 }
 
 // Assuming only one registered encoder - since only one used, no need for more.
@@ -264,8 +218,9 @@ bool VCMCodecDataBase::SetSendCodec(const VideoCodec* send_codec,
   if (new_send_codec.maxBitrate == 0) {
     // max is one bit per pixel
     new_send_codec.maxBitrate = (static_cast<int>(send_codec->height) *
-        static_cast<int>(send_codec->width) *
-        static_cast<int>(send_codec->maxFramerate)) / 1000;
+                                 static_cast<int>(send_codec->width) *
+                                 static_cast<int>(send_codec->maxFramerate)) /
+                                1000;
     if (send_codec->startBitrate > new_send_codec.maxBitrate) {
       // But if the user tries to set a higher start bit rate we will
       // increase the max accordingly.
@@ -328,8 +283,8 @@ VideoCodecType VCMCodecDataBase::SendCodec() const {
   return send_codec_.codecType;
 }
 
-bool VCMCodecDataBase::DeregisterExternalEncoder(
-    uint8_t payload_type, bool* was_send_codec) {
+bool VCMCodecDataBase::DeregisterExternalEncoder(uint8_t payload_type,
+                                                 bool* was_send_codec) {
   assert(was_send_codec);
   *was_send_codec = false;
   if (encoder_payload_type_ != payload_type) {
@@ -342,15 +297,14 @@ bool VCMCodecDataBase::DeregisterExternalEncoder(
     *was_send_codec = true;
   }
   encoder_payload_type_ = 0;
-  external_encoder_ = NULL;
+  external_encoder_ = nullptr;
   internal_source_ = false;
   return true;
 }
 
-void VCMCodecDataBase::RegisterExternalEncoder(
-    VideoEncoder* external_encoder,
-    uint8_t payload_type,
-    bool internal_source) {
+void VCMCodecDataBase::RegisterExternalEncoder(VideoEncoder* external_encoder,
+                                               uint8_t payload_type,
+                                               bool internal_source) {
   // Since only one encoder can be used at a given time, only one external
   // encoder can be registered/used.
   external_encoder_ = external_encoder;
@@ -360,9 +314,8 @@ void VCMCodecDataBase::RegisterExternalEncoder(
 }
 
 bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) {
-  if (ptr_encoder_ == NULL) {
+  if (!ptr_encoder_)
     return true;
-  }
 
   // Does not check startBitrate or maxFramerate
   if (new_send_codec.codecType != send_codec_.codecType ||
@@ -419,8 +372,7 @@ bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) {
          ++i) {
       if (memcmp(&new_send_codec.simulcastStream[i],
                  &send_codec_.simulcastStream[i],
-                 sizeof(new_send_codec.simulcastStream[i])) !=
-          0) {
+                 sizeof(new_send_codec.simulcastStream[i])) != 0) {
         return true;
       }
     }
@@ -440,22 +392,6 @@ bool VCMCodecDataBase::SetPeriodicKeyFrames(bool enable) {
   return true;
 }
 
-void VCMCodecDataBase::ResetReceiver() {
-  ReleaseDecoder(ptr_decoder_);
-  ptr_decoder_ = NULL;
-  memset(&receive_codec_, 0, sizeof(VideoCodec));
-  while (!dec_map_.empty()) {
-    DecoderMap::iterator it = dec_map_.begin();
-    delete (*it).second;
-    dec_map_.erase(it);
-  }
-  while (!dec_external_map_.empty()) {
-    ExternalDecoderMap::iterator external_it = dec_external_map_.begin();
-    delete (*external_it).second;
-    dec_external_map_.erase(external_it);
-  }
-}
-
 bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
   ExternalDecoderMap::iterator it = dec_external_map_.find(payload_type);
   if (it == dec_external_map_.end()) {
@@ -465,43 +401,36 @@ bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
   // We can't use payload_type to check if the decoder is currently in use,
   // because payload type may be out of date (e.g. before we decode the first
   // frame after RegisterReceiveCodec)
-  if (ptr_decoder_ != NULL &&
-      &ptr_decoder_->_decoder == (*it).second->external_decoder_instance) {
+  if (ptr_decoder_ != nullptr &&
+      ptr_decoder_->_decoder == (*it).second->external_decoder_instance) {
     // Release it if it was registered and in use.
     ReleaseDecoder(ptr_decoder_);
-    ptr_decoder_ = NULL;
+    ptr_decoder_ = nullptr;
   }
   DeregisterReceiveCodec(payload_type);
-  delete (*it).second;
+  delete it->second;
   dec_external_map_.erase(it);
   return true;
 }
 
 // Add the external encoder object to the list of external decoders.
 // Won't be registered as a receive codec until RegisterReceiveCodec is called.
-bool VCMCodecDataBase::RegisterExternalDecoder(
-    VideoDecoder* external_decoder,
-    uint8_t payload_type,
-    bool internal_render_timing) {
+void VCMCodecDataBase::RegisterExternalDecoder(VideoDecoder* external_decoder,
+                                               uint8_t payload_type) {
   // Check if payload value already exists, if so  - erase old and insert new.
-  VCMExtDecoderMapItem* ext_decoder = new VCMExtDecoderMapItem(
-      external_decoder, payload_type, internal_render_timing);
-  if (!ext_decoder) {
-    return false;
-  }
+  VCMExtDecoderMapItem* ext_decoder =
+      new VCMExtDecoderMapItem(external_decoder, payload_type);
   DeregisterExternalDecoder(payload_type);
   dec_external_map_[payload_type] = ext_decoder;
-  return true;
 }
 
 bool VCMCodecDataBase::DecoderRegistered() const {
   return !dec_map_.empty();
 }
 
-bool VCMCodecDataBase::RegisterReceiveCodec(
-    const VideoCodec* receive_codec,
-    int number_of_cores,
-    bool require_key_frame) {
+bool VCMCodecDataBase::RegisterReceiveCodec(const VideoCodec* receive_codec,
+                                            int number_of_cores,
+                                            bool require_key_frame) {
   if (number_of_cores < 0) {
     return false;
   }
@@ -511,20 +440,17 @@ bool VCMCodecDataBase::RegisterReceiveCodec(
     return false;
   }
   VideoCodec* new_receive_codec = new VideoCodec(*receive_codec);
-  dec_map_[receive_codec->plType] = new VCMDecoderMapItem(new_receive_codec,
-                                                          number_of_cores,
-                                                          require_key_frame);
+  dec_map_[receive_codec->plType] = new VCMDecoderMapItem(
+      new_receive_codec, number_of_cores, require_key_frame);
   return true;
 }
 
-bool VCMCodecDataBase::DeregisterReceiveCodec(
-    uint8_t payload_type) {
+bool VCMCodecDataBase::DeregisterReceiveCodec(uint8_t payload_type) {
   DecoderMap::iterator it = dec_map_.find(payload_type);
   if (it == dec_map_.end()) {
     return false;
   }
-  VCMDecoderMapItem* dec_item = (*it).second;
-  delete dec_item;
+  delete it->second;
   dec_map_.erase(it);
   if (receive_codec_.plType == payload_type) {
     // This codec is currently in use.
@@ -550,49 +476,50 @@ VideoCodecType VCMCodecDataBase::ReceiveCodec() const {
 }
 
 VCMGenericDecoder* VCMCodecDataBase::GetDecoder(
-    uint8_t payload_type, VCMDecodedFrameCallback* decoded_frame_callback) {
+    const VCMEncodedFrame& frame,
+    VCMDecodedFrameCallback* decoded_frame_callback) {
+  uint8_t payload_type = frame.PayloadType();
   if (payload_type == receive_codec_.plType || payload_type == 0) {
     return ptr_decoder_;
   }
   // Check for exisitng decoder, if exists - delete.
   if (ptr_decoder_) {
     ReleaseDecoder(ptr_decoder_);
-    ptr_decoder_ = NULL;
+    ptr_decoder_ = nullptr;
     memset(&receive_codec_, 0, sizeof(VideoCodec));
   }
-  ptr_decoder_ = CreateAndInitDecoder(payload_type, &receive_codec_);
+  ptr_decoder_ = CreateAndInitDecoder(frame, &receive_codec_);
   if (!ptr_decoder_) {
-    return NULL;
+    return nullptr;
   }
   VCMReceiveCallback* callback = decoded_frame_callback->UserReceiveCallback();
-  if (callback) callback->OnIncomingPayloadType(receive_codec_.plType);
-  if (ptr_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback)
-      < 0) {
+  if (callback)
+    callback->OnIncomingPayloadType(receive_codec_.plType);
+  if (ptr_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
+      0) {
     ReleaseDecoder(ptr_decoder_);
-    ptr_decoder_ = NULL;
+    ptr_decoder_ = nullptr;
     memset(&receive_codec_, 0, sizeof(VideoCodec));
-    return NULL;
+    return nullptr;
   }
   return ptr_decoder_;
 }
 
 void VCMCodecDataBase::ReleaseDecoder(VCMGenericDecoder* decoder) const {
   if (decoder) {
-    assert(&decoder->_decoder);
+    assert(decoder->_decoder);
     decoder->Release();
     if (!decoder->External()) {
-      delete &decoder->_decoder;
+      delete decoder->_decoder;
     }
     delete decoder;
   }
 }
 
-bool VCMCodecDataBase::SupportsRenderScheduling() const {
-  const VCMExtDecoderMapItem* ext_item = FindExternalDecoderItem(
-      receive_codec_.plType);
-  if (ext_item == nullptr)
+bool VCMCodecDataBase::PrefersLateDecoding() const {
+  if (!ptr_decoder_)
     return true;
-  return ext_item->internal_render_timing;
+  return ptr_decoder_->PrefersLateDecoding();
 }
 
 bool VCMCodecDataBase::MatchesCurrentResolution(int width, int height) const {
@@ -600,33 +527,43 @@ bool VCMCodecDataBase::MatchesCurrentResolution(int width, int height) const {
 }
 
 VCMGenericDecoder* VCMCodecDataBase::CreateAndInitDecoder(
-    uint8_t payload_type,
+    const VCMEncodedFrame& frame,
     VideoCodec* new_codec) const {
+  uint8_t payload_type = frame.PayloadType();
   assert(new_codec);
   const VCMDecoderMapItem* decoder_item = FindDecoderItem(payload_type);
   if (!decoder_item) {
     LOG(LS_ERROR) << "Can't find a decoder associated with payload type: "
                   << static_cast<int>(payload_type);
-    return NULL;
+    return nullptr;
   }
-  VCMGenericDecoder* ptr_decoder = NULL;
+  VCMGenericDecoder* ptr_decoder = nullptr;
   const VCMExtDecoderMapItem* external_dec_item =
       FindExternalDecoderItem(payload_type);
   if (external_dec_item) {
     // External codec.
     ptr_decoder = new VCMGenericDecoder(
-        *external_dec_item->external_decoder_instance, true);
+        external_dec_item->external_decoder_instance, true);
   } else {
     // Create decoder.
     ptr_decoder = CreateDecoder(decoder_item->settings->codecType);
   }
   if (!ptr_decoder)
-    return NULL;
+    return nullptr;
 
+  // Copy over input resolutions to prevent codec reinitialization due to
+  // the first frame being of a different resolution than the database values.
+  // This is best effort, since there's no guarantee that width/height have been
+  // parsed yet (and may be zero).
+  if (frame.EncodedImage()._encodedWidth > 0 &&
+      frame.EncodedImage()._encodedHeight > 0) {
+    decoder_item->settings->width = frame.EncodedImage()._encodedWidth;
+    decoder_item->settings->height = frame.EncodedImage()._encodedHeight;
+  }
   if (ptr_decoder->InitDecode(decoder_item->settings.get(),
                               decoder_item->number_of_cores) < 0) {
     ReleaseDecoder(ptr_decoder);
-    return NULL;
+    return nullptr;
   }
   memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
   return ptr_decoder;
@@ -641,30 +578,22 @@ void VCMCodecDataBase::DeleteEncoder() {
 
 VCMGenericDecoder* VCMCodecDataBase::CreateDecoder(VideoCodecType type) const {
   switch (type) {
-#ifdef VIDEOCODEC_VP8
     case kVideoCodecVP8:
-      return new VCMGenericDecoder(*(VP8Decoder::Create()));
-#endif
-#ifdef VIDEOCODEC_VP9
+      return new VCMGenericDecoder(VP8Decoder::Create());
     case kVideoCodecVP9:
-      return new VCMGenericDecoder(*(VP9Decoder::Create()));
-#endif
-#ifdef VIDEOCODEC_I420
+      return new VCMGenericDecoder(VP9Decoder::Create());
     case kVideoCodecI420:
-      return new VCMGenericDecoder(*(new I420Decoder));
-#endif
-#ifdef VIDEOCODEC_H264
+      return new VCMGenericDecoder(new I420Decoder());
     case kVideoCodecH264:
       if (H264Decoder::IsSupported()) {
-        return new VCMGenericDecoder(*(H264Decoder::Create()));
+        return new VCMGenericDecoder(H264Decoder::Create());
       }
       break;
-#endif
     default:
       break;
   }
   LOG(LS_WARNING) << "No internal decoder of this type exists.";
-  return NULL;
+  return nullptr;
 }
 
 const VCMDecoderMapItem* VCMCodecDataBase::FindDecoderItem(
@@ -673,7 +602,7 @@ const VCMDecoderMapItem* VCMCodecDataBase::FindDecoderItem(
   if (it != dec_map_.end()) {
     return (*it).second;
   }
-  return NULL;
+  return nullptr;
 }
 
 const VCMExtDecoderMapItem* VCMCodecDataBase::FindExternalDecoderItem(
@@ -682,6 +611,6 @@ const VCMExtDecoderMapItem* VCMCodecDataBase::FindExternalDecoderItem(
   if (it != dec_external_map_.end()) {
     return (*it).second;
   }
-  return NULL;
+  return nullptr;
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/codec_database.h b/webrtc/modules/video_coding/codec_database.h
index 93aa9c3ba8..62ec30a46e 100644
--- a/webrtc/modules/video_coding/main/source/codec_database.h
+++ b/webrtc/modules/video_coding/codec_database.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
 
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/generic_decoder.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/generic_decoder.h"
+#include "webrtc/modules/video_coding/generic_encoder.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -36,12 +36,10 @@ struct VCMDecoderMapItem {
 struct VCMExtDecoderMapItem {
  public:
   VCMExtDecoderMapItem(VideoDecoder* external_decoder_instance,
-                       uint8_t payload_type,
-                       bool internal_render_timing);
+                       uint8_t payload_type);
 
   uint8_t payload_type;
   VideoDecoder* external_decoder_instance;
-  bool internal_render_timing;
 };
 
 class VCMCodecDataBase {
@@ -51,16 +49,8 @@ class VCMCodecDataBase {
   ~VCMCodecDataBase();
 
   // Sender Side
-  // Returns the number of supported codecs (or -1 in case of error).
-  static int NumberOfCodecs();
-
-  // Returns the default settings for the codec with id |list_id|.
-  static bool Codec(int list_id, VideoCodec* settings);
-
   // Returns the default settings for the codec with type |codec_type|.
-  static bool Codec(VideoCodecType codec_type, VideoCodec* settings);
-
-  void ResetSender();
+  static void Codec(VideoCodecType codec_type, VideoCodec* settings);
 
   // Sets the sender side codec and initiates the desired codec given the
   // VideoCodec struct.
@@ -94,19 +84,12 @@ class VCMCodecDataBase {
 
   bool SetPeriodicKeyFrames(bool enable);
 
-  // Receiver Side
-  void ResetReceiver();
-
   // Deregisters an external decoder object specified by |payload_type|.
   bool DeregisterExternalDecoder(uint8_t payload_type);
 
   // Registers an external decoder object to the payload type |payload_type|.
-  // |internal_render_timing| is set to true if the |external_decoder| has
-  // built in rendering which is able to obey the render timestamps of the
-  // encoded frames.
-  bool RegisterExternalDecoder(VideoDecoder* external_decoder,
-                               uint8_t payload_type,
-                               bool internal_render_timing);
+  void RegisterExternalDecoder(VideoDecoder* external_decoder,
+                               uint8_t payload_type);
 
   bool DecoderRegistered() const;
 
@@ -128,16 +111,16 @@ class VCMCodecDataBase {
   // NULL is returned if no encoder with the specified payload type was found
   // and the function failed to create one.
   VCMGenericDecoder* GetDecoder(
-      uint8_t payload_type, VCMDecodedFrameCallback* decoded_frame_callback);
+      const VCMEncodedFrame& frame,
+      VCMDecodedFrameCallback* decoded_frame_callback);
 
   // Deletes the memory of the decoder instance |decoder|. Used to delete
   // deep copies returned by CreateDecoderCopy().
   void ReleaseDecoder(VCMGenericDecoder* decoder) const;
 
-  // Returns true if the currently active decoder supports render scheduling,
-  // that is, it is able to render frames according to the render timestamp of
-  // the encoded frames.
-  bool SupportsRenderScheduling() const;
+  // Returns true if the currently active decoder prefer to decode frames late.
+  // That means that frames must be decoded near the render times stamp.
+  bool PrefersLateDecoding() const;
 
   bool MatchesCurrentResolution(int width, int height) const;
 
@@ -145,7 +128,7 @@ class VCMCodecDataBase {
   typedef std::map<uint8_t, VCMDecoderMapItem*> DecoderMap;
   typedef std::map<uint8_t, VCMExtDecoderMapItem*> ExternalDecoderMap;
 
-  VCMGenericDecoder* CreateAndInitDecoder(uint8_t payload_type,
+  VCMGenericDecoder* CreateAndInitDecoder(const VCMEncodedFrame& frame,
                                           VideoCodec* new_codec) const;
 
   // Determines whether a new codec has to be created or not.
@@ -181,4 +164,4 @@ class VCMCodecDataBase {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
diff --git a/webrtc/modules/video_coding/codec_timer.cc b/webrtc/modules/video_coding/codec_timer.cc
new file mode 100644
index 0000000000..60add8fc4b
--- /dev/null
+++ b/webrtc/modules/video_coding/codec_timer.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/codec_timer.h"
+
+#include <assert.h>
+
+namespace webrtc {
+
+// The first kIgnoredSampleCount samples will be ignored.
+static const int32_t kIgnoredSampleCount = 5;
+
+VCMCodecTimer::VCMCodecTimer()
+    : _filteredMax(0), _ignoredSampleCount(0), _shortMax(0), _history() {
+  Reset();
+}
+
+void VCMCodecTimer::Reset() {
+  _filteredMax = 0;
+  _ignoredSampleCount = 0;
+  _shortMax = 0;
+  for (int i = 0; i < MAX_HISTORY_SIZE; i++) {
+    _history[i].shortMax = 0;
+    _history[i].timeMs = -1;
+  }
+}
+
+// Update the max-value filter
+void VCMCodecTimer::MaxFilter(int32_t decodeTime, int64_t nowMs) {
+  if (_ignoredSampleCount >= kIgnoredSampleCount) {
+    UpdateMaxHistory(decodeTime, nowMs);
+    ProcessHistory(nowMs);
+  } else {
+    _ignoredSampleCount++;
+  }
+}
+
+void VCMCodecTimer::UpdateMaxHistory(int32_t decodeTime, int64_t now) {
+  if (_history[0].timeMs >= 0 && now - _history[0].timeMs < SHORT_FILTER_MS) {
+    if (decodeTime > _shortMax) {
+      _shortMax = decodeTime;
+    }
+  } else {
+    // Only add a new value to the history once a second
+    if (_history[0].timeMs == -1) {
+      // First, no shift
+      _shortMax = decodeTime;
+    } else {
+      // Shift
+      for (int i = (MAX_HISTORY_SIZE - 2); i >= 0; i--) {
+        _history[i + 1].shortMax = _history[i].shortMax;
+        _history[i + 1].timeMs = _history[i].timeMs;
+      }
+    }
+    if (_shortMax == 0) {
+      _shortMax = decodeTime;
+    }
+
+    _history[0].shortMax = _shortMax;
+    _history[0].timeMs = now;
+    _shortMax = 0;
+  }
+}
+
+void VCMCodecTimer::ProcessHistory(int64_t nowMs) {
+  _filteredMax = _shortMax;
+  if (_history[0].timeMs == -1) {
+    return;
+  }
+  for (int i = 0; i < MAX_HISTORY_SIZE; i++) {
+    if (_history[i].timeMs == -1) {
+      break;
+    }
+    if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS) {
+      // This sample (and all samples after this) is too old
+      break;
+    }
+    if (_history[i].shortMax > _filteredMax) {
+      // This sample is the largest one this far into the history
+      _filteredMax = _history[i].shortMax;
+    }
+  }
+}
+
+// Get the maximum observed time within a time window
+int32_t VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const {
+  return _filteredMax;
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codec_timer.h b/webrtc/modules/video_coding/codec_timer.h
new file mode 100644
index 0000000000..8ebd82ab9c
--- /dev/null
+++ b/webrtc/modules/video_coding/codec_timer.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
+#define MAX_HISTORY_SIZE 10
+#define SHORT_FILTER_MS 1000
+
+class VCMShortMaxSample {
+ public:
+  VCMShortMaxSample() : shortMax(0), timeMs(-1) {}
+
+  int32_t shortMax;
+  int64_t timeMs;
+};
+
+class VCMCodecTimer {
+ public:
+  VCMCodecTimer();
+
+  // Updates the max filtered decode time.
+  void MaxFilter(int32_t newDecodeTimeMs, int64_t nowMs);
+
+  // Empty the list of timers.
+  void Reset();
+
+  // Get the required decode time in ms.
+  int32_t RequiredDecodeTimeMs(FrameType frameType) const;
+
+ private:
+  void UpdateMaxHistory(int32_t decodeTime, int64_t now);
+  void ProcessHistory(int64_t nowMs);
+
+  int32_t _filteredMax;
+  // The number of samples ignored so far.
+  int32_t _ignoredSampleCount;
+  int32_t _shortMax;
+  VCMShortMaxSample _history[MAX_HISTORY_SIZE];
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
index 61ef80bbf1..6fee2e6f36 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
@@ -16,7 +16,7 @@
 #include "libyuv/convert.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
 #include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
 #include "webrtc/video_frame.h"
 
@@ -106,8 +106,7 @@ namespace webrtc {
 H264VideoToolboxDecoder::H264VideoToolboxDecoder()
     : callback_(nullptr),
       video_format_(nullptr),
-      decompression_session_(nullptr) {
-}
+      decompression_session_(nullptr) {}
 
 H264VideoToolboxDecoder::~H264VideoToolboxDecoder() {
   DestroyDecompressionSession();
@@ -129,8 +128,7 @@ int H264VideoToolboxDecoder::Decode(
 
   CMSampleBufferRef sample_buffer = nullptr;
   if (!H264AnnexBBufferToCMSampleBuffer(input_image._buffer,
-                                        input_image._length,
-                                        video_format_,
+                                        input_image._length, video_format_,
                                         &sample_buffer)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
@@ -206,11 +204,8 @@ int H264VideoToolboxDecoder::ResetDecompressionSession() {
   int64_t nv12type = kCVPixelFormatType_420YpCbCr8BiPlanarFullRange;
   CFNumberRef pixel_format =
       CFNumberCreate(nullptr, kCFNumberLongType, &nv12type);
-  CFTypeRef values[attributes_size] = {
-    kCFBooleanTrue,
-    io_surface_value,
-    pixel_format
-  };
+  CFTypeRef values[attributes_size] = {kCFBooleanTrue, io_surface_value,
+                                       pixel_format};
   CFDictionaryRef attributes =
       internal::CreateCFDictionary(keys, values, attributes_size);
   if (io_surface_value) {
@@ -266,6 +261,10 @@ void H264VideoToolboxDecoder::SetVideoFormat(
   }
 }
 
+const char* H264VideoToolboxDecoder::ImplementationName() const {
+  return "VideoToolbox";
+}
+
 }  // namespace webrtc
 
 #endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h
index f54ddb9efd..6d64307a82 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h
@@ -45,6 +45,8 @@ class H264VideoToolboxDecoder : public H264Decoder {
 
   int Reset() override;
 
+  const char* ImplementationName() const override;
+
  private:
   int ResetDecompressionSession();
   void ConfigureDecompressionSession();
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
index d677f8b812..7df4ec74ba 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
@@ -99,11 +99,7 @@ struct FrameEncodeParams {
                     int32_t h,
                     int64_t rtms,
                     uint32_t ts)
-      : callback(cb),
-        width(w),
-        height(h),
-        render_time_ms(rtms),
-        timestamp(ts) {
+      : callback(cb), width(w), height(h), render_time_ms(rtms), timestamp(ts) {
     if (csi) {
       codec_specific_info = *csi;
     } else {
@@ -146,9 +142,8 @@ bool CopyVideoFrameToPixelBuffer(const webrtc::VideoFrame& frame,
   int ret = libyuv::I420ToNV12(
       frame.buffer(webrtc::kYPlane), frame.stride(webrtc::kYPlane),
       frame.buffer(webrtc::kUPlane), frame.stride(webrtc::kUPlane),
-      frame.buffer(webrtc::kVPlane), frame.stride(webrtc::kVPlane),
-      dst_y, dst_stride_y, dst_uv, dst_stride_uv,
-      frame.width(), frame.height());
+      frame.buffer(webrtc::kVPlane), frame.stride(webrtc::kVPlane), dst_y,
+      dst_stride_y, dst_uv, dst_stride_uv, frame.width(), frame.height());
   CVPixelBufferUnlockBaseAddress(pixel_buffer, 0);
   if (ret) {
     LOG(LS_ERROR) << "Error converting I420 VideoFrame to NV12 :" << ret;
@@ -188,10 +183,8 @@ void VTCompressionOutputCallback(void* encoder,
   // TODO(tkchin): Allocate buffers through a pool.
   rtc::scoped_ptr<rtc::Buffer> buffer(new rtc::Buffer());
   rtc::scoped_ptr<webrtc::RTPFragmentationHeader> header;
-  if (!H264CMSampleBufferToAnnexBBuffer(sample_buffer,
-                                        is_keyframe,
-                                        buffer.get(),
-                                        header.accept())) {
+  if (!H264CMSampleBufferToAnnexBBuffer(sample_buffer, is_keyframe,
+                                        buffer.get(), header.accept())) {
     return;
   }
   webrtc::EncodedImage frame(buffer->data(), buffer->size(), buffer->size());
@@ -215,8 +208,7 @@ void VTCompressionOutputCallback(void* encoder,
 namespace webrtc {
 
 H264VideoToolboxEncoder::H264VideoToolboxEncoder()
-    : callback_(nullptr), compression_session_(nullptr) {
-}
+    : callback_(nullptr), compression_session_(nullptr) {}
 
 H264VideoToolboxEncoder::~H264VideoToolboxEncoder() {
   DestroyCompressionSession();
@@ -289,8 +281,8 @@ int H264VideoToolboxEncoder::Encode(
       CMTimeMake(input_image.render_time_ms(), 1000);
   CFDictionaryRef frame_properties = nullptr;
   if (is_keyframe_required) {
-    CFTypeRef keys[] = { kVTEncodeFrameOptionKey_ForceKeyFrame };
-    CFTypeRef values[] = { kCFBooleanTrue };
+    CFTypeRef keys[] = {kVTEncodeFrameOptionKey_ForceKeyFrame};
+    CFTypeRef values[] = {kCFBooleanTrue};
     frame_properties = internal::CreateCFDictionary(keys, values, 1);
   }
   rtc::scoped_ptr<internal::FrameEncodeParams> encode_params;
@@ -359,11 +351,8 @@ int H264VideoToolboxEncoder::ResetCompressionSession() {
   int64_t nv12type = kCVPixelFormatType_420YpCbCr8BiPlanarFullRange;
   CFNumberRef pixel_format =
       CFNumberCreate(nullptr, kCFNumberLongType, &nv12type);
-  CFTypeRef values[attributes_size] = {
-    kCFBooleanTrue,
-    io_surface_value,
-    pixel_format
-  };
+  CFTypeRef values[attributes_size] = {kCFBooleanTrue, io_surface_value,
+                                       pixel_format};
   CFDictionaryRef source_attributes =
       internal::CreateCFDictionary(keys, values, attributes_size);
   if (io_surface_value) {
@@ -376,15 +365,11 @@ int H264VideoToolboxEncoder::ResetCompressionSession() {
   }
   OSStatus status = VTCompressionSessionCreate(
       nullptr,  // use default allocator
-      width_,
-      height_,
-      kCMVideoCodecType_H264,
+      width_, height_, kCMVideoCodecType_H264,
       nullptr,  // use default encoder
       source_attributes,
       nullptr,  // use default compressed data allocator
-      internal::VTCompressionOutputCallback,
-      this,
-      &compression_session_);
+      internal::VTCompressionOutputCallback, this, &compression_session_);
   if (source_attributes) {
     CFRelease(source_attributes);
     source_attributes = nullptr;
@@ -434,6 +419,10 @@ void H264VideoToolboxEncoder::DestroyCompressionSession() {
   }
 }
 
+const char* H264VideoToolboxEncoder::ImplementationName() const {
+  return "VideoToolbox";
+}
+
 }  // namespace webrtc
 
 #endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
index f4fb86fa04..269e0411b2 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
@@ -48,6 +48,8 @@ class H264VideoToolboxEncoder : public H264Encoder {
 
   int Release() override;
 
+  const char* ImplementationName() const override;
+
  private:
   int ResetCompressionSession();
   void ConfigureCompressionSession();
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc
index caca96d3d8..322c213f7b 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc
@@ -154,11 +154,10 @@ bool H264CMSampleBufferToAnnexBBuffer(
   return true;
 }
 
-bool H264AnnexBBufferToCMSampleBuffer(
-    const uint8_t* annexb_buffer,
-    size_t annexb_buffer_size,
-    CMVideoFormatDescriptionRef video_format,
-    CMSampleBufferRef* out_sample_buffer) {
+bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
+                                      size_t annexb_buffer_size,
+                                      CMVideoFormatDescriptionRef video_format,
+                                      CMSampleBufferRef* out_sample_buffer) {
   RTC_DCHECK(annexb_buffer);
   RTC_DCHECK(out_sample_buffer);
   *out_sample_buffer = nullptr;
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
index 230dea94a0..31ef525816 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
@@ -9,8 +9,8 @@
  *
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_
 
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 
@@ -19,7 +19,7 @@
 #include <CoreMedia/CoreMedia.h>
 
 #include "webrtc/base/buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -39,11 +39,10 @@ bool H264CMSampleBufferToAnnexBBuffer(
 // If |is_keyframe| is true then |video_format| is ignored since the format will
 // be read from the buffer. Otherwise |video_format| must be provided.
 // Caller is responsible for releasing the created sample buffer.
-bool H264AnnexBBufferToCMSampleBuffer(
-    const uint8_t* annexb_buffer,
-    size_t annexb_buffer_size,
-    CMVideoFormatDescriptionRef video_format,
-    CMSampleBufferRef* out_sample_buffer);
+bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
+                                      size_t annexb_buffer_size,
+                                      CMVideoFormatDescriptionRef video_format,
+                                      CMSampleBufferRef* out_sample_buffer);
 
 // Helper class for reading NALUs from an RTP Annex B buffer.
 class AnnexBBufferReader final {
@@ -97,4 +96,4 @@ class AvccBufferWriter final {
 }  // namespace webrtc
 
 #endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_
diff --git a/webrtc/modules/video_coding/codecs/h264/include/h264.h b/webrtc/modules/video_coding/codecs/h264/include/h264.h
index 3f52839a6c..50ca57c1c9 100644
--- a/webrtc/modules/video_coding/codecs/h264/include/h264.h
+++ b/webrtc/modules/video_coding/codecs/h264/include/h264.h
@@ -23,7 +23,7 @@
 
 #endif  // defined(WEBRTC_IOS) || defined(WEBRTC_MAC)
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_coding/codecs/i420/i420.cc b/webrtc/modules/video_coding/codecs/i420/i420.cc
index cf546a07a1..7f06b4cf7d 100644
--- a/webrtc/modules/video_coding/codecs/i420/i420.cc
+++ b/webrtc/modules/video_coding/codecs/i420/i420.cc
@@ -21,20 +21,19 @@ const size_t kI420HeaderSize = 4;
 
 namespace webrtc {
 
-I420Encoder::I420Encoder() : _inited(false), _encodedImage(),
-    _encodedCompleteCallback(NULL) {
-}
+I420Encoder::I420Encoder()
+    : _inited(false), _encodedImage(), _encodedCompleteCallback(NULL) {}
 
 I420Encoder::~I420Encoder() {
   _inited = false;
-  delete [] _encodedImage._buffer;
+  delete[] _encodedImage._buffer;
 }
 
 int I420Encoder::Release() {
   // Should allocate an encoded frame and then release it here, for that we
   // actually need an init flag.
   if (_encodedImage._buffer != NULL) {
-    delete [] _encodedImage._buffer;
+    delete[] _encodedImage._buffer;
     _encodedImage._buffer = NULL;
   }
   _inited = false;
@@ -53,7 +52,7 @@ int I420Encoder::InitEncode(const VideoCodec* codecSettings,
 
   // Allocating encoded memory.
   if (_encodedImage._buffer != NULL) {
-    delete [] _encodedImage._buffer;
+    delete[] _encodedImage._buffer;
     _encodedImage._buffer = NULL;
     _encodedImage._size = 0;
   }
@@ -101,18 +100,18 @@ int I420Encoder::Encode(const VideoFrame& inputImage,
       kI420HeaderSize;
   if (_encodedImage._size > req_length) {
     // Reallocate buffer.
-    delete [] _encodedImage._buffer;
+    delete[] _encodedImage._buffer;
 
     _encodedImage._buffer = new uint8_t[req_length];
     _encodedImage._size = req_length;
   }
 
-  uint8_t *buffer = _encodedImage._buffer;
+  uint8_t* buffer = _encodedImage._buffer;
 
   buffer = InsertHeader(buffer, width, height);
 
-  int ret_length = ExtractBuffer(inputImage, req_length - kI420HeaderSize,
-                                 buffer);
+  int ret_length =
+      ExtractBuffer(inputImage, req_length - kI420HeaderSize, buffer);
   if (ret_length < 0)
     return WEBRTC_VIDEO_CODEC_MEMORY;
   _encodedImage._length = ret_length + kI420HeaderSize;
@@ -121,7 +120,8 @@ int I420Encoder::Encode(const VideoFrame& inputImage,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-uint8_t* I420Encoder::InsertHeader(uint8_t *buffer, uint16_t width,
+uint8_t* I420Encoder::InsertHeader(uint8_t* buffer,
+                                   uint16_t width,
                                    uint16_t height) {
   *buffer++ = static_cast<uint8_t>(width >> 8);
   *buffer++ = static_cast<uint8_t>(width & 0xFF);
@@ -130,30 +130,29 @@ uint8_t* I420Encoder::InsertHeader(uint8_t *buffer, uint16_t width,
   return buffer;
 }
 
-int
-I420Encoder::RegisterEncodeCompleteCallback(EncodedImageCallback* callback) {
+int I420Encoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
   _encodedCompleteCallback = callback;
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-
-I420Decoder::I420Decoder() : _decodedImage(), _width(0), _height(0),
-    _inited(false), _decodeCompleteCallback(NULL) {
-}
+I420Decoder::I420Decoder()
+    : _decodedImage(),
+      _width(0),
+      _height(0),
+      _inited(false),
+      _decodeCompleteCallback(NULL) {}
 
 I420Decoder::~I420Decoder() {
   Release();
 }
 
-int
-I420Decoder::Reset() {
+int I420Decoder::Reset() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-
-int
-I420Decoder::InitDecode(const VideoCodec* codecSettings,
-                        int /*numberOfCores */) {
+int I420Decoder::InitDecode(const VideoCodec* codecSettings,
+                            int /*numberOfCores */) {
   if (codecSettings == NULL) {
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   } else if (codecSettings->width < 1 || codecSettings->height < 1) {
@@ -165,7 +164,8 @@ I420Decoder::InitDecode(const VideoCodec* codecSettings,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int I420Decoder::Decode(const EncodedImage& inputImage, bool /*missingFrames*/,
+int I420Decoder::Decode(const EncodedImage& inputImage,
+                        bool /*missingFrames*/,
                         const RTPFragmentationHeader* /*fragmentation*/,
                         const CodecSpecificInfo* /*codecSpecificInfo*/,
                         int64_t /*renderTimeMs*/) {
@@ -203,8 +203,8 @@ int I420Decoder::Decode(const EncodedImage& inputImage, bool /*missingFrames*/,
   }
   // Set decoded image parameters.
   int half_width = (_width + 1) / 2;
-  _decodedImage.CreateEmptyFrame(_width, _height,
-                                 _width, half_width, half_width);
+  _decodedImage.CreateEmptyFrame(_width, _height, _width, half_width,
+                                 half_width);
   // Converting from buffer to plane representation.
   int ret = ConvertToI420(kI420, buffer, 0, 0, _width, _height, 0,
                           kVideoRotation_0, &_decodedImage);
@@ -218,7 +218,8 @@ int I420Decoder::Decode(const EncodedImage& inputImage, bool /*missingFrames*/,
 }
 
 const uint8_t* I420Decoder::ExtractHeader(const uint8_t* buffer,
-                                          uint16_t* width, uint16_t* height) {
+                                          uint16_t* width,
+                                          uint16_t* height) {
   *width = static_cast<uint16_t>(*buffer++) << 8;
   *width |= *buffer++;
   *height = static_cast<uint16_t>(*buffer++) << 8;
diff --git a/webrtc/modules/video_coding/codecs/i420/include/i420.h b/webrtc/modules/video_coding/codecs/i420/include/i420.h
index 8990ccf878..9f77845e96 100644
--- a/webrtc/modules/video_coding/codecs/i420/include/i420.h
+++ b/webrtc/modules/video_coding/codecs/i420/include/i420.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_MAIN_INTERFACE_I420_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_MAIN_INTERFACE_I420_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_INCLUDE_I420_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_INCLUDE_I420_H_
 
 #include <vector>
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -24,45 +24,45 @@ class I420Encoder : public VideoEncoder {
 
   virtual ~I420Encoder();
 
-// Initialize the encoder with the information from the VideoCodec.
-//
-// Input:
-//          - codecSettings     : Codec settings.
-//          - numberOfCores     : Number of cores available for the encoder.
-//          - maxPayloadSize    : The maximum size each payload is allowed
-//                                to have. Usually MTU - overhead.
-//
-// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
-//                                <0 - Error
+  // Initialize the encoder with the information from the VideoCodec.
+  //
+  // Input:
+  //          - codecSettings     : Codec settings.
+  //          - numberOfCores     : Number of cores available for the encoder.
+  //          - maxPayloadSize    : The maximum size each payload is allowed
+  //                                to have. Usually MTU - overhead.
+  //
+  // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
+  //                                <0 - Error
   int InitEncode(const VideoCodec* codecSettings,
                  int /*numberOfCores*/,
                  size_t /*maxPayloadSize*/) override;
 
-// "Encode" an I420 image (as a part of a video stream). The encoded image
-// will be returned to the user via the encode complete callback.
-//
-// Input:
-//          - inputImage        : Image to be encoded.
-//          - codecSpecificInfo : Pointer to codec specific data.
-//          - frameType         : Frame type to be sent (Key /Delta).
-//
-// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
-//                                <0 - Error
+  // "Encode" an I420 image (as a part of a video stream). The encoded image
+  // will be returned to the user via the encode complete callback.
+  //
+  // Input:
+  //          - inputImage        : Image to be encoded.
+  //          - codecSpecificInfo : Pointer to codec specific data.
+  //          - frameType         : Frame type to be sent (Key /Delta).
+  //
+  // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
+  //                                <0 - Error
   int Encode(const VideoFrame& inputImage,
              const CodecSpecificInfo* /*codecSpecificInfo*/,
              const std::vector<FrameType>* /*frame_types*/) override;
 
-// Register an encode complete callback object.
-//
-// Input:
-//          - callback         : Callback object which handles encoded images.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+  // Register an encode complete callback object.
+  //
+  // Input:
+  //          - callback         : Callback object which handles encoded images.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
 
-// Free encoder memory.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+  // Free encoder memory.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
   int Release() override;
 
   int SetRates(uint32_t /*newBitRate*/, uint32_t /*frameRate*/) override {
@@ -76,12 +76,13 @@ class I420Encoder : public VideoEncoder {
   void OnDroppedFrame() override {}
 
  private:
-  static uint8_t* InsertHeader(uint8_t* buffer, uint16_t width,
+  static uint8_t* InsertHeader(uint8_t* buffer,
+                               uint16_t width,
                                uint16_t height);
 
-  bool                     _inited;
-  EncodedImage             _encodedImage;
-  EncodedImageCallback*    _encodedCompleteCallback;
+  bool _inited;
+  EncodedImage _encodedImage;
+  EncodedImageCallback* _encodedCompleteCallback;
 };  // class I420Encoder
 
 class I420Decoder : public VideoDecoder {
@@ -90,50 +91,50 @@ class I420Decoder : public VideoDecoder {
 
   virtual ~I420Decoder();
 
-// Initialize the decoder.
-// The user must notify the codec of width and height values.
-//
-// Return value         :  WEBRTC_VIDEO_CODEC_OK.
-//                        <0 - Errors
+  // Initialize the decoder.
+  // The user must notify the codec of width and height values.
+  //
+  // Return value         :  WEBRTC_VIDEO_CODEC_OK.
+  //                        <0 - Errors
   int InitDecode(const VideoCodec* codecSettings,
                  int /*numberOfCores*/) override;
 
-// Decode encoded image (as a part of a video stream). The decoded image
-// will be returned to the user through the decode complete callback.
-//
-// Input:
-//          - inputImage        : Encoded image to be decoded
-//          - missingFrames     : True if one or more frames have been lost
-//                                since the previous decode call.
-//          - codecSpecificInfo : pointer to specific codec data
-//          - renderTimeMs      : Render time in Ms
-//
-// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
-//                                 <0 - Error
+  // Decode encoded image (as a part of a video stream). The decoded image
+  // will be returned to the user through the decode complete callback.
+  //
+  // Input:
+  //          - inputImage        : Encoded image to be decoded
+  //          - missingFrames     : True if one or more frames have been lost
+  //                                since the previous decode call.
+  //          - codecSpecificInfo : pointer to specific codec data
+  //          - renderTimeMs      : Render time in Ms
+  //
+  // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
+  //                                 <0 - Error
   int Decode(const EncodedImage& inputImage,
              bool missingFrames,
              const RTPFragmentationHeader* /*fragmentation*/,
              const CodecSpecificInfo* /*codecSpecificInfo*/,
              int64_t /*renderTimeMs*/) override;
 
-// Register a decode complete callback object.
-//
-// Input:
-//          - callback         : Callback object which handles decoded images.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+  // Register a decode complete callback object.
+  //
+  // Input:
+  //          - callback         : Callback object which handles decoded images.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
   int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
 
-// Free decoder memory.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK.
-//                                  <0 - Error
+  // Free decoder memory.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK.
+  //                                  <0 - Error
   int Release() override;
 
-// Reset decoder state and prepare for a new call.
-//
-// Return value         :  WEBRTC_VIDEO_CODEC_OK.
-//                          <0 - Error
+  // Reset decoder state and prepare for a new call.
+  //
+  // Return value         :  WEBRTC_VIDEO_CODEC_OK.
+  //                          <0 - Error
   int Reset() override;
 
  private:
@@ -142,12 +143,12 @@ class I420Decoder : public VideoDecoder {
                                       uint16_t* height);
 
   VideoFrame _decodedImage;
-  int                         _width;
-  int                         _height;
-  bool                        _inited;
-  DecodedImageCallback*       _decodeCompleteCallback;
+  int _width;
+  int _height;
+  bool _inited;
+  DecodedImageCallback* _decodeCompleteCallback;
 };  // class I420Decoder
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_MAIN_INTERFACE_I420_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_INCLUDE_I420_H_
diff --git a/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h b/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
index 6c926d4794..d727e896ad 100644
--- a/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
+++ b/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
@@ -11,27 +11,32 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
 
+#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
+    "use video_coding/include")
 #include <string>
+#include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class MockEncodedImageCallback : public EncodedImageCallback {
  public:
-  MOCK_METHOD3(Encoded, int32_t(const EncodedImage& encodedImage,
-                                const CodecSpecificInfo* codecSpecificInfo,
-                                const RTPFragmentationHeader* fragmentation));
+  MOCK_METHOD3(Encoded,
+               int32_t(const EncodedImage& encodedImage,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       const RTPFragmentationHeader* fragmentation));
 };
 
 class MockVideoEncoder : public VideoEncoder {
  public:
-  MOCK_CONST_METHOD2(Version, int32_t(int8_t *version, int32_t length));
-  MOCK_METHOD3(InitEncode, int32_t(const VideoCodec* codecSettings,
-                                   int32_t numberOfCores,
-                                   size_t maxPayloadSize));
+  MOCK_CONST_METHOD2(Version, int32_t(int8_t* version, int32_t length));
+  MOCK_METHOD3(InitEncode,
+               int32_t(const VideoCodec* codecSettings,
+                       int32_t numberOfCores,
+                       size_t maxPayloadSize));
   MOCK_METHOD3(Encode,
                int32_t(const VideoFrame& inputImage,
                        const CodecSpecificInfo* codecSpecificInfo,
@@ -47,22 +52,24 @@ class MockVideoEncoder : public VideoEncoder {
 
 class MockDecodedImageCallback : public DecodedImageCallback {
  public:
-  MOCK_METHOD1(Decoded, int32_t(VideoFrame& decodedImage));
+  MOCK_METHOD1(Decoded, int32_t(const VideoFrame& decodedImage));
+  MOCK_METHOD2(Decoded,
+               int32_t(const VideoFrame& decodedImage, int64_t decode_time_ms));
   MOCK_METHOD1(ReceivedDecodedReferenceFrame,
                int32_t(const uint64_t pictureId));
-  MOCK_METHOD1(ReceivedDecodedFrame,
-               int32_t(const uint64_t pictureId));
+  MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
 };
 
 class MockVideoDecoder : public VideoDecoder {
  public:
-  MOCK_METHOD2(InitDecode, int32_t(const VideoCodec* codecSettings,
-                                   int32_t numberOfCores));
-  MOCK_METHOD5(Decode, int32_t(const EncodedImage& inputImage,
-                               bool missingFrames,
-                               const RTPFragmentationHeader* fragmentation,
-                               const CodecSpecificInfo* codecSpecificInfo,
-                               int64_t renderTimeMs));
+  MOCK_METHOD2(InitDecode,
+               int32_t(const VideoCodec* codecSettings, int32_t numberOfCores));
+  MOCK_METHOD5(Decode,
+               int32_t(const EncodedImage& inputImage,
+                       bool missingFrames,
+                       const RTPFragmentationHeader* fragmentation,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       int64_t renderTimeMs));
   MOCK_METHOD1(RegisterDecodeCompleteCallback,
                int32_t(DecodedImageCallback* callback));
   MOCK_METHOD0(Release, int32_t());
diff --git a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
index 6363ab7332..6bcfa909bd 100644
--- a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
+++ b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
@@ -8,23 +8,24 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
 
+#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
+    "use video_coding/include")
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_error_codes.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_decoder.h"
 #include "webrtc/video_encoder.h"
 #include "webrtc/video_frame.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-class RTPFragmentationHeader; // forward declaration
+class RTPFragmentationHeader;  // forward declaration
 
 // Note: if any pointers are added to this struct, it must be fitted
 // with a copy-constructor. See below.
@@ -68,6 +69,10 @@ struct CodecSpecificInfoVP9 {
   uint16_t width[kMaxVp9NumberOfSpatialLayers];
   uint16_t height[kMaxVp9NumberOfSpatialLayers];
   GofInfoVP9 gof;
+
+  // Frame reference data.
+  uint8_t num_ref_pics;
+  uint8_t p_diff[kMaxVp9RefPics];
 };
 
 struct CodecSpecificInfoGeneric {
@@ -86,12 +91,11 @@ union CodecSpecificInfoUnion {
 // Note: if any pointers are added to this struct or its sub-structs, it
 // must be fitted with a copy-constructor. This is because it is copied
 // in the copy-constructor of VCMEncodedFrame.
-struct CodecSpecificInfo
-{
-    VideoCodecType   codecType;
-    CodecSpecificInfoUnion codecSpecific;
+struct CodecSpecificInfo {
+  VideoCodecType codecType;
+  CodecSpecificInfoUnion codecSpecific;
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
diff --git a/webrtc/modules/video_coding/codecs/interface/video_error_codes.h b/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
index 28e5a32d43..ea8829df80 100644
--- a/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
+++ b/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
@@ -8,8 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
+
+#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
+    "use video_coding/include")
 
 // NOTE: in sync with video_coding_module_defines.h
 
@@ -29,4 +32,4 @@
 #define WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE -13
 #define WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT -14
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
diff --git a/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc b/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
index 36ba0e8272..b554b4e9ae 100644
--- a/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
+++ b/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
@@ -57,7 +57,7 @@ int PacketManipulatorImpl::ManipulatePackets(
       active_burst_packets_--;
       nbr_packets_dropped++;
     } else if (RandomUniform() < config_.packet_loss_probability ||
-        packet_loss_has_occurred) {
+               packet_loss_has_occurred) {
       packet_loss_has_occurred = true;
       nbr_packets_dropped++;
       if (config_.packet_loss_mode == kBurst) {
@@ -91,9 +91,9 @@ inline double PacketManipulatorImpl::RandomUniform() {
   // get the same behavior as long as we're using a fixed initial seed.
   critsect_->Enter();
   srand(random_seed_);
-  random_seed_ = rand();
+  random_seed_ = rand();  // NOLINT (rand_r instead of rand)
   critsect_->Leave();
-  return (random_seed_ + 1.0)/(RAND_MAX + 1.0);
+  return (random_seed_ + 1.0) / (RAND_MAX + 1.0);
 }
 
 const char* PacketLossModeToStr(PacketLossMode e) {
@@ -109,4 +109,4 @@ const char* PacketLossModeToStr(PacketLossMode e) {
 }
 
 }  // namespace test
-}  // namespace webrtcc
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/test/packet_manipulator.h b/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
index 16a9dc22ef..3334be072b 100644
--- a/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
+++ b/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
@@ -13,7 +13,7 @@
 
 #include <stdlib.h>
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/test/testsupport/packet_reader.h"
 
@@ -36,10 +36,11 @@ const char* PacketLossModeToStr(PacketLossMode e);
 // scenarios caused by network interference.
 struct NetworkingConfig {
   NetworkingConfig()
-  : packet_size_in_bytes(1500), max_payload_size_in_bytes(1440),
-    packet_loss_mode(kUniform), packet_loss_probability(0.0),
-    packet_loss_burst_length(1) {
-  }
+      : packet_size_in_bytes(1500),
+        max_payload_size_in_bytes(1440),
+        packet_loss_mode(kUniform),
+        packet_loss_probability(0.0),
+        packet_loss_burst_length(1) {}
 
   // Packet size in bytes. Default: 1500 bytes.
   size_t packet_size_in_bytes;
@@ -93,9 +94,11 @@ class PacketManipulatorImpl : public PacketManipulator {
   virtual ~PacketManipulatorImpl();
   int ManipulatePackets(webrtc::EncodedImage* encoded_image) override;
   virtual void InitializeRandomSeed(unsigned int seed);
+
  protected:
   // Returns a uniformly distributed random value between 0.0 and 1.0
   virtual double RandomUniform();
+
  private:
   PacketReader* packet_reader_;
   const NetworkingConfig& config_;
diff --git a/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc b/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
index ace7bc0507..8c3d30dc0d 100644
--- a/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
@@ -13,7 +13,7 @@
 #include <queue>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h"
 #include "webrtc/test/testsupport/unittest_utils.h"
 #include "webrtc/typedefs.h"
@@ -25,7 +25,7 @@ const double kNeverDropProbability = 0.0;
 const double kAlwaysDropProbability = 1.0;
 const int kBurstLength = 1;
 
-class PacketManipulatorTest: public PacketRelatedTest {
+class PacketManipulatorTest : public PacketRelatedTest {
  protected:
   PacketReader packet_reader_;
   EncodedImage image_;
@@ -50,19 +50,15 @@ class PacketManipulatorTest: public PacketRelatedTest {
 
   virtual ~PacketManipulatorTest() {}
 
-  void SetUp() {
-    PacketRelatedTest::SetUp();
-  }
+  void SetUp() { PacketRelatedTest::SetUp(); }
 
-  void TearDown() {
-    PacketRelatedTest::TearDown();
-  }
+  void TearDown() { PacketRelatedTest::TearDown(); }
 
   void VerifyPacketLoss(int expected_nbr_packets_dropped,
                         int actual_nbr_packets_dropped,
                         size_t expected_packet_data_length,
                         uint8_t* expected_packet_data,
-                        EncodedImage& actual_image) {
+                        const EncodedImage& actual_image) {
     EXPECT_EQ(expected_nbr_packets_dropped, actual_nbr_packets_dropped);
     EXPECT_EQ(expected_packet_data_length, image_._length);
     EXPECT_EQ(0, memcmp(expected_packet_data, actual_image._buffer,
@@ -75,10 +71,10 @@ TEST_F(PacketManipulatorTest, Constructor) {
 }
 
 TEST_F(PacketManipulatorTest, DropNone) {
-  PacketManipulatorImpl manipulator(&packet_reader_,  no_drop_config_, false);
+  PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
   int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
-  VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength,
-                   packet_data_, image_);
+  VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength, packet_data_,
+                   image_);
 }
 
 TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
@@ -87,15 +83,14 @@ TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
   PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
   int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
 
-  VerifyPacketLoss(0, nbr_packets_dropped, data_length,
-                     packet_data_, image_);
+  VerifyPacketLoss(0, nbr_packets_dropped, data_length, packet_data_, image_);
 }
 
 TEST_F(PacketManipulatorTest, UniformDropAll) {
   PacketManipulatorImpl manipulator(&packet_reader_, drop_config_, false);
   int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
-  VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped,
-                   0, packet_data_, image_);
+  VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped, 0,
+                   packet_data_, image_);
 }
 
 // Use our customized test class to make the second packet being lost
diff --git a/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc b/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
index c92cfa48a7..9eba205a88 100644
--- a/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
+++ b/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
@@ -19,13 +19,11 @@ namespace webrtc {
 namespace test {
 
 PredictivePacketManipulator::PredictivePacketManipulator(
-    PacketReader* packet_reader, const NetworkingConfig& config)
-    : PacketManipulatorImpl(packet_reader, config, false) {
-}
-
-PredictivePacketManipulator::~PredictivePacketManipulator() {
-}
+    PacketReader* packet_reader,
+    const NetworkingConfig& config)
+    : PacketManipulatorImpl(packet_reader, config, false) {}
 
+PredictivePacketManipulator::~PredictivePacketManipulator() {}
 
 void PredictivePacketManipulator::AddRandomResult(double result) {
   assert(result >= 0.0 && result <= 1.0);
@@ -33,8 +31,9 @@ void PredictivePacketManipulator::AddRandomResult(double result) {
 }
 
 double PredictivePacketManipulator::RandomUniform() {
-  if(random_results_.size() == 0u) {
-    fprintf(stderr, "No more stored results, please make sure AddRandomResult()"
+  if (random_results_.size() == 0u) {
+    fprintf(stderr,
+            "No more stored results, please make sure AddRandomResult()"
             "is called same amount of times you're going to invoke the "
             "RandomUniform() function, i.e. once per packet.\n");
     assert(false);
@@ -45,4 +44,4 @@ double PredictivePacketManipulator::RandomUniform() {
 }
 
 }  // namespace test
-}  // namespace webrtcc
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h b/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
index 082712d870..45c7848c67 100644
--- a/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
+++ b/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
@@ -31,6 +31,7 @@ class PredictivePacketManipulator : public PacketManipulatorImpl {
   // FIFO queue so they will be returned in the same order they were added.
   // Result parameter must be 0.0 to 1.0.
   void AddRandomResult(double result);
+
  protected:
   // Returns a uniformly distributed random value between 0.0 and 1.0
   double RandomUniform() override;
diff --git a/webrtc/modules/video_coding/codecs/test/stats.cc b/webrtc/modules/video_coding/codecs/test/stats.cc
index f87407d223..478b2f4901 100644
--- a/webrtc/modules/video_coding/codecs/test/stats.cc
+++ b/webrtc/modules/video_coding/codecs/test/stats.cc
@@ -39,19 +39,19 @@ Stats::Stats() {}
 Stats::~Stats() {}
 
 bool LessForEncodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.encode_time_in_us < s2.encode_time_in_us;
+  return s1.encode_time_in_us < s2.encode_time_in_us;
 }
 
 bool LessForDecodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.decode_time_in_us < s2.decode_time_in_us;
+  return s1.decode_time_in_us < s2.decode_time_in_us;
 }
 
 bool LessForEncodedSize(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
+  return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
 }
 
 bool LessForBitRate(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
+  return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
 }
 
 FrameStatistic& Stats::NewFrame(int frame_number) {
@@ -78,8 +78,7 @@ void Stats::PrintSummary() {
   size_t nbr_keyframes = 0;
   size_t nbr_nonkeyframes = 0;
 
-  for (FrameStatisticsIterator it = stats_.begin();
-      it != stats_.end(); ++it) {
+  for (FrameStatisticsIterator it = stats_.begin(); it != stats_.end(); ++it) {
     total_encoding_time_in_us += it->encode_time_in_us;
     total_decoding_time_in_us += it->decode_time_in_us;
     total_encoded_frames_lengths += it->encoded_frame_length_in_bytes;
@@ -96,15 +95,13 @@ void Stats::PrintSummary() {
 
   // ENCODING
   printf("Encoding time:\n");
-  frame = std::min_element(stats_.begin(),
-                      stats_.end(), LessForEncodeTime);
-  printf("  Min     : %7d us (frame %d)\n",
-         frame->encode_time_in_us, frame->frame_number);
+  frame = std::min_element(stats_.begin(), stats_.end(), LessForEncodeTime);
+  printf("  Min     : %7d us (frame %d)\n", frame->encode_time_in_us,
+         frame->frame_number);
 
-  frame = std::max_element(stats_.begin(),
-                      stats_.end(), LessForEncodeTime);
-  printf("  Max     : %7d us (frame %d)\n",
-         frame->encode_time_in_us, frame->frame_number);
+  frame = std::max_element(stats_.begin(), stats_.end(), LessForEncodeTime);
+  printf("  Max     : %7d us (frame %d)\n", frame->encode_time_in_us,
+         frame->frame_number);
 
   printf("  Average : %7d us\n",
          static_cast<int>(total_encoding_time_in_us / stats_.size()));
@@ -115,7 +112,7 @@ void Stats::PrintSummary() {
   // failures)
   std::vector<FrameStatistic> decoded_frames;
   for (std::vector<FrameStatistic>::iterator it = stats_.begin();
-      it != stats_.end(); ++it) {
+       it != stats_.end(); ++it) {
     if (it->decoding_successful) {
       decoded_frames.push_back(*it);
     }
@@ -123,15 +120,15 @@ void Stats::PrintSummary() {
   if (decoded_frames.size() == 0) {
     printf("No successfully decoded frames exist in this statistics.\n");
   } else {
-    frame = std::min_element(decoded_frames.begin(),
-                        decoded_frames.end(), LessForDecodeTime);
-    printf("  Min     : %7d us (frame %d)\n",
-           frame->decode_time_in_us, frame->frame_number);
+    frame = std::min_element(decoded_frames.begin(), decoded_frames.end(),
+                             LessForDecodeTime);
+    printf("  Min     : %7d us (frame %d)\n", frame->decode_time_in_us,
+           frame->frame_number);
 
-    frame = std::max_element(decoded_frames.begin(),
-                        decoded_frames.end(), LessForDecodeTime);
-    printf("  Max     : %7d us (frame %d)\n",
-           frame->decode_time_in_us, frame->frame_number);
+    frame = std::max_element(decoded_frames.begin(), decoded_frames.end(),
+                             LessForDecodeTime);
+    printf("  Max     : %7d us (frame %d)\n", frame->decode_time_in_us,
+           frame->frame_number);
 
     printf("  Average : %7d us\n",
            static_cast<int>(total_decoding_time_in_us / decoded_frames.size()));
@@ -141,13 +138,11 @@ void Stats::PrintSummary() {
 
   // SIZE
   printf("Frame sizes:\n");
-  frame = std::min_element(stats_.begin(),
-                      stats_.end(), LessForEncodedSize);
+  frame = std::min_element(stats_.begin(), stats_.end(), LessForEncodedSize);
   printf("  Min     : %7" PRIuS " bytes (frame %d)\n",
          frame->encoded_frame_length_in_bytes, frame->frame_number);
 
-  frame = std::max_element(stats_.begin(),
-                      stats_.end(), LessForEncodedSize);
+  frame = std::max_element(stats_.begin(), stats_.end(), LessForEncodedSize);
   printf("  Max     : %7" PRIuS " bytes (frame %d)\n",
          frame->encoded_frame_length_in_bytes, frame->frame_number);
 
@@ -167,21 +162,17 @@ void Stats::PrintSummary() {
 
   // BIT RATE
   printf("Bit rates:\n");
-  frame = std::min_element(stats_.begin(),
-                      stats_.end(), LessForBitRate);
-  printf("  Min bit rate: %7d kbps (frame %d)\n",
-         frame->bit_rate_in_kbps, frame->frame_number);
+  frame = std::min_element(stats_.begin(), stats_.end(), LessForBitRate);
+  printf("  Min bit rate: %7d kbps (frame %d)\n", frame->bit_rate_in_kbps,
+         frame->frame_number);
 
-  frame = std::max_element(stats_.begin(),
-                      stats_.end(), LessForBitRate);
-  printf("  Max bit rate: %7d kbps (frame %d)\n",
-         frame->bit_rate_in_kbps, frame->frame_number);
+  frame = std::max_element(stats_.begin(), stats_.end(), LessForBitRate);
+  printf("  Max bit rate: %7d kbps (frame %d)\n", frame->bit_rate_in_kbps,
+         frame->frame_number);
 
   printf("\n");
-  printf("Total encoding time  : %7d ms.\n",
-         total_encoding_time_in_us / 1000);
-  printf("Total decoding time  : %7d ms.\n",
-         total_decoding_time_in_us / 1000);
+  printf("Total encoding time  : %7d ms.\n", total_encoding_time_in_us / 1000);
+  printf("Total decoding time  : %7d ms.\n", total_decoding_time_in_us / 1000);
   printf("Total processing time: %7d ms.\n",
          (total_encoding_time_in_us + total_decoding_time_in_us) / 1000);
 }
diff --git a/webrtc/modules/video_coding/codecs/test/stats.h b/webrtc/modules/video_coding/codecs/test/stats.h
index 83ba108bb7..9092631ca1 100644
--- a/webrtc/modules/video_coding/codecs/test/stats.h
+++ b/webrtc/modules/video_coding/codecs/test/stats.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/common_video/interface/video_image.h"
+#include "webrtc/common_video/include/video_image.h"
 
 namespace webrtc {
 namespace test {
diff --git a/webrtc/modules/video_coding/codecs/test/stats_unittest.cc b/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
index a2d27e71d6..0403ccfdb3 100644
--- a/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
@@ -16,21 +16,15 @@
 namespace webrtc {
 namespace test {
 
-class StatsTest: public testing::Test {
+class StatsTest : public testing::Test {
  protected:
-  StatsTest() {
-  }
+  StatsTest() {}
 
-  virtual ~StatsTest() {
-  }
+  virtual ~StatsTest() {}
 
-  void SetUp() {
-    stats_ = new Stats();
-  }
+  void SetUp() { stats_ = new Stats(); }
 
-  void TearDown() {
-    delete stats_;
-  }
+  void TearDown() { delete stats_; }
 
   Stats* stats_;
 };
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
index c814dfe0e7..7376000bd5 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
@@ -93,14 +93,18 @@ bool VideoProcessorImpl::Init() {
   int32_t register_result =
       encoder_->RegisterEncodeCompleteCallback(encode_callback_);
   if (register_result != WEBRTC_VIDEO_CODEC_OK) {
-    fprintf(stderr, "Failed to register encode complete callback, return code: "
-        "%d\n", register_result);
+    fprintf(stderr,
+            "Failed to register encode complete callback, return code: "
+            "%d\n",
+            register_result);
     return false;
   }
   register_result = decoder_->RegisterDecodeCompleteCallback(decode_callback_);
   if (register_result != WEBRTC_VIDEO_CODEC_OK) {
-    fprintf(stderr, "Failed to register decode complete callback, return code: "
-            "%d\n", register_result);
+    fprintf(stderr,
+            "Failed to register decode complete callback, return code: "
+            "%d\n",
+            register_result);
     return false;
   }
   // Init the encoder and decoder
@@ -146,13 +150,14 @@ VideoProcessorImpl::~VideoProcessorImpl() {
   delete decode_callback_;
 }
 
-
 void VideoProcessorImpl::SetRates(int bit_rate, int frame_rate) {
   int set_rates_result = encoder_->SetRates(bit_rate, frame_rate);
   assert(set_rates_result >= 0);
   if (set_rates_result < 0) {
-    fprintf(stderr, "Failed to update encoder with new rate %d, "
-            "return code: %d\n", bit_rate, set_rates_result);
+    fprintf(stderr,
+            "Failed to update encoder with new rate %d, "
+            "return code: %d\n",
+            bit_rate, set_rates_result);
   }
   num_dropped_frames_ = 0;
   num_spatial_resizes_ = 0;
@@ -175,7 +180,7 @@ int VideoProcessorImpl::NumberSpatialResizes() {
 }
 
 bool VideoProcessorImpl::ProcessFrame(int frame_number) {
-  assert(frame_number >=0);
+  assert(frame_number >= 0);
   if (!initialized_) {
     fprintf(stderr, "Attempting to use uninitialized VideoProcessor!\n");
     return false;
@@ -186,10 +191,8 @@ bool VideoProcessorImpl::ProcessFrame(int frame_number) {
   }
   if (frame_reader_->ReadFrame(source_buffer_)) {
     // Copy the source frame to the newly read frame data.
-    source_frame_.CreateFrame(source_buffer_,
-                              config_.codec_settings->width,
-                              config_.codec_settings->height,
-                              kVideoRotation_0);
+    source_frame_.CreateFrame(source_buffer_, config_.codec_settings->width,
+                              config_.codec_settings->height, kVideoRotation_0);
 
     // Ensure we have a new statistics data object we can fill:
     FrameStatistic& stat = stats_->NewFrame(frame_number);
@@ -224,10 +227,10 @@ bool VideoProcessorImpl::ProcessFrame(int frame_number) {
 
 void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
   // Timestamp is frame number, so this gives us #dropped frames.
-  int num_dropped_from_prev_encode =  encoded_image._timeStamp -
-      prev_time_stamp_ - 1;
-  num_dropped_frames_ +=  num_dropped_from_prev_encode;
-  prev_time_stamp_ =  encoded_image._timeStamp;
+  int num_dropped_from_prev_encode =
+      encoded_image._timeStamp - prev_time_stamp_ - 1;
+  num_dropped_frames_ += num_dropped_from_prev_encode;
+  prev_time_stamp_ = encoded_image._timeStamp;
   if (num_dropped_from_prev_encode > 0) {
     // For dropped frames, we write out the last decoded frame to avoid getting
     // out of sync for the computation of PSNR and SSIM.
@@ -244,15 +247,16 @@ void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
   TickTime encode_stop = TickTime::Now();
   int frame_number = encoded_image._timeStamp;
   FrameStatistic& stat = stats_->stats_[frame_number];
-  stat.encode_time_in_us = GetElapsedTimeMicroseconds(encode_start_,
-                                                      encode_stop);
+  stat.encode_time_in_us =
+      GetElapsedTimeMicroseconds(encode_start_, encode_stop);
   stat.encoding_successful = true;
   stat.encoded_frame_length_in_bytes = encoded_image._length;
   stat.frame_number = encoded_image._timeStamp;
   stat.frame_type = encoded_image._frameType;
   stat.bit_rate_in_kbps = encoded_image._length * bit_rate_factor_;
-  stat.total_packets = encoded_image._length /
-      config_.networking_config.packet_size_in_bytes + 1;
+  stat.total_packets =
+      encoded_image._length / config_.networking_config.packet_size_in_bytes +
+      1;
 
   // Perform packet loss if criteria is fullfilled:
   bool exclude_this_frame = false;
@@ -280,7 +284,7 @@ void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
   copied_image._buffer = copied_buffer.get();
   if (!exclude_this_frame) {
     stat.packets_dropped =
-          packet_manipulator_->ManipulatePackets(&copied_image);
+        packet_manipulator_->ManipulatePackets(&copied_image);
   }
 
   // Keep track of if frames are lost due to packet loss so we can tell
@@ -305,26 +309,25 @@ void VideoProcessorImpl::FrameDecoded(const VideoFrame& image) {
   int frame_number = image.timestamp();
   // Report stats
   FrameStatistic& stat = stats_->stats_[frame_number];
-  stat.decode_time_in_us = GetElapsedTimeMicroseconds(decode_start_,
-                                                      decode_stop);
+  stat.decode_time_in_us =
+      GetElapsedTimeMicroseconds(decode_start_, decode_stop);
   stat.decoding_successful = true;
 
   // Check for resize action (either down or up):
   if (static_cast<int>(image.width()) != last_encoder_frame_width_ ||
-      static_cast<int>(image.height()) != last_encoder_frame_height_ ) {
+      static_cast<int>(image.height()) != last_encoder_frame_height_) {
     ++num_spatial_resizes_;
     last_encoder_frame_width_ = image.width();
     last_encoder_frame_height_ = image.height();
   }
   // Check if codec size is different from native/original size, and if so,
   // upsample back to original size: needed for PSNR and SSIM computations.
-  if (image.width() !=  config_.codec_settings->width ||
+  if (image.width() != config_.codec_settings->width ||
       image.height() != config_.codec_settings->height) {
     VideoFrame up_image;
-    int ret_val = scaler_.Set(image.width(), image.height(),
-                              config_.codec_settings->width,
-                              config_.codec_settings->height,
-                              kI420, kI420, kScaleBilinear);
+    int ret_val = scaler_.Set(
+        image.width(), image.height(), config_.codec_settings->width,
+        config_.codec_settings->height, kI420, kI420, kScaleBilinear);
     assert(ret_val >= 0);
     if (ret_val < 0) {
       fprintf(stderr, "Failed to set scalar for frame: %d, return code: %d\n",
@@ -366,7 +369,8 @@ void VideoProcessorImpl::FrameDecoded(const VideoFrame& image) {
 }
 
 int VideoProcessorImpl::GetElapsedTimeMicroseconds(
-    const webrtc::TickTime& start, const webrtc::TickTime& stop) {
+    const webrtc::TickTime& start,
+    const webrtc::TickTime& stop) {
   uint64_t encode_time = (stop - start).Microseconds();
   assert(encode_time <
          static_cast<unsigned int>(std::numeric_limits<int>::max()));
@@ -404,8 +408,7 @@ const char* VideoCodecTypeToStr(webrtc::VideoCodecType e) {
 }
 
 // Callbacks
-int32_t
-VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
+int32_t VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
     const EncodedImage& encoded_image,
     const webrtc::CodecSpecificInfo* codec_specific_info,
     const webrtc::RTPFragmentationHeader* fragmentation) {
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor.h b/webrtc/modules/video_coding/codecs/test/videoprocessor.h
index 0b094ae73e..3ee08fd46a 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor.h
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor.h
@@ -13,9 +13,10 @@
 
 #include <string>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/scaler.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/stats.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
@@ -242,12 +243,16 @@ class VideoProcessorImpl : public VideoProcessor {
 
   // Callback class required to implement according to the VideoDecoder API.
   class VideoProcessorDecodeCompleteCallback
-    : public webrtc::DecodedImageCallback {
+      : public webrtc::DecodedImageCallback {
    public:
-      explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
-      : video_processor_(vp) {
+    explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
+        : video_processor_(vp) {}
+    int32_t Decoded(webrtc::VideoFrame& image) override;
+    int32_t Decoded(webrtc::VideoFrame& image,
+                    int64_t decode_time_ms) override {
+      RTC_NOTREACHED();
+      return -1;
     }
-      int32_t Decoded(webrtc::VideoFrame& image) override;
 
    private:
     VideoProcessorImpl* video_processor_;
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index 3d6aedb22a..7b92616e1b 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -12,17 +12,16 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/test/testsupport/metrics/video_metrics.h"
 #include "webrtc/test/testsupport/packet_reader.h"
 #include "webrtc/typedefs.h"
@@ -81,7 +80,6 @@ struct RateControlMetrics {
   int num_key_frames;
 };
 
-
 // Sequence used is foreman (CIF): may be better to use VGA for resize test.
 const int kCIFWidth = 352;
 const int kCIFHeight = 288;
@@ -101,7 +99,7 @@ const float kScaleKeyFrameSize = 0.5f;
 // dropping/spatial resize, and temporal layers. The limits for the rate
 // control metrics are set to be fairly conservative, so failure should only
 // happen when some significant regression or breakdown occurs.
-class VideoProcessorIntegrationTest: public testing::Test {
+class VideoProcessorIntegrationTest : public testing::Test {
  protected:
   VideoEncoder* encoder_;
   VideoDecoder* decoder_;
@@ -148,7 +146,6 @@ class VideoProcessorIntegrationTest: public testing::Test {
   bool frame_dropper_on_;
   bool spatial_resize_on_;
 
-
   VideoProcessorIntegrationTest() {}
   virtual ~VideoProcessorIntegrationTest() {}
 
@@ -165,14 +162,13 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
     // CIF is currently used for all tests below.
     // Setup the TestConfig struct for processing of a clip in CIF resolution.
-    config_.input_filename =
-        webrtc::test::ResourcePath("foreman_cif", "yuv");
+    config_.input_filename = webrtc::test::ResourcePath("foreman_cif", "yuv");
 
     // Generate an output filename in a safe way.
     config_.output_filename = webrtc::test::TempFilename(
         webrtc::test::OutputPath(), "videoprocessor_integrationtest");
-    config_.frame_length_in_bytes = CalcBufferSize(kI420,
-                                                   kCIFWidth, kCIFHeight);
+    config_.frame_length_in_bytes =
+        CalcBufferSize(kI420, kCIFWidth, kCIFHeight);
     config_.verbose = false;
     // Only allow encoder/decoder to use single core, for predictability.
     config_.use_single_core = true;
@@ -188,52 +184,46 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
     // These features may be set depending on the test.
     switch (config_.codec_settings->codecType) {
-     case kVideoCodecVP8:
-       config_.codec_settings->codecSpecific.VP8.errorConcealmentOn =
-           error_concealment_on_;
-       config_.codec_settings->codecSpecific.VP8.denoisingOn =
-           denoising_on_;
-       config_.codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
-           num_temporal_layers_;
-       config_.codec_settings->codecSpecific.VP8.frameDroppingOn =
-           frame_dropper_on_;
-       config_.codec_settings->codecSpecific.VP8.automaticResizeOn =
-           spatial_resize_on_;
-       config_.codec_settings->codecSpecific.VP8.keyFrameInterval =
-           kBaseKeyFrameInterval;
-       break;
-     case kVideoCodecVP9:
-       config_.codec_settings->codecSpecific.VP9.denoisingOn =
-           denoising_on_;
-       config_.codec_settings->codecSpecific.VP9.numberOfTemporalLayers =
-           num_temporal_layers_;
-       config_.codec_settings->codecSpecific.VP9.frameDroppingOn =
-           frame_dropper_on_;
-       config_.codec_settings->codecSpecific.VP9.automaticResizeOn =
-           spatial_resize_on_;
-       config_.codec_settings->codecSpecific.VP9.keyFrameInterval =
-           kBaseKeyFrameInterval;
-       break;
-     default:
-       assert(false);
-       break;
-     }
-    frame_reader_ =
-        new webrtc::test::FrameReaderImpl(config_.input_filename,
-                                          config_.frame_length_in_bytes);
-    frame_writer_ =
-        new webrtc::test::FrameWriterImpl(config_.output_filename,
-                                          config_.frame_length_in_bytes);
+      case kVideoCodecVP8:
+        config_.codec_settings->codecSpecific.VP8.errorConcealmentOn =
+            error_concealment_on_;
+        config_.codec_settings->codecSpecific.VP8.denoisingOn = denoising_on_;
+        config_.codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
+            num_temporal_layers_;
+        config_.codec_settings->codecSpecific.VP8.frameDroppingOn =
+            frame_dropper_on_;
+        config_.codec_settings->codecSpecific.VP8.automaticResizeOn =
+            spatial_resize_on_;
+        config_.codec_settings->codecSpecific.VP8.keyFrameInterval =
+            kBaseKeyFrameInterval;
+        break;
+      case kVideoCodecVP9:
+        config_.codec_settings->codecSpecific.VP9.denoisingOn = denoising_on_;
+        config_.codec_settings->codecSpecific.VP9.numberOfTemporalLayers =
+            num_temporal_layers_;
+        config_.codec_settings->codecSpecific.VP9.frameDroppingOn =
+            frame_dropper_on_;
+        config_.codec_settings->codecSpecific.VP9.automaticResizeOn =
+            spatial_resize_on_;
+        config_.codec_settings->codecSpecific.VP9.keyFrameInterval =
+            kBaseKeyFrameInterval;
+        break;
+      default:
+        assert(false);
+        break;
+    }
+    frame_reader_ = new webrtc::test::FrameReaderImpl(
+        config_.input_filename, config_.frame_length_in_bytes);
+    frame_writer_ = new webrtc::test::FrameWriterImpl(
+        config_.output_filename, config_.frame_length_in_bytes);
     ASSERT_TRUE(frame_reader_->Init());
     ASSERT_TRUE(frame_writer_->Init());
 
     packet_manipulator_ = new webrtc::test::PacketManipulatorImpl(
         &packet_reader_, config_.networking_config, config_.verbose);
-    processor_ = new webrtc::test::VideoProcessorImpl(encoder_, decoder_,
-                                                      frame_reader_,
-                                                      frame_writer_,
-                                                      packet_manipulator_,
-                                                      config_, &stats_);
+    processor_ = new webrtc::test::VideoProcessorImpl(
+        encoder_, decoder_, frame_reader_, frame_writer_, packet_manipulator_,
+        config_, &stats_);
     ASSERT_TRUE(processor_->Init());
   }
 
@@ -247,7 +237,7 @@ class VideoProcessorIntegrationTest: public testing::Test {
       encoding_bitrate_[i] = 0.0f;
       // Update layer per-frame-bandwidth.
       per_frame_bandwidth_[i] = static_cast<float>(bit_rate_layer_[i]) /
-             static_cast<float>(frame_rate_layer_[i]);
+                                static_cast<float>(frame_rate_layer_[i]);
     }
     // Set maximum size of key frames, following setting in the VP8 wrapper.
     float max_key_size = kScaleKeyFrameSize * kOptimalBufferSize * frame_rate_;
@@ -274,28 +264,28 @@ class VideoProcessorIntegrationTest: public testing::Test {
     // Update rate mismatch relative to per-frame bandwidth for delta frames.
     if (frame_type == kVideoFrameDelta) {
       // TODO(marpan): Should we count dropped (zero size) frames in mismatch?
-      sum_frame_size_mismatch_[layer_] += fabs(encoded_size_kbits -
-                                               per_frame_bandwidth_[layer_]) /
-                                               per_frame_bandwidth_[layer_];
+      sum_frame_size_mismatch_[layer_] +=
+          fabs(encoded_size_kbits - per_frame_bandwidth_[layer_]) /
+          per_frame_bandwidth_[layer_];
     } else {
-      float target_size = (frame_num == 1) ? target_size_key_frame_initial_ :
-          target_size_key_frame_;
-      sum_key_frame_size_mismatch_ += fabs(encoded_size_kbits - target_size) /
-          target_size;
+      float target_size = (frame_num == 1) ? target_size_key_frame_initial_
+                                           : target_size_key_frame_;
+      sum_key_frame_size_mismatch_ +=
+          fabs(encoded_size_kbits - target_size) / target_size;
       num_key_frames_ += 1;
     }
     sum_encoded_frame_size_[layer_] += encoded_size_kbits;
     // Encoding bitrate per layer: from the start of the update/run to the
     // current frame.
     encoding_bitrate_[layer_] = sum_encoded_frame_size_[layer_] *
-        frame_rate_layer_[layer_] /
-        num_frames_per_update_[layer_];
+                                frame_rate_layer_[layer_] /
+                                num_frames_per_update_[layer_];
     // Total encoding rate: from the start of the update/run to current frame.
     sum_encoded_frame_size_total_ += encoded_size_kbits;
-    encoding_bitrate_total_ = sum_encoded_frame_size_total_ * frame_rate_ /
-        num_frames_total_;
-    perc_encoding_rate_mismatch_ =  100 * fabs(encoding_bitrate_total_ -
-                                               bit_rate_) / bit_rate_;
+    encoding_bitrate_total_ =
+        sum_encoded_frame_size_total_ * frame_rate_ / num_frames_total_;
+    perc_encoding_rate_mismatch_ =
+        100 * fabs(encoding_bitrate_total_ - bit_rate_) / bit_rate_;
     if (perc_encoding_rate_mismatch_ < kPercTargetvsActualMismatch &&
         !encoding_rate_within_target_) {
       num_frames_to_hit_target_ = num_frames_total_;
@@ -314,34 +304,38 @@ class VideoProcessorIntegrationTest: public testing::Test {
                          int num_key_frames) {
     int num_dropped_frames = processor_->NumberDroppedFrames();
     int num_resize_actions = processor_->NumberSpatialResizes();
-    printf("For update #: %d,\n "
+    printf(
+        "For update #: %d,\n "
         " Target Bitrate: %d,\n"
         " Encoding bitrate: %f,\n"
         " Frame rate: %d \n",
         update_index, bit_rate_, encoding_bitrate_total_, frame_rate_);
-    printf(" Number of frames to approach target rate = %d, \n"
-           " Number of dropped frames = %d, \n"
-           " Number of spatial resizes = %d, \n",
-           num_frames_to_hit_target_, num_dropped_frames, num_resize_actions);
+    printf(
+        " Number of frames to approach target rate = %d, \n"
+        " Number of dropped frames = %d, \n"
+        " Number of spatial resizes = %d, \n",
+        num_frames_to_hit_target_, num_dropped_frames, num_resize_actions);
     EXPECT_LE(perc_encoding_rate_mismatch_, max_encoding_rate_mismatch);
     if (num_key_frames_ > 0) {
-      int perc_key_frame_size_mismatch = 100 * sum_key_frame_size_mismatch_ /
-              num_key_frames_;
-      printf(" Number of Key frames: %d \n"
-             " Key frame rate mismatch: %d \n",
-             num_key_frames_, perc_key_frame_size_mismatch);
+      int perc_key_frame_size_mismatch =
+          100 * sum_key_frame_size_mismatch_ / num_key_frames_;
+      printf(
+          " Number of Key frames: %d \n"
+          " Key frame rate mismatch: %d \n",
+          num_key_frames_, perc_key_frame_size_mismatch);
       EXPECT_LE(perc_key_frame_size_mismatch, max_key_frame_size_mismatch);
     }
     printf("\n");
     printf("Rates statistics for Layer data \n");
-    for (int i = 0; i < num_temporal_layers_ ; i++) {
+    for (int i = 0; i < num_temporal_layers_; i++) {
       printf("Layer #%d \n", i);
-      int perc_frame_size_mismatch = 100 * sum_frame_size_mismatch_[i] /
-        num_frames_per_update_[i];
-      int perc_encoding_rate_mismatch = 100 * fabs(encoding_bitrate_[i] -
-                                                   bit_rate_layer_[i]) /
-                                                   bit_rate_layer_[i];
-      printf(" Target Layer Bit rate: %f \n"
+      int perc_frame_size_mismatch =
+          100 * sum_frame_size_mismatch_[i] / num_frames_per_update_[i];
+      int perc_encoding_rate_mismatch =
+          100 * fabs(encoding_bitrate_[i] - bit_rate_layer_[i]) /
+          bit_rate_layer_[i];
+      printf(
+          " Target Layer Bit rate: %f \n"
           " Layer frame rate: %f, \n"
           " Layer per frame bandwidth: %f, \n"
           " Layer Encoding bit rate: %f, \n"
@@ -366,13 +360,13 @@ class VideoProcessorIntegrationTest: public testing::Test {
     if (num_temporal_layers_ == 1) {
       layer_ = 0;
     } else if (num_temporal_layers_ == 2) {
-        // layer 0:  0     2     4 ...
-        // layer 1:     1     3
-        if (frame_number % 2 == 0) {
-          layer_ = 0;
-        } else {
-          layer_ = 1;
-        }
+      // layer 0:  0     2     4 ...
+      // layer 1:     1     3
+      if (frame_number % 2 == 0) {
+        layer_ = 0;
+      } else {
+        layer_ = 1;
+      }
     } else if (num_temporal_layers_ == 3) {
       // layer 0:  0            4            8 ...
       // layer 1:        2            6
@@ -391,20 +385,20 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
   // Set the bitrate and frame rate per layer, for up to 3 layers.
   void SetLayerRates() {
-    assert(num_temporal_layers_<= 3);
+    assert(num_temporal_layers_ <= 3);
     for (int i = 0; i < num_temporal_layers_; i++) {
       float bit_rate_ratio =
           kVp8LayerRateAlloction[num_temporal_layers_ - 1][i];
       if (i > 0) {
-        float bit_rate_delta_ratio = kVp8LayerRateAlloction
-            [num_temporal_layers_ - 1][i] -
+        float bit_rate_delta_ratio =
+            kVp8LayerRateAlloction[num_temporal_layers_ - 1][i] -
             kVp8LayerRateAlloction[num_temporal_layers_ - 1][i - 1];
         bit_rate_layer_[i] = bit_rate_ * bit_rate_delta_ratio;
       } else {
         bit_rate_layer_[i] = bit_rate_ * bit_rate_ratio;
       }
-      frame_rate_layer_[i] = frame_rate_ / static_cast<float>(
-          1 << (num_temporal_layers_ - 1));
+      frame_rate_layer_[i] =
+          frame_rate_ / static_cast<float>(1 << (num_temporal_layers_ - 1));
     }
     if (num_temporal_layers_ == 3) {
       frame_rate_layer_[2] = frame_rate_ / 2.0f;
@@ -437,12 +431,12 @@ class VideoProcessorIntegrationTest: public testing::Test {
     spatial_resize_on_ = process.spatial_resize_on;
     SetUpCodecConfig();
     // Update the layers and the codec with the initial rates.
-    bit_rate_ =  rate_profile.target_bit_rate[0];
+    bit_rate_ = rate_profile.target_bit_rate[0];
     frame_rate_ = rate_profile.input_frame_rate[0];
     SetLayerRates();
     // Set the initial target size for key frame.
-    target_size_key_frame_initial_ = 0.5 * kInitialBufferSize *
-        bit_rate_layer_[0];
+    target_size_key_frame_initial_ =
+        0.5 * kInitialBufferSize * bit_rate_layer_[0];
     processor_->SetRates(bit_rate_, frame_rate_);
     // Process each frame, up to |num_frames|.
     int num_frames = rate_profile.num_frames;
@@ -452,7 +446,7 @@ class VideoProcessorIntegrationTest: public testing::Test {
     int frame_number = 0;
     FrameType frame_type = kVideoFrameDelta;
     while (processor_->ProcessFrame(frame_number) &&
-        frame_number < num_frames) {
+           frame_number < num_frames) {
       // Get the layer index for the frame |frame_number|.
       LayerIndexForFrame(frame_number);
       // Get the frame_type.
@@ -468,8 +462,7 @@ class VideoProcessorIntegrationTest: public testing::Test {
       if (frame_number ==
           rate_profile.frame_index_rate_update[update_index + 1]) {
         VerifyRateControl(
-            update_index,
-            rc_metrics[update_index].max_key_frame_size_mismatch,
+            update_index, rc_metrics[update_index].max_key_frame_size_mismatch,
             rc_metrics[update_index].max_delta_frame_size_mismatch,
             rc_metrics[update_index].max_encoding_rate_mismatch,
             rc_metrics[update_index].max_time_hit_target,
@@ -478,23 +471,22 @@ class VideoProcessorIntegrationTest: public testing::Test {
             rc_metrics[update_index].num_key_frames);
         // Update layer rates and the codec with new rates.
         ++update_index;
-        bit_rate_ =  rate_profile.target_bit_rate[update_index];
+        bit_rate_ = rate_profile.target_bit_rate[update_index];
         frame_rate_ = rate_profile.input_frame_rate[update_index];
         SetLayerRates();
-        ResetRateControlMetrics(rate_profile.
-                                frame_index_rate_update[update_index + 1]);
+        ResetRateControlMetrics(
+            rate_profile.frame_index_rate_update[update_index + 1]);
         processor_->SetRates(bit_rate_, frame_rate_);
       }
     }
-    VerifyRateControl(
-        update_index,
-        rc_metrics[update_index].max_key_frame_size_mismatch,
-        rc_metrics[update_index].max_delta_frame_size_mismatch,
-        rc_metrics[update_index].max_encoding_rate_mismatch,
-        rc_metrics[update_index].max_time_hit_target,
-        rc_metrics[update_index].max_num_dropped_frames,
-        rc_metrics[update_index].num_spatial_resizes,
-        rc_metrics[update_index].num_key_frames);
+    VerifyRateControl(update_index,
+                      rc_metrics[update_index].max_key_frame_size_mismatch,
+                      rc_metrics[update_index].max_delta_frame_size_mismatch,
+                      rc_metrics[update_index].max_encoding_rate_mismatch,
+                      rc_metrics[update_index].max_time_hit_target,
+                      rc_metrics[update_index].max_num_dropped_frames,
+                      rc_metrics[update_index].num_spatial_resizes,
+                      rc_metrics[update_index].num_key_frames);
     EXPECT_EQ(num_frames, frame_number);
     EXPECT_EQ(num_frames + 1, static_cast<int>(stats_.stats_.size()));
 
@@ -507,16 +499,14 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
     // TODO(marpan): should compute these quality metrics per SetRates update.
     webrtc::test::QualityMetricsResult psnr_result, ssim_result;
-    EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
-        config_.input_filename.c_str(),
-        config_.output_filename.c_str(),
-        config_.codec_settings->width,
-        config_.codec_settings->height,
-        &psnr_result,
-        &ssim_result));
+    EXPECT_EQ(
+        0, webrtc::test::I420MetricsFromFiles(
+               config_.input_filename.c_str(), config_.output_filename.c_str(),
+               config_.codec_settings->width, config_.codec_settings->height,
+               &psnr_result, &ssim_result));
     printf("PSNR avg: %f, min: %f    SSIM avg: %f, min: %f\n",
-           psnr_result.average, psnr_result.min,
-           ssim_result.average, ssim_result.min);
+           psnr_result.average, psnr_result.min, ssim_result.average,
+           ssim_result.min);
     stats_.PrintSummary();
     EXPECT_GT(psnr_result.average, quality_metrics.minimum_avg_psnr);
     EXPECT_GT(psnr_result.min, quality_metrics.minimum_min_psnr);
@@ -549,7 +539,7 @@ void SetCodecParameters(CodecConfigPars* process_settings,
                         bool spatial_resize_on) {
   process_settings->codec_type = codec_type;
   process_settings->packet_loss = packet_loss;
-  process_settings->key_frame_interval =  key_frame_interval;
+  process_settings->key_frame_interval = key_frame_interval;
   process_settings->num_temporal_layers = num_temporal_layers,
   process_settings->error_concealment_on = error_concealment_on;
   process_settings->denoising_on = denoising_on;
@@ -608,9 +598,7 @@ TEST_F(VideoProcessorIntegrationTest, Process0PercentPacketLossVP9) {
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -632,13 +620,10 @@ TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLossVP9) {
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
-
 // VP9: Run with no packet loss, with varying bitrate (3 rate updates):
 // low to high to medium. Check that quality and encoder response to the new
 // target rate/per-frame bandwidth (for each rate update) is within limits.
@@ -657,15 +642,13 @@ TEST_F(VideoProcessorIntegrationTest, ProcessNoLossChangeBitRateVP9) {
                      false, true, false);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
-  SetQualityMetrics(&quality_metrics, 35.9, 30.0, 0.90, 0.85);
+  SetQualityMetrics(&quality_metrics, 35.7, 30.0, 0.90, 0.85);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
   SetRateControlMetrics(rc_metrics, 0, 0, 30, 20, 20, 30, 0, 1);
   SetRateControlMetrics(rc_metrics, 1, 2, 0, 20, 20, 60, 0, 0);
   SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 20, 40, 0, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -695,12 +678,10 @@ TEST_F(VideoProcessorIntegrationTest,
   SetQualityMetrics(&quality_metrics, 31.5, 18.0, 0.80, 0.44);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
-  SetRateControlMetrics(rc_metrics, 0, 35, 50, 70, 15, 45, 0, 1);
+  SetRateControlMetrics(rc_metrics, 0, 38, 50, 75, 15, 45, 0, 1);
   SetRateControlMetrics(rc_metrics, 1, 10, 0, 40, 10, 30, 0, 0);
   SetRateControlMetrics(rc_metrics, 2, 5, 0, 30, 5, 20, 0, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -721,19 +702,13 @@ TEST_F(VideoProcessorIntegrationTest, ProcessNoLossDenoiserOnVP9) {
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
 // Run with no packet loss, at low bitrate.
-// spatial_resize is on, and for this low bitrate expect two resizes during the
-// sequence; first resize is 3/4, second is 1/2 (from original).
+// spatial_resize is on, for this low bitrate expect one resize in sequence.
 // Resize happens on delta frame. Expect only one key frame (first frame).
-// Disable for msan, see
-// https://code.google.com/p/webrtc/issues/detail?id=5110 for details.
-#if !defined(MEMORY_SANITIZER)
 TEST_F(VideoProcessorIntegrationTest, ProcessNoLossSpatialResizeFrameDropVP9) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
@@ -743,20 +718,17 @@ TEST_F(VideoProcessorIntegrationTest, ProcessNoLossSpatialResizeFrameDropVP9) {
   rate_profile.num_frames = kNbrFramesLong;
   // Codec/network settings.
   CodecConfigPars process_settings;
-  SetCodecParameters(&process_settings, kVideoCodecVP9, 0.0f, -1,
-                     1, false, false, true, true);
+  SetCodecParameters(&process_settings, kVideoCodecVP9, 0.0f, -1, 1, false,
+                     false, true, true);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
-  SetQualityMetrics(&quality_metrics, 25.0, 13.0, 0.70, 0.40);
+  SetQualityMetrics(&quality_metrics, 24.0, 13.0, 0.65, 0.37);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 180, 70, 130, 15, 80, 2, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 228, 70, 160, 15, 80, 1, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
-#endif
 
 // TODO(marpan): Add temporal layer test for VP9, once changes are in
 // vp9 wrapper for this.
@@ -780,9 +752,7 @@ TEST_F(VideoProcessorIntegrationTest, ProcessZeroPacketLoss) {
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -804,9 +774,7 @@ TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLoss) {
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -828,9 +796,7 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -847,8 +813,13 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
 // low to high to medium. Check that quality and encoder response to the new
 // target rate/per-frame bandwidth (for each rate update) is within limits.
 // One key frame (first frame only) in sequence.
-TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossChangeBitRateVP8)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossChangeBitRateVP8 \
+  DISABLED_ProcessNoLossChangeBitRateVP8
+#else
+#define MAYBE_ProcessNoLossChangeBitRateVP8 ProcessNoLossChangeBitRateVP8
+#endif
+TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossChangeBitRateVP8) {
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
   SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
@@ -868,9 +839,7 @@ TEST_F(VideoProcessorIntegrationTest,
   SetRateControlMetrics(rc_metrics, 0, 0, 45, 20, 10, 15, 0, 1);
   SetRateControlMetrics(rc_metrics, 1, 0, 0, 25, 20, 10, 0, 0);
   SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 15, 10, 0, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -881,8 +850,15 @@ TEST_F(VideoProcessorIntegrationTest,
 // for the rate control metrics can be lower. One key frame (first frame only).
 // Note: quality after update should be higher but we currently compute quality
 // metrics averaged over whole sequence run.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
+  DISABLED_ProcessNoLossChangeFrameRateFrameDropVP8
+#else
+#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
+  ProcessNoLossChangeFrameRateFrameDropVP8
+#endif
 TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossChangeFrameRateFrameDropVP8)) {
+       MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -903,16 +879,21 @@ TEST_F(VideoProcessorIntegrationTest,
   SetRateControlMetrics(rc_metrics, 0, 40, 20, 75, 15, 60, 0, 1);
   SetRateControlMetrics(rc_metrics, 1, 10, 0, 25, 10, 35, 0, 0);
   SetRateControlMetrics(rc_metrics, 2, 0, 0, 20, 10, 15, 0, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
 // Run with no packet loss, at low bitrate. During this time we should've
 // resized once. Expect 2 key frames generated (first and one for resize).
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
+  DISABLED_ProcessNoLossSpatialResizeFrameDropVP8
+#else
+#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
+  ProcessNoLossSpatialResizeFrameDropVP8
+#endif
 TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossSpatialResizeFrameDropVP8)) {
+       MAYBE_ProcessNoLossSpatialResizeFrameDropVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -921,17 +902,15 @@ TEST_F(VideoProcessorIntegrationTest,
   rate_profile.num_frames = kNbrFramesLong;
   // Codec/network settings.
   CodecConfigPars process_settings;
-  SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, -1,
-                     1, false, true, true, true);
+  SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, -1, 1, false,
+                     true, true, true);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
   SetQualityMetrics(&quality_metrics, 25.0, 15.0, 0.70, 0.40);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 160, 60, 120, 20, 70, 1, 2);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -940,8 +919,13 @@ TEST_F(VideoProcessorIntegrationTest,
 // encoding rate mismatch are applied to each layer.
 // No dropped frames in this test, and internal spatial resizer is off.
 // One key frame (first frame only) in sequence, so no spatial resizing.
-TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossTemporalLayersVP8)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossTemporalLayersVP8 \
+  DISABLED_ProcessNoLossTemporalLayersVP8
+#else
+#define MAYBE_ProcessNoLossTemporalLayersVP8 ProcessNoLossTemporalLayersVP8
+#endif
+TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossTemporalLayersVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -960,9 +944,7 @@ TEST_F(VideoProcessorIntegrationTest,
   RateControlMetrics rc_metrics[2];
   SetRateControlMetrics(rc_metrics, 0, 0, 20, 30, 10, 10, 0, 1);
   SetRateControlMetrics(rc_metrics, 1, 0, 0, 30, 15, 10, 0, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
index 88b5467f1f..148d8dc74a 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
@@ -10,10 +10,10 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/test/testsupport/mock/mock_frame_reader.h"
 #include "webrtc/test/testsupport/mock/mock_frame_writer.h"
 #include "webrtc/test/testsupport/packet_reader.h"
@@ -29,7 +29,7 @@ namespace test {
 
 // Very basic testing for VideoProcessor. It's mostly tested by running the
 // video_quality_measurement program.
-class VideoProcessorTest: public testing::Test {
+class VideoProcessorTest : public testing::Test {
  protected:
   MockVideoEncoder encoder_mock_;
   MockVideoDecoder decoder_mock_;
@@ -53,44 +53,34 @@ class VideoProcessorTest: public testing::Test {
   void TearDown() {}
 
   void ExpectInit() {
-    EXPECT_CALL(encoder_mock_, InitEncode(_, _, _))
-      .Times(1);
+    EXPECT_CALL(encoder_mock_, InitEncode(_, _, _)).Times(1);
     EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_))
-      .Times(AtLeast(1));
-    EXPECT_CALL(decoder_mock_, InitDecode(_, _))
-      .Times(1);
+        .Times(AtLeast(1));
+    EXPECT_CALL(decoder_mock_, InitDecode(_, _)).Times(1);
     EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_))
-      .Times(AtLeast(1));
-    EXPECT_CALL(frame_reader_mock_, NumberOfFrames())
-      .WillOnce(Return(1));
-    EXPECT_CALL(frame_reader_mock_, FrameLength())
-      .WillOnce(Return(152064));
+        .Times(AtLeast(1));
+    EXPECT_CALL(frame_reader_mock_, NumberOfFrames()).WillOnce(Return(1));
+    EXPECT_CALL(frame_reader_mock_, FrameLength()).WillOnce(Return(152064));
   }
 };
 
 TEST_F(VideoProcessorTest, Init) {
   ExpectInit();
-  VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
-                                     &frame_reader_mock_,
-                                     &frame_writer_mock_,
-                                     &packet_manipulator_mock_, config_,
-                                     &stats_);
+  VideoProcessorImpl video_processor(
+      &encoder_mock_, &decoder_mock_, &frame_reader_mock_, &frame_writer_mock_,
+      &packet_manipulator_mock_, config_, &stats_);
   ASSERT_TRUE(video_processor.Init());
 }
 
 TEST_F(VideoProcessorTest, ProcessFrame) {
   ExpectInit();
-  EXPECT_CALL(encoder_mock_, Encode(_, _, _))
-    .Times(1);
-  EXPECT_CALL(frame_reader_mock_, ReadFrame(_))
-    .WillOnce(Return(true));
+  EXPECT_CALL(encoder_mock_, Encode(_, _, _)).Times(1);
+  EXPECT_CALL(frame_reader_mock_, ReadFrame(_)).WillOnce(Return(true));
   // Since we don't return any callback from the mock, the decoder will not
   // be more than initialized...
-  VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
-                                     &frame_reader_mock_,
-                                     &frame_writer_mock_,
-                                     &packet_manipulator_mock_, config_,
-                                     &stats_);
+  VideoProcessorImpl video_processor(
+      &encoder_mock_, &decoder_mock_, &frame_reader_mock_, &frame_writer_mock_,
+      &packet_manipulator_mock_, config_, &stats_);
   ASSERT_TRUE(video_processor.Init());
   video_processor.ProcessFrame(0);
 }
diff --git a/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc b/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
index 22be5a83cc..37fad483f7 100644
--- a/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
+++ b/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
@@ -16,7 +16,7 @@
 #include <sys/stat.h>  // To check for directory existence.
 
 #ifndef S_ISDIR  // Not defined in stat.h on Windows.
-#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
+#define S_ISDIR(mode) (((mode)&S_IFMT) == S_IFDIR)
 #endif
 
 #include "gflags/gflags.h"
@@ -26,7 +26,7 @@
 #include "webrtc/modules/video_coding/codecs/test/stats.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
@@ -34,68 +34,102 @@
 #include "webrtc/test/testsupport/packet_reader.h"
 
 DEFINE_string(test_name, "Quality test", "The name of the test to run. ");
-DEFINE_string(test_description, "", "A more detailed description about what "
+DEFINE_string(test_description,
+              "",
+              "A more detailed description about what "
               "the current test is about.");
-DEFINE_string(input_filename, "", "Input file. "
+DEFINE_string(input_filename,
+              "",
+              "Input file. "
               "The source video file to be encoded and decoded. Must be in "
               ".yuv format");
 DEFINE_int32(width, -1, "Width in pixels of the frames in the input file.");
 DEFINE_int32(height, -1, "Height in pixels of the frames in the input file.");
-DEFINE_int32(framerate, 30, "Frame rate of the input file, in FPS "
+DEFINE_int32(framerate,
+             30,
+             "Frame rate of the input file, in FPS "
              "(frames-per-second). ");
-DEFINE_string(output_dir, ".", "Output directory. "
+DEFINE_string(output_dir,
+              ".",
+              "Output directory. "
               "The directory where the output file will be put. Must already "
               "exist.");
-DEFINE_bool(use_single_core, false, "Force using a single core. If set to "
+DEFINE_bool(use_single_core,
+            false,
+            "Force using a single core. If set to "
             "true, only one core will be used for processing. Using a single "
             "core is necessary to get a deterministic behavior for the"
             "encoded frames - using multiple cores will produce different "
             "encoded frames since multiple cores are competing to consume the "
             "byte budget for each frame in parallel. If set to false, "
             "the maximum detected number of cores will be used. ");
-DEFINE_bool(disable_fixed_random_seed , false, "Set this flag to disable the"
+DEFINE_bool(disable_fixed_random_seed,
+            false,
+            "Set this flag to disable the"
             "usage of a fixed random seed for the random generator used "
             "for packet loss. Disabling this will cause consecutive runs "
             "loose packets at different locations, which is bad for "
             "reproducibility.");
-DEFINE_string(output_filename, "", "Output file. "
+DEFINE_string(output_filename,
+              "",
+              "Output file. "
               "The name of the output video file resulting of the processing "
               "of the source file. By default this is the same name as the "
               "input file with '_out' appended before the extension.");
 DEFINE_int32(bitrate, 500, "Bit rate in kilobits/second.");
-DEFINE_int32(keyframe_interval, 0, "Forces a keyframe every Nth frame. "
+DEFINE_int32(keyframe_interval,
+             0,
+             "Forces a keyframe every Nth frame. "
              "0 means the encoder decides when to insert keyframes.  Note that "
              "the encoder may create a keyframe in other locations in addition "
              "to the interval that is set using this parameter.");
-DEFINE_int32(temporal_layers, 0, "The number of temporal layers to use "
+DEFINE_int32(temporal_layers,
+             0,
+             "The number of temporal layers to use "
              "(VP8 specific codec setting). Must be 0-4.");
-DEFINE_int32(packet_size, 1500, "Simulated network packet size in bytes (MTU). "
+DEFINE_int32(packet_size,
+             1500,
+             "Simulated network packet size in bytes (MTU). "
              "Used for packet loss simulation.");
-DEFINE_int32(max_payload_size, 1440, "Max payload size in bytes for the "
+DEFINE_int32(max_payload_size,
+             1440,
+             "Max payload size in bytes for the "
              "encoder.");
-DEFINE_string(packet_loss_mode, "uniform", "Packet loss mode. Two different "
+DEFINE_string(packet_loss_mode,
+              "uniform",
+              "Packet loss mode. Two different "
               "packet loss models are supported: uniform or burst. This "
               "setting has no effect unless packet_loss_rate is >0. ");
-DEFINE_double(packet_loss_probability, 0.0, "Packet loss probability. A value "
+DEFINE_double(packet_loss_probability,
+              0.0,
+              "Packet loss probability. A value "
               "between 0.0 and 1.0 that defines the probability of a packet "
               "being lost. 0.1 means 10% and so on.");
-DEFINE_int32(packet_loss_burst_length, 1, "Packet loss burst length. Defines "
+DEFINE_int32(packet_loss_burst_length,
+             1,
+             "Packet loss burst length. Defines "
              "how many packets will be lost in a burst when a packet has been "
              "decided to be lost. Must be >=1.");
-DEFINE_bool(csv, false, "CSV output. Enabling this will output all frame "
+DEFINE_bool(csv,
+            false,
+            "CSV output. Enabling this will output all frame "
             "statistics at the end of execution. Recommended to run combined "
             "with --noverbose to avoid mixing output.");
-DEFINE_bool(python, false, "Python output. Enabling this will output all frame "
+DEFINE_bool(python,
+            false,
+            "Python output. Enabling this will output all frame "
             "statistics as a Python script at the end of execution. "
             "Recommended to run combine with --noverbose to avoid mixing "
             "output.");
-DEFINE_bool(verbose, true, "Verbose mode. Prints a lot of debugging info. "
+DEFINE_bool(verbose,
+            true,
+            "Verbose mode. Prints a lot of debugging info. "
             "Suitable for tracking progress but not for capturing output. "
             "Disable with --noverbose flag.");
 
 // Custom log method that only prints if the verbose flag is given.
 // Supports all the standard printf parameters and formatting (just forwarded).
-int Log(const char *format, ...) {
+int Log(const char* format, ...) {
   int result = 0;
   if (FLAGS_verbose) {
     va_list args;
@@ -132,9 +166,9 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
   // Verify the output dir exists.
   struct stat dir_info;
   if (!(stat(FLAGS_output_dir.c_str(), &dir_info) == 0 &&
-      S_ISDIR(dir_info.st_mode))) {
+        S_ISDIR(dir_info.st_mode))) {
     fprintf(stderr, "Cannot find output directory: %s\n",
-              FLAGS_output_dir.c_str());
+            FLAGS_output_dir.c_str());
     return 3;
   }
   config->output_dir = FLAGS_output_dir;
@@ -148,16 +182,16 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
       startIndex = 0;
     }
     FLAGS_output_filename =
-        FLAGS_input_filename.substr(startIndex,
-                                    FLAGS_input_filename.find_last_of(".")
-                                    - startIndex) + "_out.yuv";
+        FLAGS_input_filename.substr(
+            startIndex, FLAGS_input_filename.find_last_of(".") - startIndex) +
+        "_out.yuv";
   }
 
   // Verify output file can be written.
   if (FLAGS_output_dir == ".") {
     config->output_filename = FLAGS_output_filename;
   } else {
-    config->output_filename = FLAGS_output_dir + "/"+ FLAGS_output_filename;
+    config->output_filename = FLAGS_output_dir + "/" + FLAGS_output_filename;
   }
   test_file = fopen(config->output_filename.c_str(), "wb");
   if (test_file == NULL) {
@@ -232,27 +266,32 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
   // Check packet loss settings
   if (FLAGS_packet_loss_mode != "uniform" &&
       FLAGS_packet_loss_mode != "burst") {
-    fprintf(stderr, "Unsupported packet loss mode, must be 'uniform' or "
+    fprintf(stderr,
+            "Unsupported packet loss mode, must be 'uniform' or "
             "'burst'\n.");
     return 10;
   }
   config->networking_config.packet_loss_mode = webrtc::test::kUniform;
   if (FLAGS_packet_loss_mode == "burst") {
-    config->networking_config.packet_loss_mode =  webrtc::test::kBurst;
+    config->networking_config.packet_loss_mode = webrtc::test::kBurst;
   }
 
   if (FLAGS_packet_loss_probability < 0.0 ||
       FLAGS_packet_loss_probability > 1.0) {
-    fprintf(stderr, "Invalid packet loss probability. Must be 0.0 - 1.0, "
-            "was: %f\n", FLAGS_packet_loss_probability);
+    fprintf(stderr,
+            "Invalid packet loss probability. Must be 0.0 - 1.0, "
+            "was: %f\n",
+            FLAGS_packet_loss_probability);
     return 11;
   }
   config->networking_config.packet_loss_probability =
       FLAGS_packet_loss_probability;
 
   if (FLAGS_packet_loss_burst_length < 1) {
-    fprintf(stderr, "Invalid packet loss burst length, must be >=1, "
-            "was: %d\n", FLAGS_packet_loss_burst_length);
+    fprintf(stderr,
+            "Invalid packet loss burst length, must be >=1, "
+            "was: %d\n",
+            FLAGS_packet_loss_burst_length);
     return 12;
   }
   config->networking_config.packet_loss_burst_length =
@@ -264,10 +303,9 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
 void CalculateSsimVideoMetrics(webrtc::test::TestConfig* config,
                                webrtc::test::QualityMetricsResult* result) {
   Log("Calculating SSIM...\n");
-  I420SSIMFromFiles(config->input_filename.c_str(),
-                    config->output_filename.c_str(),
-                    config->codec_settings->width,
-                    config->codec_settings->height, result);
+  I420SSIMFromFiles(
+      config->input_filename.c_str(), config->output_filename.c_str(),
+      config->codec_settings->width, config->codec_settings->height, result);
   Log("  Average: %3.2f\n", result->average);
   Log("  Min    : %3.2f (frame %d)\n", result->min, result->min_frame_number);
   Log("  Max    : %3.2f (frame %d)\n", result->max, result->max_frame_number);
@@ -276,10 +314,9 @@ void CalculateSsimVideoMetrics(webrtc::test::TestConfig* config,
 void CalculatePsnrVideoMetrics(webrtc::test::TestConfig* config,
                                webrtc::test::QualityMetricsResult* result) {
   Log("Calculating PSNR...\n");
-  I420PSNRFromFiles(config->input_filename.c_str(),
-                    config->output_filename.c_str(),
-                    config->codec_settings->width,
-                    config->codec_settings->height, result);
+  I420PSNRFromFiles(
+      config->input_filename.c_str(), config->output_filename.c_str(),
+      config->codec_settings->width, config->codec_settings->height, result);
   Log("  Average: %3.2f\n", result->average);
   Log("  Min    : %3.2f (frame %d)\n", result->min, result->min_frame_number);
   Log("  Max    : %3.2f (frame %d)\n", result->max, result->max_frame_number);
@@ -309,9 +346,11 @@ void PrintConfigurationSummary(const webrtc::test::TestConfig& config) {
 void PrintCsvOutput(const webrtc::test::Stats& stats,
                     const webrtc::test::QualityMetricsResult& ssim_result,
                     const webrtc::test::QualityMetricsResult& psnr_result) {
-  Log("\nCSV output (recommended to run with --noverbose to skip the "
-              "above output)\n");
-  printf("frame_number encoding_successful decoding_successful "
+  Log(
+      "\nCSV output (recommended to run with --noverbose to skip the "
+      "above output)\n");
+  printf(
+      "frame_number encoding_successful decoding_successful "
       "encode_return_code decode_return_code "
       "encode_time_in_us decode_time_in_us "
       "bit_rate_in_kbps encoded_frame_length_in_bytes frame_type "
@@ -322,22 +361,13 @@ void PrintCsvOutput(const webrtc::test::Stats& stats,
     const webrtc::test::FrameStatistic& f = stats.stats_[i];
     const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
     const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
-    printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7" PRIuS ", %d, %2d, %2"
-           PRIuS ", %5.3f, %5.2f\n",
-           f.frame_number,
-           f.encoding_successful,
-           f.decoding_successful,
-           f.encode_return_code,
-           f.decode_return_code,
-           f.encode_time_in_us,
-           f.decode_time_in_us,
-           f.bit_rate_in_kbps,
-           f.encoded_frame_length_in_bytes,
-           f.frame_type,
-           f.packets_dropped,
-           f.total_packets,
-           ssim.value,
-           psnr.value);
+    printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7" PRIuS
+           ", %d, %2d, %2" PRIuS ", %5.3f, %5.2f\n",
+           f.frame_number, f.encoding_successful, f.decoding_successful,
+           f.encode_return_code, f.decode_return_code, f.encode_time_in_us,
+           f.decode_time_in_us, f.bit_rate_in_kbps,
+           f.encoded_frame_length_in_bytes, f.frame_type, f.packets_dropped,
+           f.total_packets, ssim.value, psnr.value);
   }
 }
 
@@ -345,91 +375,85 @@ void PrintPythonOutput(const webrtc::test::TestConfig& config,
                        const webrtc::test::Stats& stats,
                        const webrtc::test::QualityMetricsResult& ssim_result,
                        const webrtc::test::QualityMetricsResult& psnr_result) {
-  Log("\nPython output (recommended to run with --noverbose to skip the "
-               "above output)\n");
-  printf("test_configuration = ["
-         "{'name': 'name',                      'value': '%s'},\n"
-         "{'name': 'description',               'value': '%s'},\n"
-         "{'name': 'test_number',               'value': '%d'},\n"
-         "{'name': 'input_filename',            'value': '%s'},\n"
-         "{'name': 'output_filename',           'value': '%s'},\n"
-         "{'name': 'output_dir',                'value': '%s'},\n"
-         "{'name': 'packet_size_in_bytes',      'value': '%" PRIuS "'},\n"
-         "{'name': 'max_payload_size_in_bytes', 'value': '%" PRIuS "'},\n"
-         "{'name': 'packet_loss_mode',          'value': '%s'},\n"
-         "{'name': 'packet_loss_probability',   'value': '%f'},\n"
-         "{'name': 'packet_loss_burst_length',  'value': '%d'},\n"
-         "{'name': 'exclude_frame_types',       'value': '%s'},\n"
-         "{'name': 'frame_length_in_bytes',     'value': '%" PRIuS "'},\n"
-         "{'name': 'use_single_core',           'value': '%s'},\n"
-         "{'name': 'keyframe_interval;',        'value': '%d'},\n"
-         "{'name': 'video_codec_type',          'value': '%s'},\n"
-         "{'name': 'width',                     'value': '%d'},\n"
-         "{'name': 'height',                    'value': '%d'},\n"
-         "{'name': 'bit_rate_in_kbps',          'value': '%d'},\n"
-         "]\n",
-         config.name.c_str(),
-         config.description.c_str(),
-         config.test_number,
-         config.input_filename.c_str(),
-         config.output_filename.c_str(),
-         config.output_dir.c_str(),
-         config.networking_config.packet_size_in_bytes,
-         config.networking_config.max_payload_size_in_bytes,
-         PacketLossModeToStr(config.networking_config.packet_loss_mode),
-         config.networking_config.packet_loss_probability,
-         config.networking_config.packet_loss_burst_length,
-         ExcludeFrameTypesToStr(config.exclude_frame_types),
-         config.frame_length_in_bytes,
-         config.use_single_core ? "True " : "False",
-         config.keyframe_interval,
-         webrtc::test::VideoCodecTypeToStr(config.codec_settings->codecType),
-         config.codec_settings->width,
-         config.codec_settings->height,
-         config.codec_settings->startBitrate);
-  printf("frame_data_types = {"
-         "'frame_number': ('number', 'Frame number'),\n"
-         "'encoding_successful': ('boolean', 'Encoding successful?'),\n"
-         "'decoding_successful': ('boolean', 'Decoding successful?'),\n"
-         "'encode_time': ('number', 'Encode time (us)'),\n"
-         "'decode_time': ('number', 'Decode time (us)'),\n"
-         "'encode_return_code': ('number', 'Encode return code'),\n"
-         "'decode_return_code': ('number', 'Decode return code'),\n"
-         "'bit_rate': ('number', 'Bit rate (kbps)'),\n"
-         "'encoded_frame_length': "
-         "('number', 'Encoded frame length (bytes)'),\n"
-         "'frame_type': ('string', 'Frame type'),\n"
-         "'packets_dropped': ('number', 'Packets dropped'),\n"
-         "'total_packets': ('number', 'Total packets'),\n"
-         "'ssim': ('number', 'SSIM'),\n"
-         "'psnr': ('number', 'PSNR (dB)'),\n"
-         "}\n");
+  Log(
+      "\nPython output (recommended to run with --noverbose to skip the "
+      "above output)\n");
+  printf(
+      "test_configuration = ["
+      "{'name': 'name',                      'value': '%s'},\n"
+      "{'name': 'description',               'value': '%s'},\n"
+      "{'name': 'test_number',               'value': '%d'},\n"
+      "{'name': 'input_filename',            'value': '%s'},\n"
+      "{'name': 'output_filename',           'value': '%s'},\n"
+      "{'name': 'output_dir',                'value': '%s'},\n"
+      "{'name': 'packet_size_in_bytes',      'value': '%" PRIuS
+      "'},\n"
+      "{'name': 'max_payload_size_in_bytes', 'value': '%" PRIuS
+      "'},\n"
+      "{'name': 'packet_loss_mode',          'value': '%s'},\n"
+      "{'name': 'packet_loss_probability',   'value': '%f'},\n"
+      "{'name': 'packet_loss_burst_length',  'value': '%d'},\n"
+      "{'name': 'exclude_frame_types',       'value': '%s'},\n"
+      "{'name': 'frame_length_in_bytes',     'value': '%" PRIuS
+      "'},\n"
+      "{'name': 'use_single_core',           'value': '%s'},\n"
+      "{'name': 'keyframe_interval;',        'value': '%d'},\n"
+      "{'name': 'video_codec_type',          'value': '%s'},\n"
+      "{'name': 'width',                     'value': '%d'},\n"
+      "{'name': 'height',                    'value': '%d'},\n"
+      "{'name': 'bit_rate_in_kbps',          'value': '%d'},\n"
+      "]\n",
+      config.name.c_str(), config.description.c_str(), config.test_number,
+      config.input_filename.c_str(), config.output_filename.c_str(),
+      config.output_dir.c_str(), config.networking_config.packet_size_in_bytes,
+      config.networking_config.max_payload_size_in_bytes,
+      PacketLossModeToStr(config.networking_config.packet_loss_mode),
+      config.networking_config.packet_loss_probability,
+      config.networking_config.packet_loss_burst_length,
+      ExcludeFrameTypesToStr(config.exclude_frame_types),
+      config.frame_length_in_bytes, config.use_single_core ? "True " : "False",
+      config.keyframe_interval,
+      webrtc::test::VideoCodecTypeToStr(config.codec_settings->codecType),
+      config.codec_settings->width, config.codec_settings->height,
+      config.codec_settings->startBitrate);
+  printf(
+      "frame_data_types = {"
+      "'frame_number': ('number', 'Frame number'),\n"
+      "'encoding_successful': ('boolean', 'Encoding successful?'),\n"
+      "'decoding_successful': ('boolean', 'Decoding successful?'),\n"
+      "'encode_time': ('number', 'Encode time (us)'),\n"
+      "'decode_time': ('number', 'Decode time (us)'),\n"
+      "'encode_return_code': ('number', 'Encode return code'),\n"
+      "'decode_return_code': ('number', 'Decode return code'),\n"
+      "'bit_rate': ('number', 'Bit rate (kbps)'),\n"
+      "'encoded_frame_length': "
+      "('number', 'Encoded frame length (bytes)'),\n"
+      "'frame_type': ('string', 'Frame type'),\n"
+      "'packets_dropped': ('number', 'Packets dropped'),\n"
+      "'total_packets': ('number', 'Total packets'),\n"
+      "'ssim': ('number', 'SSIM'),\n"
+      "'psnr': ('number', 'PSNR (dB)'),\n"
+      "}\n");
   printf("frame_data = [");
   for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
     const webrtc::test::FrameStatistic& f = stats.stats_[i];
     const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
     const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
-    printf("{'frame_number': %d, "
-           "'encoding_successful': %s, 'decoding_successful': %s, "
-           "'encode_time': %d, 'decode_time': %d, "
-           "'encode_return_code': %d, 'decode_return_code': %d, "
-           "'bit_rate': %d, 'encoded_frame_length': %" PRIuS ", "
-           "'frame_type': %s, 'packets_dropped': %d, "
-           "'total_packets': %" PRIuS ", 'ssim': %f, 'psnr': %f},\n",
-           f.frame_number,
-           f.encoding_successful ? "True " : "False",
-           f.decoding_successful ? "True " : "False",
-           f.encode_time_in_us,
-           f.decode_time_in_us,
-           f.encode_return_code,
-           f.decode_return_code,
-           f.bit_rate_in_kbps,
-           f.encoded_frame_length_in_bytes,
-           f.frame_type == webrtc::kVideoFrameDelta ? "'Delta'" : "'Other'",
-           f.packets_dropped,
-           f.total_packets,
-           ssim.value,
-           psnr.value);
+    printf(
+        "{'frame_number': %d, "
+        "'encoding_successful': %s, 'decoding_successful': %s, "
+        "'encode_time': %d, 'decode_time': %d, "
+        "'encode_return_code': %d, 'decode_return_code': %d, "
+        "'bit_rate': %d, 'encoded_frame_length': %" PRIuS
+        ", "
+        "'frame_type': %s, 'packets_dropped': %d, "
+        "'total_packets': %" PRIuS ", 'ssim': %f, 'psnr': %f},\n",
+        f.frame_number, f.encoding_successful ? "True " : "False",
+        f.decoding_successful ? "True " : "False", f.encode_time_in_us,
+        f.decode_time_in_us, f.encode_return_code, f.decode_return_code,
+        f.bit_rate_in_kbps, f.encoded_frame_length_in_bytes,
+        f.frame_type == webrtc::kVideoFrameDelta ? "'Delta'" : "'Other'",
+        f.packets_dropped, f.total_packets, ssim.value, psnr.value);
   }
   printf("]\n");
 }
@@ -438,10 +462,14 @@ void PrintPythonOutput(const webrtc::test::TestConfig& config,
 // The input file must be in YUV format.
 int main(int argc, char* argv[]) {
   std::string program_name = argv[0];
-  std::string usage = "Quality test application for video comparisons.\n"
-    "Run " + program_name + " --helpshort for usage.\n"
-    "Example usage:\n" + program_name +
-    " --input_filename=filename.yuv --width=352 --height=288\n";
+  std::string usage =
+      "Quality test application for video comparisons.\n"
+      "Run " +
+      program_name +
+      " --helpshort for usage.\n"
+      "Example usage:\n" +
+      program_name +
+      " --input_filename=filename.yuv --width=352 --height=288\n";
   google::SetUsageMessage(usage);
 
   google::ParseCommandLineFlags(&argc, &argv, true);
@@ -478,10 +506,8 @@ int main(int argc, char* argv[]) {
     packet_manipulator.InitializeRandomSeed(time(NULL));
   }
   webrtc::test::VideoProcessor* processor =
-      new webrtc::test::VideoProcessorImpl(encoder, decoder,
-                                           &frame_reader,
-                                           &frame_writer,
-                                           &packet_manipulator,
+      new webrtc::test::VideoProcessorImpl(encoder, decoder, &frame_reader,
+                                           &frame_writer, &packet_manipulator,
                                            config, &stats);
   processor->Init();
 
diff --git a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
index da6008ba3d..9226fa774c 100644
--- a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
@@ -13,8 +13,8 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 
 #include "vpx/vpx_encoder.h"
@@ -41,7 +41,7 @@ int DefaultTemporalLayers::CurrentLayerId() const {
   int index = pattern_idx_ % temporal_ids_length_;
   assert(index >= 0);
   return temporal_ids_[index];
- }
+}
 
 bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
                                               int max_bitrate_kbit,
@@ -56,8 +56,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_periodicity = temporal_ids_length_;
       cfg->ts_target_bitrate[0] = bitrateKbit;
       cfg->ts_rate_decimator[0] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 1;
       temporal_pattern_[0] = kTemporalUpdateLastRefAll;
@@ -74,8 +73,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_target_bitrate[1] = bitrateKbit;
       cfg->ts_rate_decimator[0] = 2;
       cfg->ts_rate_decimator[1] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 8;
       temporal_pattern_[0] = kTemporalUpdateLastAndGoldenRefAltRef;
@@ -103,8 +101,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_rate_decimator[0] = 4;
       cfg->ts_rate_decimator[1] = 2;
       cfg->ts_rate_decimator[2] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 8;
       temporal_pattern_[0] = kTemporalUpdateLastAndGoldenRefAltRef;
@@ -138,8 +135,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_rate_decimator[1] = 4;
       cfg->ts_rate_decimator[2] = 2;
       cfg->ts_rate_decimator[3] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 16;
       temporal_pattern_[0] = kTemporalUpdateLast;
@@ -243,7 +239,7 @@ int DefaultTemporalLayers::EncodeFlags(uint32_t timestamp) {
 
 void DefaultTemporalLayers::PopulateCodecSpecific(
     bool base_layer_sync,
-    CodecSpecificInfoVP8 *vp8_info,
+    CodecSpecificInfoVP8* vp8_info,
     uint32_t timestamp) {
   assert(number_of_temporal_layers_ > 0);
   assert(0 < temporal_ids_length_);
@@ -254,8 +250,8 @@ void DefaultTemporalLayers::PopulateCodecSpecific(
     vp8_info->tl0PicIdx = kNoTl0PicIdx;
   } else {
     if (base_layer_sync) {
-    vp8_info->temporalIdx = 0;
-    vp8_info->layerSync = true;
+      vp8_info->temporalIdx = 0;
+      vp8_info->layerSync = true;
     } else {
       vp8_info->temporalIdx = CurrentLayerId();
       TemporalReferences temporal_reference =
@@ -267,7 +263,7 @@ void DefaultTemporalLayers::PopulateCodecSpecific(
               kTemporalUpdateGoldenWithoutDependencyRefAltRef ||
           temporal_reference == kTemporalUpdateNoneNoRefGoldenRefAltRef ||
           (temporal_reference == kTemporalUpdateNone &&
-              number_of_temporal_layers_ == 4)) {
+           number_of_temporal_layers_ == 4)) {
         vp8_info->layerSync = true;
       } else {
         vp8_info->layerSync = false;
diff --git a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
index 34121cbcf6..461ba69a72 100644
--- a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
@@ -8,9 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h"
 
 #include "vpx/vpx_encoder.h"
@@ -19,47 +18,36 @@
 namespace webrtc {
 
 enum {
-  kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF |
-                        VP8_EFLAG_NO_UPD_ARF |
+  kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
                         VP8_EFLAG_NO_REF_GF |
                         VP8_EFLAG_NO_REF_ARF,
-  kTemporalUpdateGoldenWithoutDependency = VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_UPD_ARF |
-                                           VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateGolden = VP8_EFLAG_NO_REF_ARF |
-                          VP8_EFLAG_NO_UPD_ARF |
-                          VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateAltrefWithoutDependency = VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF |
-                          VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF |
-                        VP8_EFLAG_NO_UPD_ARF |
+  kTemporalUpdateGoldenWithoutDependency =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF |
+      VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateGolden =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateAltrefWithoutDependency =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF |
+      VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
                         VP8_EFLAG_NO_UPD_LAST |
                         VP8_EFLAG_NO_UPD_ENTROPY,
-  kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF |
-                                   VP8_EFLAG_NO_UPD_GF |
+  kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF |
                                    VP8_EFLAG_NO_UPD_ARF |
                                    VP8_EFLAG_NO_UPD_LAST |
                                    VP8_EFLAG_NO_UPD_ENTROPY,
-  kTemporalUpdateNoneNoRefGolden = VP8_EFLAG_NO_REF_GF |
-                                   VP8_EFLAG_NO_UPD_GF |
+  kTemporalUpdateNoneNoRefGolden = VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF |
                                    VP8_EFLAG_NO_UPD_ARF |
                                    VP8_EFLAG_NO_UPD_LAST |
                                    VP8_EFLAG_NO_UPD_ENTROPY,
-  kTemporalUpdateGoldenWithoutDependencyRefAltRef = VP8_EFLAG_NO_REF_GF |
-                                                    VP8_EFLAG_NO_UPD_ARF |
-                                                    VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF |
-                                   VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateLastRefAltRef = VP8_EFLAG_NO_UPD_GF |
-                                 VP8_EFLAG_NO_UPD_ARF |
-                                 VP8_EFLAG_NO_REF_GF,
-  kTemporalUpdateLastAndGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF |
-                                          VP8_EFLAG_NO_REF_GF,
+  kTemporalUpdateGoldenWithoutDependencyRefAltRef =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateLastRefAltRef =
+      VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF,
+  kTemporalUpdateLastAndGoldenRefAltRef =
+      VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF,
 };
 
 TEST(TemporalLayersTest, 2Layers) {
@@ -68,29 +56,30 @@ TEST(TemporalLayersTest, 2Layers) {
   CodecSpecificInfoVP8 vp8_info;
   tl.ConfigureBitrates(500, 500, 30, &cfg);
 
-  int expected_flags[16] = { kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-   };
-  int expected_temporal_idx[16] =
-      { 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1 };
-
-  bool expected_layer_sync[16] =
-      { false, true, false, false, false, false, false, false,
-        false, true, false, false, false, false, false, false };
+  int expected_flags[16] = {
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+  };
+  int expected_temporal_idx[16] = {0, 1, 0, 1, 0, 1, 0, 1,
+                                   0, 1, 0, 1, 0, 1, 0, 1};
+
+  bool expected_layer_sync[16] = {false, true,  false, false, false, false,
+                                  false, false, false, true,  false, false,
+                                  false, false, false, false};
 
   uint32_t timestamp = 0;
   for (int i = 0; i < 16; ++i) {
@@ -108,29 +97,30 @@ TEST(TemporalLayersTest, 3Layers) {
   CodecSpecificInfoVP8 vp8_info;
   tl.ConfigureBitrates(500, 500, 30, &cfg);
 
-  int expected_flags[16] = { kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateNoneNoRefGolden,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateNoneNoRefGolden,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateNone,
+  int expected_flags[16] = {
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateNoneNoRefGolden,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateNoneNoRefGolden,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateNone,
   };
-  int expected_temporal_idx[16] =
-      { 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2 };
+  int expected_temporal_idx[16] = {0, 2, 1, 2, 0, 2, 1, 2,
+                                   0, 2, 1, 2, 0, 2, 1, 2};
 
-  bool expected_layer_sync[16] =
-      { false, true, true, false, false, false, false, false,
-        false, true, true, false, false, false, false, false };
+  bool expected_layer_sync[16] = {false, true,  true,  false, false, false,
+                                  false, false, false, true,  true,  false,
+                                  false, false, false, false};
 
   unsigned int timestamp = 0;
   for (int i = 0; i < 16; ++i) {
@@ -165,12 +155,12 @@ TEST(TemporalLayersTest, 4Layers) {
       kTemporalUpdateAltref,
       kTemporalUpdateNone,
   };
-  int expected_temporal_idx[16] =
-      { 0, 3, 2, 3, 1, 3, 2, 3, 0, 3, 2, 3, 1, 3, 2, 3 };
+  int expected_temporal_idx[16] = {0, 3, 2, 3, 1, 3, 2, 3,
+                                   0, 3, 2, 3, 1, 3, 2, 3};
 
-  bool expected_layer_sync[16] =
-      { false, true, true, true, true, true, false, true,
-        false, true, false, true, false, true, false, true };
+  bool expected_layer_sync[16] = {false, true, true,  true, true,  true,
+                                  false, true, false, true, false, true,
+                                  false, true, false, true};
 
   uint32_t timestamp = 0;
   for (int i = 0; i < 16; ++i) {
@@ -198,8 +188,7 @@ TEST(TemporalLayersTest, KeyFrame) {
       kTemporalUpdateGoldenRefAltRef,
       kTemporalUpdateNone,
   };
-  int expected_temporal_idx[8] =
-      { 0, 0, 0, 0, 0, 0, 0, 2};
+  int expected_temporal_idx[8] = {0, 0, 0, 0, 0, 0, 0, 2};
 
   uint32_t timestamp = 0;
   for (int i = 0; i < 7; ++i) {
diff --git a/webrtc/modules/video_coding/codecs/vp8/include/vp8.h b/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
index f5dae471d2..dd3514235d 100644
--- a/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
+++ b/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
@@ -13,7 +13,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
@@ -21,16 +21,15 @@ class VP8Encoder : public VideoEncoder {
  public:
   static VP8Encoder* Create();
 
-  virtual ~VP8Encoder() {};
+  virtual ~VP8Encoder() {}
 };  // end of VP8Encoder class
 
-
 class VP8Decoder : public VideoDecoder {
  public:
   static VP8Decoder* Create();
 
-  virtual ~VP8Decoder() {};
+  virtual ~VP8Decoder() {}
 };  // end of VP8Decoder class
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
diff --git a/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h b/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
index c2cefdd94e..7a27e4429a 100644
--- a/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
+++ b/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
 
 #include "webrtc/common_types.h"
 
@@ -19,11 +19,11 @@ namespace webrtc {
 // Values as required for the VP8 codec (accumulating).
 static const float
     kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = {
-      {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
-      {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
-      {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
-      {0.25f, 0.4f, 0.6f, 1.0f}  // 4 layers {25%, 15%, 20%, 40%}
+        {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
+        {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
+        {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
+        {0.25f, 0.4f, 0.6f, 1.0f}  // 4 layers {25%, 15%, 20%, 40%}
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
diff --git a/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc b/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
index 15b5af9200..d22601358f 100644
--- a/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
@@ -12,7 +12,7 @@
 
 #include "vpx/vpx_encoder.h"
 #include "vpx/vp8cx.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
 
@@ -23,7 +23,8 @@ namespace webrtc {
 namespace {
 enum {
   kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
-                        VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF,
+                        VP8_EFLAG_NO_REF_GF |
+                        VP8_EFLAG_NO_REF_ARF,
 
   kTemporalUpdateGolden =
       VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
@@ -37,13 +38,15 @@ enum {
       kTemporalUpdateAltref | VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF,
 
   kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
-                        VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+                        VP8_EFLAG_NO_UPD_LAST |
+                        VP8_EFLAG_NO_UPD_ENTROPY,
 
   kTemporalUpdateNoneNoRefAltref = kTemporalUpdateNone | VP8_EFLAG_NO_REF_ARF,
 
   kTemporalUpdateNoneNoRefGoldenRefAltRef =
       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
-      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+      VP8_EFLAG_NO_UPD_LAST |
+      VP8_EFLAG_NO_UPD_ENTROPY,
 
   kTemporalUpdateGoldenWithoutDependencyRefAltRef =
       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
@@ -133,12 +136,14 @@ class RealTimeTemporalLayers : public TemporalLayers {
         layer_ids_length_ = sizeof(layer_ids) / sizeof(*layer_ids);
 
         static const int encode_flags[] = {
-          kTemporalUpdateLastAndGoldenRefAltRef,
-          kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateGoldenRefAltRef,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateGoldenRefAltRef,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateNone
-        };
+            kTemporalUpdateLastAndGoldenRefAltRef,
+            kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateGoldenRefAltRef,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateGoldenRefAltRef,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateNone};
         encode_flags_length_ = sizeof(encode_flags) / sizeof(*layer_ids);
         encode_flags_ = encode_flags;
 
@@ -153,12 +158,14 @@ class RealTimeTemporalLayers : public TemporalLayers {
         layer_ids_length_ = sizeof(layer_ids) / sizeof(*layer_ids);
 
         static const int encode_flags[] = {
-          kTemporalUpdateLastAndGoldenRefAltRef,
-          kTemporalUpdateNoneNoRefGoldenRefAltRef,
-          kTemporalUpdateGoldenWithoutDependencyRefAltRef, kTemporalUpdateNone,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateNone,
-          kTemporalUpdateGoldenRefAltRef, kTemporalUpdateNone
-        };
+            kTemporalUpdateLastAndGoldenRefAltRef,
+            kTemporalUpdateNoneNoRefGoldenRefAltRef,
+            kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+            kTemporalUpdateNone,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateNone,
+            kTemporalUpdateGoldenRefAltRef,
+            kTemporalUpdateNone};
         encode_flags_length_ = sizeof(encode_flags) / sizeof(*layer_ids);
         encode_flags_ = encode_flags;
 
@@ -172,8 +179,8 @@ class RealTimeTemporalLayers : public TemporalLayers {
         assert(false);
         return false;
     }
-    memcpy(
-        cfg->ts_layer_id, layer_ids_, sizeof(unsigned int) * layer_ids_length_);
+    memcpy(cfg->ts_layer_id, layer_ids_,
+           sizeof(unsigned int) * layer_ids_length_);
     return true;
   }
 
diff --git a/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc b/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc
index a922e35712..1838e32eb7 100644
--- a/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc
@@ -25,8 +25,7 @@ ReferencePictureSelection::ReferencePictureSelection()
       last_sent_ref_update_time_(0),
       established_ref_picture_id_(0),
       last_refresh_time_(0),
-      rtt_(0) {
-}
+      rtt_(0) {}
 
 void ReferencePictureSelection::Init() {
   update_golden_next_ = true;
@@ -62,7 +61,8 @@ bool ReferencePictureSelection::ReceivedSLI(uint32_t now_ts) {
   return send_refresh;
 }
 
-int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
+int ReferencePictureSelection::EncodeFlags(int picture_id,
+                                           bool send_refresh,
                                            uint32_t now_ts) {
   int flags = 0;
   // We can't refresh the decoder until we have established the key frame.
@@ -87,12 +87,12 @@ int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
       received_ack_) {
     flags |= VP8_EFLAG_NO_REF_LAST;  // Don't reference the last frame.
     if (update_golden_next_) {
-      flags |= VP8_EFLAG_FORCE_GF;  // Update the golden reference.
+      flags |= VP8_EFLAG_FORCE_GF;    // Update the golden reference.
       flags |= VP8_EFLAG_NO_UPD_ARF;  // Don't update alt-ref.
-      flags |= VP8_EFLAG_NO_REF_GF;  // Don't reference the golden frame.
+      flags |= VP8_EFLAG_NO_REF_GF;   // Don't reference the golden frame.
     } else {
-      flags |= VP8_EFLAG_FORCE_ARF;  // Update the alt-ref reference.
-      flags |= VP8_EFLAG_NO_UPD_GF;  // Don't update the golden frame.
+      flags |= VP8_EFLAG_FORCE_ARF;   // Update the alt-ref reference.
+      flags |= VP8_EFLAG_NO_UPD_GF;   // Don't update the golden frame.
       flags |= VP8_EFLAG_NO_REF_ARF;  // Don't reference the alt-ref frame.
     }
     last_sent_ref_picture_id_ = picture_id;
@@ -103,9 +103,9 @@ int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
     if (established_golden_)
       flags |= VP8_EFLAG_NO_REF_ARF;  // Don't reference the alt-ref frame.
     else
-      flags |= VP8_EFLAG_NO_REF_GF;   // Don't reference the golden frame.
-    flags |= VP8_EFLAG_NO_UPD_GF;  // Don't update the golden frame.
-    flags |= VP8_EFLAG_NO_UPD_ARF;  // Don't update the alt-ref frame.
+      flags |= VP8_EFLAG_NO_REF_GF;  // Don't reference the golden frame.
+    flags |= VP8_EFLAG_NO_UPD_GF;    // Don't update the golden frame.
+    flags |= VP8_EFLAG_NO_UPD_ARF;   // Don't update the alt-ref frame.
   }
   return flags;
 }
diff --git a/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc
index c6474e5bd1..742bb96e91 100644
--- a/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc
@@ -22,25 +22,19 @@ static const uint32_t kMinUpdateInterval = 10;
 // Should match the values set in reference_picture_selection.h
 static const int kRtt = 10;
 
-static const int kNoPropagationGolden    = VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_UPD_ARF;
-static const int kNoPropagationAltRef    = VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_UPD_ARF;
-static const int kPropagateGolden        = VP8_EFLAG_FORCE_GF |
-                                           VP8_EFLAG_NO_UPD_ARF |
-                                           VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_REF_LAST;
-static const int kPropagateAltRef        = VP8_EFLAG_FORCE_ARF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_REF_LAST;
-static const int kRefreshFromGolden      = VP8_EFLAG_NO_REF_LAST |
-                                           VP8_EFLAG_NO_REF_ARF;
-static const int kRefreshFromAltRef      = VP8_EFLAG_NO_REF_LAST |
-                                           VP8_EFLAG_NO_REF_GF;
-
+static const int kNoPropagationGolden =
+    VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF;
+static const int kNoPropagationAltRef =
+    VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF;
+static const int kPropagateGolden = VP8_EFLAG_FORCE_GF | VP8_EFLAG_NO_UPD_ARF |
+                                    VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_LAST;
+static const int kPropagateAltRef = VP8_EFLAG_FORCE_ARF | VP8_EFLAG_NO_UPD_GF |
+                                    VP8_EFLAG_NO_REF_ARF |
+                                    VP8_EFLAG_NO_REF_LAST;
+static const int kRefreshFromGolden =
+    VP8_EFLAG_NO_REF_LAST | VP8_EFLAG_NO_REF_ARF;
+static const int kRefreshFromAltRef =
+    VP8_EFLAG_NO_REF_LAST | VP8_EFLAG_NO_REF_GF;
 
 class TestRPS : public ::testing::Test {
  protected:
@@ -84,15 +78,15 @@ TEST_F(TestRPS, TestDecoderRefresh) {
   EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
   // Enough time have elapsed since the previous reference propagation, we will
   // therefore get both a refresh from golden and a propagation of alt-ref.
-  EXPECT_EQ(rps_.EncodeFlags(5, true, 90 * time), kRefreshFromGolden |
-            kPropagateAltRef);
+  EXPECT_EQ(rps_.EncodeFlags(5, true, 90 * time),
+            kRefreshFromGolden | kPropagateAltRef);
   rps_.ReceivedRPSI(5);
   time += kRtt + 1;
   // Enough time for a new refresh, but not enough time for a reference
   // propagation.
   EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
-  EXPECT_EQ(rps_.EncodeFlags(6, true, 90 * time), kRefreshFromAltRef |
-            kNoPropagationAltRef);
+  EXPECT_EQ(rps_.EncodeFlags(6, true, 90 * time),
+            kRefreshFromAltRef | kNoPropagationAltRef);
 }
 
 TEST_F(TestRPS, TestWrap) {
diff --git a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
index 0fbb2a6c40..536587a13e 100644
--- a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
@@ -11,10 +11,12 @@
 
 #include <stdlib.h>
 
+#include <algorithm>
+
 #include "webrtc/base/checks.h"
 #include "vpx/vpx_encoder.h"
 #include "vpx/vp8cx.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
@@ -188,7 +190,7 @@ void ScreenshareLayers::FrameEncoded(unsigned int size,
 }
 
 void ScreenshareLayers::PopulateCodecSpecific(bool base_layer_sync,
-                                              CodecSpecificInfoVP8 *vp8_info,
+                                              CodecSpecificInfoVP8* vp8_info,
                                               uint32_t timestamp) {
   int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
   if (number_of_temporal_layers_ == 1) {
diff --git a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
index 90a8b1b883..7628758209 100644
--- a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
+++ b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
@@ -15,7 +15,7 @@
 
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
index 628e336568..f31ed5e4d8 100644
--- a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
@@ -12,9 +12,9 @@
 #include "vpx/vpx_encoder.h"
 #include "vpx/vp8cx.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
-#include "webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h"
 
 using ::testing::_;
 using ::testing::NiceMock;
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
index 5dc4ac78f1..40e438f7e4 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
@@ -215,9 +215,7 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
     }
 
     VideoEncoder* encoder = factory_->Create();
-    ret = encoder->InitEncode(&stream_codec,
-                              number_of_cores,
-                              max_payload_size);
+    ret = encoder->InitEncode(&stream_codec, number_of_cores, max_payload_size);
     if (ret < 0) {
       Release();
       return ret;
@@ -284,35 +282,25 @@ int SimulcastEncoderAdapter::Encode(
     // scale it to match what the encoder expects (below).
     if ((dst_width == src_width && dst_height == src_height) ||
         input_image.IsZeroSize()) {
-      streaminfos_[stream_idx].encoder->Encode(input_image,
-                                               codec_specific_info,
+      streaminfos_[stream_idx].encoder->Encode(input_image, codec_specific_info,
                                                &stream_frame_types);
     } else {
       VideoFrame dst_frame;
       // Making sure that destination frame is of sufficient size.
       // Aligning stride values based on width.
-      dst_frame.CreateEmptyFrame(dst_width, dst_height,
-                                 dst_width, (dst_width + 1) / 2,
-                                 (dst_width + 1) / 2);
-      libyuv::I420Scale(input_image.buffer(kYPlane),
-                        input_image.stride(kYPlane),
-                        input_image.buffer(kUPlane),
-                        input_image.stride(kUPlane),
-                        input_image.buffer(kVPlane),
-                        input_image.stride(kVPlane),
-                        src_width, src_height,
-                        dst_frame.buffer(kYPlane),
-                        dst_frame.stride(kYPlane),
-                        dst_frame.buffer(kUPlane),
-                        dst_frame.stride(kUPlane),
-                        dst_frame.buffer(kVPlane),
-                        dst_frame.stride(kVPlane),
-                        dst_width, dst_height,
-                        libyuv::kFilterBilinear);
+      dst_frame.CreateEmptyFrame(dst_width, dst_height, dst_width,
+                                 (dst_width + 1) / 2, (dst_width + 1) / 2);
+      libyuv::I420Scale(
+          input_image.buffer(kYPlane), input_image.stride(kYPlane),
+          input_image.buffer(kUPlane), input_image.stride(kUPlane),
+          input_image.buffer(kVPlane), input_image.stride(kVPlane), src_width,
+          src_height, dst_frame.buffer(kYPlane), dst_frame.stride(kYPlane),
+          dst_frame.buffer(kUPlane), dst_frame.stride(kUPlane),
+          dst_frame.buffer(kVPlane), dst_frame.stride(kVPlane), dst_width,
+          dst_height, libyuv::kFilterBilinear);
       dst_frame.set_timestamp(input_image.timestamp());
       dst_frame.set_render_time_ms(input_image.render_time_ms());
-      streaminfos_[stream_idx].encoder->Encode(dst_frame,
-                                               codec_specific_info,
+      streaminfos_[stream_idx].encoder->Encode(dst_frame, codec_specific_info,
                                                &stream_frame_types);
     }
   }
@@ -426,16 +414,17 @@ uint32_t SimulcastEncoderAdapter::GetStreamBitrate(
     // current stream's |targetBitrate|, otherwise it's capped by |maxBitrate|.
     if (stream_idx < codec_.numberOfSimulcastStreams - 1) {
       unsigned int max_rate = codec_.simulcastStream[stream_idx].maxBitrate;
-      if (new_bitrate_kbit >= SumStreamTargetBitrate(stream_idx + 1, codec_) +
-          codec_.simulcastStream[stream_idx + 1].minBitrate) {
+      if (new_bitrate_kbit >=
+          SumStreamTargetBitrate(stream_idx + 1, codec_) +
+              codec_.simulcastStream[stream_idx + 1].minBitrate) {
         max_rate = codec_.simulcastStream[stream_idx].targetBitrate;
       }
       return std::min(new_bitrate_kbit - sum_target_lower_streams, max_rate);
     } else {
-        // For the highest stream (highest resolution), the |targetBitRate| and
-        // |maxBitrate| are not used. Any excess bitrate (above the targets of
-        // all lower streams) is given to this (highest resolution) stream.
-        return new_bitrate_kbit - sum_target_lower_streams;
+      // For the highest stream (highest resolution), the |targetBitRate| and
+      // |maxBitrate| are not used. Any excess bitrate (above the targets of
+      // all lower streams) is given to this (highest resolution) stream.
+      return new_bitrate_kbit - sum_target_lower_streams;
     }
   } else {
     // Not enough bitrate for this stream.
@@ -507,4 +496,11 @@ bool SimulcastEncoderAdapter::SupportsNativeHandle() const {
   return streaminfos_[0].encoder->SupportsNativeHandle();
 }
 
+const char* SimulcastEncoderAdapter::ImplementationName() const {
+  // We should not be calling this method before streaminfos_ are configured.
+  RTC_DCHECK(!streaminfos_.empty());
+  // TODO(pbos): Support multiple implementation names for different encoders.
+  return streaminfos_[0].encoder->ImplementationName();
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
index afec024abc..05a96c7336 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
@@ -59,6 +59,7 @@ class SimulcastEncoderAdapter : public VP8Encoder {
 
   int GetTargetFramerate() override;
   bool SupportsNativeHandle() const override;
+  const char* ImplementationName() const override;
 
  private:
   struct StreamInfo {
@@ -71,8 +72,8 @@ class SimulcastEncoderAdapter : public VP8Encoder {
           send_stream(true) {}
     StreamInfo(VideoEncoder* encoder,
                EncodedImageCallback* callback,
-               unsigned short width,
-               unsigned short height,
+               uint16_t width,
+               uint16_t height,
                bool send_stream)
         : encoder(encoder),
           callback(callback),
@@ -83,8 +84,8 @@ class SimulcastEncoderAdapter : public VP8Encoder {
     // Deleted by SimulcastEncoderAdapter::Release().
     VideoEncoder* encoder;
     EncodedImageCallback* callback;
-    unsigned short width;
-    unsigned short height;
+    uint16_t width;
+    uint16_t height;
     bool key_frame_request;
     bool send_stream;
   };
@@ -118,4 +119,3 @@ class SimulcastEncoderAdapter : public VP8Encoder {
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_ENCODER_ADAPTER_H_
-
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
index 218b5e2d1a..86b8e0b345 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
@@ -11,7 +11,7 @@
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h"
 #include "webrtc/modules/video_coding/codecs/vp8/vp8_factory.h"
@@ -27,12 +27,10 @@ static VP8Encoder* CreateTestEncoderAdapter() {
 class TestSimulcastEncoderAdapter : public TestVp8Simulcast {
  public:
   TestSimulcastEncoderAdapter()
-     : TestVp8Simulcast(CreateTestEncoderAdapter(),
-                        VP8Decoder::Create()) {}
+      : TestVp8Simulcast(CreateTestEncoderAdapter(), VP8Decoder::Create()) {}
+
  protected:
-  virtual void SetUp() {
-    TestVp8Simulcast::SetUp();
-  }
+  virtual void SetUp() { TestVp8Simulcast::SetUp(); }
   virtual void TearDown() {
     TestVp8Simulcast::TearDown();
     VP8EncoderFactoryConfig::set_use_simulcast_adapter(false);
@@ -97,8 +95,7 @@ TEST_F(TestSimulcastEncoderAdapter, TestSpatioTemporalLayers321PatternEncoder) {
 
 // TODO(ronghuawu): Enable this test when SkipEncodingUnusedStreams option is
 // implemented for SimulcastEncoderAdapter.
-TEST_F(TestSimulcastEncoderAdapter,
-    DISABLED_TestSkipEncodingUnusedStreams) {
+TEST_F(TestSimulcastEncoderAdapter, DISABLED_TestSkipEncodingUnusedStreams) {
   TestVp8Simulcast::TestSkipEncodingUnusedStreams();
 }
 
@@ -127,23 +124,17 @@ class MockVideoEncoder : public VideoEncoder {
     return 0;
   }
 
-  int32_t Release() override {
-    return 0;
-  }
+  int32_t Release() override { return 0; }
 
   int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override {
     return 0;
   }
 
-  MOCK_METHOD2(SetChannelParameters,
-      int32_t(uint32_t packetLoss, int64_t rtt));
+  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
 
-  bool SupportsNativeHandle() const override {
-    return supports_native_handle_;
-  }
+  bool SupportsNativeHandle() const override { return supports_native_handle_; }
 
-  virtual ~MockVideoEncoder() {
-  }
+  virtual ~MockVideoEncoder() {}
 
   const VideoCodec& codec() const { return codec_; }
 
@@ -200,7 +191,8 @@ class TestSimulcastEncoderAdapterFakeHelper {
     EXPECT_TRUE(!factory_->encoders().empty());
     for (size_t i = 0; i < factory_->encoders().size(); ++i) {
       EXPECT_CALL(*factory_->encoders()[i],
-                  SetChannelParameters(packetLoss, rtt)).Times(1);
+                  SetChannelParameters(packetLoss, rtt))
+          .Times(1);
     }
   }
 
@@ -249,8 +241,7 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
 
   void SetupCodec() {
     TestVp8Simulcast::DefaultSettings(
-      &codec_,
-      static_cast<const int*>(kTestTemporalLayerProfile));
+        &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
     EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
     adapter_->RegisterEncodeCompleteCallback(this);
   }
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
index 373a55237f..f23affee41 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
@@ -13,18 +13,14 @@
 namespace webrtc {
 namespace testing {
 
-class TestVp8Impl
-    : public TestVp8Simulcast {
+class TestVp8Impl : public TestVp8Simulcast {
  public:
   TestVp8Impl()
-     : TestVp8Simulcast(VP8Encoder::Create(), VP8Decoder::Create()) {}
+      : TestVp8Simulcast(VP8Encoder::Create(), VP8Decoder::Create()) {}
+
  protected:
-  virtual void SetUp() {
-    TestVp8Simulcast::SetUp();
-  }
-  virtual void TearDown() {
-    TestVp8Simulcast::TearDown();
-  }
+  virtual void SetUp() { TestVp8Simulcast::SetUp(); }
+  virtual void TearDown() { TestVp8Simulcast::TearDown(); }
 };
 
 TEST_F(TestVp8Impl, TestKeyFrameRequestsOnAllStreams) {
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
index e4fc986545..7a7a2c253b 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
@@ -14,10 +14,11 @@
 #include <algorithm>
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
 #include "webrtc/video_frame.h"
@@ -43,10 +44,8 @@ const int kMinBitrates[kNumberOfSimulcastStreams] = {50, 150, 600};
 const int kTargetBitrates[kNumberOfSimulcastStreams] = {100, 450, 1000};
 const int kDefaultTemporalLayerProfile[3] = {3, 3, 3};
 
-template<typename T> void SetExpectedValues3(T value0,
-                                             T value1,
-                                             T value2,
-                                             T* expected_values) {
+template <typename T>
+void SetExpectedValues3(T value0, T value1, T value2, T* expected_values) {
   expected_values[0] = value0;
   expected_values[1] = value1;
   expected_values[2] = value2;
@@ -54,15 +53,14 @@ template<typename T> void SetExpectedValues3(T value0,
 
 class Vp8TestEncodedImageCallback : public EncodedImageCallback {
  public:
-  Vp8TestEncodedImageCallback()
-       : picture_id_(-1) {
+  Vp8TestEncodedImageCallback() : picture_id_(-1) {
     memset(temporal_layer_, -1, sizeof(temporal_layer_));
     memset(layer_sync_, false, sizeof(layer_sync_));
   }
 
   ~Vp8TestEncodedImageCallback() {
-    delete [] encoded_key_frame_._buffer;
-    delete [] encoded_frame_._buffer;
+    delete[] encoded_key_frame_._buffer;
+    delete[] encoded_frame_._buffer;
   }
 
   virtual int32_t Encoded(const EncodedImage& encoded_image,
@@ -71,22 +69,20 @@ class Vp8TestEncodedImageCallback : public EncodedImageCallback {
     // Only store the base layer.
     if (codec_specific_info->codecSpecific.VP8.simulcastIdx == 0) {
       if (encoded_image._frameType == kVideoFrameKey) {
-        delete [] encoded_key_frame_._buffer;
+        delete[] encoded_key_frame_._buffer;
         encoded_key_frame_._buffer = new uint8_t[encoded_image._size];
         encoded_key_frame_._size = encoded_image._size;
         encoded_key_frame_._length = encoded_image._length;
         encoded_key_frame_._frameType = kVideoFrameKey;
         encoded_key_frame_._completeFrame = encoded_image._completeFrame;
-        memcpy(encoded_key_frame_._buffer,
-               encoded_image._buffer,
+        memcpy(encoded_key_frame_._buffer, encoded_image._buffer,
                encoded_image._length);
       } else {
-        delete [] encoded_frame_._buffer;
+        delete[] encoded_frame_._buffer;
         encoded_frame_._buffer = new uint8_t[encoded_image._size];
         encoded_frame_._size = encoded_image._size;
         encoded_frame_._length = encoded_image._length;
-        memcpy(encoded_frame_._buffer,
-               encoded_image._buffer,
+        memcpy(encoded_frame_._buffer, encoded_image._buffer,
                encoded_image._length);
       }
     }
@@ -97,8 +93,10 @@ class Vp8TestEncodedImageCallback : public EncodedImageCallback {
         codec_specific_info->codecSpecific.VP8.temporalIdx;
     return 0;
   }
-  void GetLastEncodedFrameInfo(int* picture_id, int* temporal_layer,
-                               bool* layer_sync, int stream) {
+  void GetLastEncodedFrameInfo(int* picture_id,
+                               int* temporal_layer,
+                               bool* layer_sync,
+                               int stream) {
     *picture_id = picture_id_;
     *temporal_layer = temporal_layer_[stream];
     *layer_sync = layer_sync_[stream];
@@ -120,10 +118,8 @@ class Vp8TestEncodedImageCallback : public EncodedImageCallback {
 
 class Vp8TestDecodedImageCallback : public DecodedImageCallback {
  public:
-  Vp8TestDecodedImageCallback()
-      : decoded_frames_(0) {
-  }
-  virtual int32_t Decoded(VideoFrame& decoded_image) {
+  Vp8TestDecodedImageCallback() : decoded_frames_(0) {}
+  int32_t Decoded(VideoFrame& decoded_image) override {
     for (int i = 0; i < decoded_image.width(); ++i) {
       EXPECT_NEAR(kColorY, decoded_image.buffer(kYPlane)[i], 1);
     }
@@ -136,9 +132,11 @@ class Vp8TestDecodedImageCallback : public DecodedImageCallback {
     decoded_frames_++;
     return 0;
   }
-  int DecodedFrames() {
-    return decoded_frames_;
+  int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override {
+    RTC_NOTREACHED();
+    return -1;
   }
+  int DecodedFrames() { return decoded_frames_; }
 
  private:
   int decoded_frames_;
@@ -161,8 +159,7 @@ class SkipEncodingUnusedStreamsTest {
     std::vector<unsigned int> configured_bitrates;
     for (std::vector<TemporalLayers*>::const_iterator it =
              spy_factory->spying_layers_.begin();
-         it != spy_factory->spying_layers_.end();
-         ++it) {
+         it != spy_factory->spying_layers_.end(); ++it) {
       configured_bitrates.push_back(
           static_cast<SpyingTemporalLayers*>(*it)->configured_bitrate_);
     }
@@ -185,8 +182,8 @@ class SkipEncodingUnusedStreamsTest {
                            int framerate,
                            vpx_codec_enc_cfg_t* cfg) override {
       configured_bitrate_ = bitrate_kbit;
-      return layers_->ConfigureBitrates(
-          bitrate_kbit, max_bitrate_kbit, framerate, cfg);
+      return layers_->ConfigureBitrates(bitrate_kbit, max_bitrate_kbit,
+                                        framerate, cfg);
     }
 
     void PopulateCodecSpecific(bool base_layer_sync,
@@ -228,16 +225,15 @@ class SkipEncodingUnusedStreamsTest {
 class TestVp8Simulcast : public ::testing::Test {
  public:
   TestVp8Simulcast(VP8Encoder* encoder, VP8Decoder* decoder)
-     : encoder_(encoder),
-       decoder_(decoder) {}
+      : encoder_(encoder), decoder_(decoder) {}
 
   // Creates an VideoFrame from |plane_colors|.
   static void CreateImage(VideoFrame* frame, int plane_colors[kNumOfPlanes]) {
     for (int plane_num = 0; plane_num < kNumOfPlanes; ++plane_num) {
-      int width = (plane_num != kYPlane ? (frame->width() + 1) / 2 :
-        frame->width());
-      int height = (plane_num != kYPlane ? (frame->height() + 1) / 2 :
-        frame->height());
+      int width =
+          (plane_num != kYPlane ? (frame->width() + 1) / 2 : frame->width());
+      int height =
+          (plane_num != kYPlane ? (frame->height() + 1) / 2 : frame->height());
       PlaneType plane_type = static_cast<PlaneType>(plane_num);
       uint8_t* data = frame->buffer(plane_type);
       // Setting allocated area to zero - setting only image size to
@@ -267,24 +263,15 @@ class TestVp8Simulcast : public ::testing::Test {
     settings->height = kDefaultHeight;
     settings->numberOfSimulcastStreams = kNumberOfSimulcastStreams;
     ASSERT_EQ(3, kNumberOfSimulcastStreams);
-    ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4,
-                    kMaxBitrates[0],
-                    kMinBitrates[0],
-                    kTargetBitrates[0],
-                    &settings->simulcastStream[0],
-                    temporal_layer_profile[0]);
-    ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2,
-                    kMaxBitrates[1],
-                    kMinBitrates[1],
-                    kTargetBitrates[1],
-                    &settings->simulcastStream[1],
-                    temporal_layer_profile[1]);
-    ConfigureStream(kDefaultWidth, kDefaultHeight,
-                    kMaxBitrates[2],
-                    kMinBitrates[2],
-                    kTargetBitrates[2],
-                    &settings->simulcastStream[2],
-                    temporal_layer_profile[2]);
+    ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, kMaxBitrates[0],
+                    kMinBitrates[0], kTargetBitrates[0],
+                    &settings->simulcastStream[0], temporal_layer_profile[0]);
+    ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, kMaxBitrates[1],
+                    kMinBitrates[1], kTargetBitrates[1],
+                    &settings->simulcastStream[1], temporal_layer_profile[1]);
+    ConfigureStream(kDefaultWidth, kDefaultHeight, kMaxBitrates[2],
+                    kMinBitrates[2], kTargetBitrates[2],
+                    &settings->simulcastStream[2], temporal_layer_profile[2]);
     settings->codecSpecific.VP8.resilience = kResilientStream;
     settings->codecSpecific.VP8.denoisingOn = true;
     settings->codecSpecific.VP8.errorConcealmentOn = false;
@@ -312,9 +299,7 @@ class TestVp8Simulcast : public ::testing::Test {
   }
 
  protected:
-  virtual void SetUp() {
-    SetUpCodec(kDefaultTemporalLayerProfile);
-  }
+  virtual void SetUp() { SetUpCodec(kDefaultTemporalLayerProfile); }
 
   virtual void SetUpCodec(const int* temporal_layer_profile) {
     encoder_->RegisterEncodeCompleteCallback(&encoder_callback_);
@@ -323,14 +308,14 @@ class TestVp8Simulcast : public ::testing::Test {
     EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
     EXPECT_EQ(0, decoder_->InitDecode(&settings_, 1));
     int half_width = (kDefaultWidth + 1) / 2;
-    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight,
-                                  kDefaultWidth, half_width, half_width);
+    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight, kDefaultWidth,
+                                  half_width, half_width);
     memset(input_frame_.buffer(kYPlane), 0,
-        input_frame_.allocated_size(kYPlane));
+           input_frame_.allocated_size(kYPlane));
     memset(input_frame_.buffer(kUPlane), 0,
-        input_frame_.allocated_size(kUPlane));
+           input_frame_.allocated_size(kUPlane));
     memset(input_frame_.buffer(kVPlane), 0,
-        input_frame_.allocated_size(kVPlane));
+           input_frame_.allocated_size(kVPlane));
   }
 
   virtual void TearDown() {
@@ -342,28 +327,34 @@ class TestVp8Simulcast : public ::testing::Test {
     ASSERT_GE(expected_video_streams, 0);
     ASSERT_LE(expected_video_streams, kNumberOfSimulcastStreams);
     if (expected_video_streams >= 1) {
-      EXPECT_CALL(encoder_callback_, Encoded(
-          AllOf(Field(&EncodedImage::_frameType, frame_type),
-                Field(&EncodedImage::_encodedWidth, kDefaultWidth / 4),
-                Field(&EncodedImage::_encodedHeight, kDefaultHeight / 4)), _, _)
-                  )
+      EXPECT_CALL(
+          encoder_callback_,
+          Encoded(
+              AllOf(Field(&EncodedImage::_frameType, frame_type),
+                    Field(&EncodedImage::_encodedWidth, kDefaultWidth / 4),
+                    Field(&EncodedImage::_encodedHeight, kDefaultHeight / 4)),
+              _, _))
           .Times(1)
           .WillRepeatedly(Return(0));
     }
     if (expected_video_streams >= 2) {
-      EXPECT_CALL(encoder_callback_, Encoded(
-          AllOf(Field(&EncodedImage::_frameType, frame_type),
-                Field(&EncodedImage::_encodedWidth, kDefaultWidth / 2),
-                Field(&EncodedImage::_encodedHeight, kDefaultHeight / 2)), _, _)
-                  )
+      EXPECT_CALL(
+          encoder_callback_,
+          Encoded(
+              AllOf(Field(&EncodedImage::_frameType, frame_type),
+                    Field(&EncodedImage::_encodedWidth, kDefaultWidth / 2),
+                    Field(&EncodedImage::_encodedHeight, kDefaultHeight / 2)),
+              _, _))
           .Times(1)
           .WillRepeatedly(Return(0));
     }
     if (expected_video_streams >= 3) {
-      EXPECT_CALL(encoder_callback_, Encoded(
-          AllOf(Field(&EncodedImage::_frameType, frame_type),
-                Field(&EncodedImage::_encodedWidth, kDefaultWidth),
-                Field(&EncodedImage::_encodedHeight, kDefaultHeight)), _, _))
+      EXPECT_CALL(
+          encoder_callback_,
+          Encoded(AllOf(Field(&EncodedImage::_frameType, frame_type),
+                        Field(&EncodedImage::_encodedWidth, kDefaultWidth),
+                        Field(&EncodedImage::_encodedHeight, kDefaultHeight)),
+                  _, _))
           .Times(1)
           .WillRepeatedly(Return(0));
     }
@@ -477,8 +468,8 @@ class TestVp8Simulcast : public ::testing::Test {
   void TestPaddingOneStreamTwoMaxedOut() {
     // We are just below limit of sending third stream, so we should get
     // first stream's rate maxed out at |targetBitrate|, second at |maxBitrate|.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2] - 1, 30);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] - 1, 30);
     std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
                                        kVideoFrameDelta);
     ExpectStreams(kVideoFrameKey, 2);
@@ -491,8 +482,8 @@ class TestVp8Simulcast : public ::testing::Test {
 
   void TestSendAllStreams() {
     // We have just enough to send all streams.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2], 30);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2], 30);
     std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
                                        kVideoFrameDelta);
     ExpectStreams(kVideoFrameKey, 3);
@@ -505,8 +496,7 @@ class TestVp8Simulcast : public ::testing::Test {
 
   void TestDisablingStreams() {
     // We should get three media streams.
-    encoder_->SetRates(kMaxBitrates[0] + kMaxBitrates[1] +
-                       kMaxBitrates[2], 30);
+    encoder_->SetRates(kMaxBitrates[0] + kMaxBitrates[1] + kMaxBitrates[2], 30);
     std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
                                        kVideoFrameDelta);
     ExpectStreams(kVideoFrameKey, 3);
@@ -517,8 +507,8 @@ class TestVp8Simulcast : public ::testing::Test {
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We should only get two streams and padding for one.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2] / 2, 30);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
     ExpectStreams(kVideoFrameDelta, 2);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -537,16 +527,16 @@ class TestVp8Simulcast : public ::testing::Test {
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We should only get two streams and padding for one.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2] / 2, 30);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
     // We get a key frame because a new stream is being enabled.
     ExpectStreams(kVideoFrameKey, 2);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We should get all three streams.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kTargetBitrates[2], 30);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kTargetBitrates[2], 30);
     // We get a key frame because a new stream is being enabled.
     ExpectStreams(kVideoFrameKey, 3);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
@@ -571,20 +561,20 @@ class TestVp8Simulcast : public ::testing::Test {
     input_frame_.CreateEmptyFrame(settings_.width, settings_.height,
                                   settings_.width, half_width, half_width);
     memset(input_frame_.buffer(kYPlane), 0,
-        input_frame_.allocated_size(kYPlane));
+           input_frame_.allocated_size(kYPlane));
     memset(input_frame_.buffer(kUPlane), 0,
-        input_frame_.allocated_size(kUPlane));
+           input_frame_.allocated_size(kUPlane));
     memset(input_frame_.buffer(kVPlane), 0,
-        input_frame_.allocated_size(kVPlane));
+           input_frame_.allocated_size(kVPlane));
 
     // The for loop above did not set the bitrate of the highest layer.
-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].
-        maxBitrate = 0;
+    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1]
+        .maxBitrate = 0;
     // The highest layer has to correspond to the non-simulcast resolution.
-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].
-        width = settings_.width;
-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].
-        height = settings_.height;
+    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].width =
+        settings_.width;
+    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].height =
+        settings_.height;
     EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
 
     // Encode one frame and verify.
@@ -612,21 +602,17 @@ class TestVp8Simulcast : public ::testing::Test {
     input_frame_.CreateEmptyFrame(settings_.width, settings_.height,
                                   settings_.width, half_width, half_width);
     memset(input_frame_.buffer(kYPlane), 0,
-        input_frame_.allocated_size(kYPlane));
+           input_frame_.allocated_size(kYPlane));
     memset(input_frame_.buffer(kUPlane), 0,
-        input_frame_.allocated_size(kUPlane));
+           input_frame_.allocated_size(kUPlane));
     memset(input_frame_.buffer(kVPlane), 0,
-        input_frame_.allocated_size(kVPlane));
+           input_frame_.allocated_size(kVPlane));
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
 
-  void TestSwitchingToOneStream() {
-    SwitchingToOneStream(1024, 768);
-  }
+  void TestSwitchingToOneStream() { SwitchingToOneStream(1024, 768); }
 
-  void TestSwitchingToOneOddStream() {
-    SwitchingToOneStream(1023, 769);
-  }
+  void TestSwitchingToOneOddStream() { SwitchingToOneStream(1023, 769); }
 
   void TestRPSIEncoder() {
     Vp8TestEncodedImageCallback encoder_callback;
@@ -777,67 +763,55 @@ class TestVp8Simulcast : public ::testing::Test {
     encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
     encoder_->SetRates(kMaxBitrates[2], 30);  // To get all three streams.
 
-    int expected_temporal_idx[3] = { -1, -1, -1};
+    int expected_temporal_idx[3] = {-1, -1, -1};
     bool expected_layer_sync[3] = {false, false, false};
 
     // First frame: #0.
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #1.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #2.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(1, 1, 1, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #3.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #4.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #5.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
   }
 
   // Test the layer pattern and sync flag for various spatial-temporal patterns.
@@ -858,67 +832,55 @@ class TestVp8Simulcast : public ::testing::Test {
     encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
     encoder_->SetRates(kMaxBitrates[2], 30);  // To get all three streams.
 
-    int expected_temporal_idx[3] = { -1, -1, -1};
+    int expected_temporal_idx[3] = {-1, -1, -1};
     bool expected_layer_sync[3] = {false, false, false};
 
     // First frame: #0.
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #1.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #2.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(1, 0, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(true, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #3.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #4.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #5.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
   }
 
   void TestStrideEncodeDecode() {
@@ -932,8 +894,8 @@ class TestVp8Simulcast : public ::testing::Test {
     // 1. stride > width 2. stride_y != stride_uv/2
     int stride_y = kDefaultWidth + 20;
     int stride_uv = ((kDefaultWidth + 1) / 2) + 5;
-    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight,
-                                 stride_y, stride_uv, stride_uv);
+    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight, stride_y,
+                                  stride_uv, stride_uv);
     // Set color.
     int plane_offset[kNumOfPlanes];
     plane_offset[kYPlane] = kColorY;
@@ -963,10 +925,9 @@ class TestVp8Simulcast : public ::testing::Test {
   void TestSkipEncodingUnusedStreams() {
     SkipEncodingUnusedStreamsTest test;
     std::vector<unsigned int> configured_bitrate =
-        test.RunTest(encoder_.get(),
-                     &settings_,
-                     1);    // Target bit rate 1, to force all streams but the
-                            // base one to be exceeding bandwidth constraints.
+        test.RunTest(encoder_.get(), &settings_,
+                     1);  // Target bit rate 1, to force all streams but the
+                          // base one to be exceeding bandwidth constraints.
     EXPECT_EQ(static_cast<size_t>(kNumberOfSimulcastStreams),
               configured_bitrate.size());
 
@@ -975,8 +936,7 @@ class TestVp8Simulcast : public ::testing::Test {
     int stream = 0;
     for (std::vector<unsigned int>::const_iterator it =
              configured_bitrate.begin();
-         it != configured_bitrate.end();
-         ++it) {
+         it != configured_bitrate.end(); ++it) {
       if (stream == 0) {
         EXPECT_EQ(min_bitrate, *it);
       } else {
diff --git a/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h b/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
index 7607210d5c..47112c64aa 100644
--- a/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
+++ b/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
@@ -14,7 +14,8 @@
 
 #include "vpx/vpx_encoder.h"
 
-#include "webrtc/common_video/interface/video_image.h"
+#include "webrtc/common.h"
+#include "webrtc/common_video/include/video_image.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -30,6 +31,8 @@ class TemporalLayers {
     virtual ~Factory() {}
     virtual TemporalLayers* Create(int temporal_layers,
                                    uint8_t initial_tl0_pic_idx) const;
+    static const ConfigOptionID identifier =
+        ConfigOptionID::kTemporalLayersFactory;
   };
 
   virtual ~TemporalLayers() {}
diff --git a/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 5ec674f16a..c3d77da063 100644
--- a/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -11,12 +11,12 @@
 #include <stdio.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -78,7 +78,11 @@ class Vp8UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback {
  public:
   explicit Vp8UnitTestDecodeCompleteCallback(VideoFrame* frame)
       : decoded_frame_(frame), decode_complete(false) {}
-  int Decoded(webrtc::VideoFrame& frame);
+  int32_t Decoded(VideoFrame& frame) override;
+  int32_t Decoded(VideoFrame& frame, int64_t decode_time_ms) override {
+    RTC_NOTREACHED();
+    return -1;
+  }
   bool DecodeComplete();
 
  private:
@@ -216,7 +220,12 @@ TEST_F(TestVp8Impl, EncoderParameterTest) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_inst_, 1));
 }
 
-TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AlignedStrideEncodeDecode DISABLED_AlignedStrideEncodeDecode
+#else
+#define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
+#endif
+TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
   SetUpEncodeDecode();
   encoder_->Encode(input_frame_, NULL, NULL);
   EXPECT_GT(WaitForEncodedFrame(), 0u);
@@ -232,7 +241,12 @@ TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
   EXPECT_EQ(kTestNtpTimeMs, decoded_frame_.ntp_time_ms());
 }
 
-TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(DecodeWithACompleteKeyFrame)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_DecodeWithACompleteKeyFrame DISABLED_DecodeWithACompleteKeyFrame
+#else
+#define MAYBE_DecodeWithACompleteKeyFrame DecodeWithACompleteKeyFrame
+#endif
+TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
   SetUpEncodeDecode();
   encoder_->Encode(input_frame_, NULL, NULL);
   EXPECT_GT(WaitForEncodedFrame(), 0u);
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h b/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h
index 84745ea5a1..52f8aa30b8 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h
@@ -32,4 +32,3 @@ class VP8EncoderFactoryConfig {
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_FACTORY_H_
-
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
index 029ccd1f27..5a04f6a43d 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
@@ -16,7 +16,7 @@
 #include <algorithm>
 
 // NOTE(ajm): Path provided by gyp.
-#include "libyuv/scale.h"  // NOLINT
+#include "libyuv/scale.h"    // NOLINT
 #include "libyuv/convert.h"  // NOLINT
 
 #include "webrtc/base/checks.h"
@@ -24,8 +24,8 @@
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
@@ -68,10 +68,9 @@ std::vector<int> GetStreamBitratesKbps(const VideoCodec& codec,
   std::vector<int> bitrates_kbps(codec.numberOfSimulcastStreams);
   // Allocate min -> target bitrates as long as we have bitrate to spend.
   size_t last_active_stream = 0;
-  for (size_t i = 0;
-       i < static_cast<size_t>(codec.numberOfSimulcastStreams) &&
-           bitrate_to_allocate_kbps >=
-               static_cast<int>(codec.simulcastStream[i].minBitrate);
+  for (size_t i = 0; i < static_cast<size_t>(codec.numberOfSimulcastStreams) &&
+                     bitrate_to_allocate_kbps >=
+                         static_cast<int>(codec.simulcastStream[i].minBitrate);
        ++i) {
     last_active_stream = i;
     int allocated_bitrate_kbps =
@@ -132,7 +131,7 @@ bool ValidSimulcastResolutions(const VideoCodec& codec, int num_streams) {
   return true;
 }
 
-int NumStreamsDisabled(std::vector<bool>& streams) {
+int NumStreamsDisabled(const std::vector<bool>& streams) {
   int num_disabled = 0;
   for (bool stream : streams) {
     if (!stream)
@@ -183,7 +182,7 @@ int VP8EncoderImpl::Release() {
 
   while (!encoded_images_.empty()) {
     EncodedImage& image = encoded_images_.back();
-    delete [] image._buffer;
+    delete[] image._buffer;
     encoded_images_.pop_back();
   }
   while (!encoders_.empty()) {
@@ -289,10 +288,8 @@ int VP8EncoderImpl::SetRates(uint32_t new_bitrate_kbit,
       target_bitrate = tl0_bitrate;
     }
     configurations_[i].rc_target_bitrate = target_bitrate;
-    temporal_layers_[stream_idx]->ConfigureBitrates(target_bitrate,
-                                                    max_bitrate,
-                                                    framerate,
-                                                    &configurations_[i]);
+    temporal_layers_[stream_idx]->ConfigureBitrates(
+        target_bitrate, max_bitrate, framerate, &configurations_[i]);
     if (vpx_codec_enc_config_set(&encoders_[i], &configurations_[i])) {
       return WEBRTC_VIDEO_CODEC_ERROR;
     }
@@ -301,6 +298,10 @@ int VP8EncoderImpl::SetRates(uint32_t new_bitrate_kbit,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
+const char* VP8EncoderImpl::ImplementationName() const {
+  return "libvpx";
+}
+
 void VP8EncoderImpl::SetStreamState(bool send_stream,
                                             int stream_idx) {
   if (send_stream && !send_stream_[stream_idx]) {
@@ -311,8 +312,8 @@ void VP8EncoderImpl::SetStreamState(bool send_stream,
 }
 
 void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
-                                                 int num_temporal_layers,
-                                                 const VideoCodec& codec) {
+                                         int num_temporal_layers,
+                                         const VideoCodec& codec) {
   const Config default_options;
   const TemporalLayers::Factory& tl_factory =
       (codec.extra_options ? codec.extra_options : &default_options)
@@ -330,15 +331,16 @@ void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
     for (int i = 0; i < num_streams; ++i) {
       // TODO(andresp): crash if layers is invalid.
       int layers = codec.simulcastStream[i].numberOfTemporalLayers;
-      if (layers < 1) layers = 1;
+      if (layers < 1)
+        layers = 1;
       temporal_layers_.push_back(tl_factory.Create(layers, rand()));
     }
   }
 }
 
 int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
-                                       int number_of_cores,
-                                       size_t /*maxPayloadSize */) {
+                               int number_of_cores,
+                               size_t /*maxPayloadSize */) {
   if (inst == NULL) {
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   }
@@ -375,12 +377,13 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   }
 
-  int num_temporal_layers = doing_simulcast ?
-      inst->simulcastStream[0].numberOfTemporalLayers :
-      inst->codecSpecific.VP8.numberOfTemporalLayers;
+  int num_temporal_layers =
+      doing_simulcast ? inst->simulcastStream[0].numberOfTemporalLayers
+                      : inst->codecSpecific.VP8.numberOfTemporalLayers;
 
   // TODO(andresp): crash if num temporal layers is bananas.
-  if (num_temporal_layers < 1) num_temporal_layers = 1;
+  if (num_temporal_layers < 1)
+    num_temporal_layers = 1;
   SetupTemporalLayers(number_of_streams, num_temporal_layers, *inst);
 
   feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
@@ -410,7 +413,7 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
   int idx = number_of_streams - 1;
   for (int i = 0; i < (number_of_streams - 1); ++i, --idx) {
     int gcd = GCD(inst->simulcastStream[idx].width,
-                  inst->simulcastStream[idx-1].width);
+                  inst->simulcastStream[idx - 1].width);
     downsampling_factors_[i].num = inst->simulcastStream[idx].width / gcd;
     downsampling_factors_[i].den = inst->simulcastStream[idx - 1].width / gcd;
     send_stream_[i] = false;
@@ -422,20 +425,20 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
   }
   for (int i = 0; i < number_of_streams; ++i) {
     // Random start, 16 bits is enough.
-    picture_id_[i] = static_cast<uint16_t>(rand()) & 0x7FFF;
+    picture_id_[i] = static_cast<uint16_t>(rand()) & 0x7FFF;  // NOLINT
     last_key_frame_picture_id_[i] = -1;
     // allocate memory for encoded image
     if (encoded_images_[i]._buffer != NULL) {
-      delete [] encoded_images_[i]._buffer;
+      delete[] encoded_images_[i]._buffer;
     }
-    encoded_images_[i]._size = CalcBufferSize(kI420,
-                                              codec_.width, codec_.height);
+    encoded_images_[i]._size =
+        CalcBufferSize(kI420, codec_.width, codec_.height);
     encoded_images_[i]._buffer = new uint8_t[encoded_images_[i]._size];
     encoded_images_[i]._completeFrame = true;
   }
   // populate encoder configuration with default values
-  if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(),
-                                   &configurations_[0], 0)) {
+  if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), &configurations_[0],
+                                   0)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
   // setting the time base of the codec
@@ -459,8 +462,8 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
       break;
     case kResilientFrames:
 #ifdef INDEPENDENT_PARTITIONS
-      configurations_[0]-g_error_resilient = VPX_ERROR_RESILIENT_DEFAULT |
-      VPX_ERROR_RESILIENT_PARTITIONS;
+      configurations_[0] - g_error_resilient =
+          VPX_ERROR_RESILIENT_DEFAULT | VPX_ERROR_RESILIENT_PARTITIONS;
       break;
 #else
       return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;  // Not supported
@@ -536,20 +539,18 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
 
   // Determine number of threads based on the image size and #cores.
   // TODO(fbarchard): Consider number of Simulcast layers.
-  configurations_[0].g_threads = NumberOfThreads(configurations_[0].g_w,
-                                                 configurations_[0].g_h,
-                                                 number_of_cores);
+  configurations_[0].g_threads = NumberOfThreads(
+      configurations_[0].g_w, configurations_[0].g_h, number_of_cores);
 
   // Creating a wrapper to the image - setting image data to NULL.
   // Actual pointer will be set in encode. Setting align to 1, as it
   // is meaningless (no memory allocation is done here).
-  vpx_img_wrap(&raw_images_[0], VPX_IMG_FMT_I420, inst->width, inst->height,
-               1, NULL);
+  vpx_img_wrap(&raw_images_[0], VPX_IMG_FMT_I420, inst->width, inst->height, 1,
+               NULL);
 
   if (encoders_.size() == 1) {
     configurations_[0].rc_target_bitrate = inst->startBitrate;
-    temporal_layers_[0]->ConfigureBitrates(inst->startBitrate,
-                                           inst->maxBitrate,
+    temporal_layers_[0]->ConfigureBitrates(inst->startBitrate, inst->maxBitrate,
                                            inst->maxFramerate,
                                            &configurations_[0]);
   } else {
@@ -641,20 +642,15 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
   flags |= VPX_CODEC_USE_OUTPUT_PARTITION;
 
   if (encoders_.size() > 1) {
-    int error = vpx_codec_enc_init_multi(&encoders_[0],
-                                 vpx_codec_vp8_cx(),
-                                 &configurations_[0],
-                                 encoders_.size(),
-                                 flags,
-                                 &downsampling_factors_[0]);
+    int error = vpx_codec_enc_init_multi(&encoders_[0], vpx_codec_vp8_cx(),
+                                         &configurations_[0], encoders_.size(),
+                                         flags, &downsampling_factors_[0]);
     if (error) {
       return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
     }
   } else {
-    if (vpx_codec_enc_init(&encoders_[0],
-                           vpx_codec_vp8_cx(),
-                           &configurations_[0],
-                           flags)) {
+    if (vpx_codec_enc_init(&encoders_[0], vpx_codec_vp8_cx(),
+                           &configurations_[0], flags)) {
       return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
     }
   }
@@ -671,13 +667,13 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
 #else
   denoiser_state = kDenoiserOnAdaptive;
 #endif
-  vpx_codec_control(&encoders_[0], VP8E_SET_NOISE_SENSITIVITY,
-                    codec_.codecSpecific.VP8.denoisingOn ?
-                    denoiser_state : kDenoiserOff);
+  vpx_codec_control(
+      &encoders_[0], VP8E_SET_NOISE_SENSITIVITY,
+      codec_.codecSpecific.VP8.denoisingOn ? denoiser_state : kDenoiserOff);
   if (encoders_.size() > 2) {
-    vpx_codec_control(&encoders_[1], VP8E_SET_NOISE_SENSITIVITY,
-                      codec_.codecSpecific.VP8.denoisingOn ?
-                      denoiser_state : kDenoiserOff);
+    vpx_codec_control(
+        &encoders_[1], VP8E_SET_NOISE_SENSITIVITY,
+        codec_.codecSpecific.VP8.denoisingOn ? denoiser_state : kDenoiserOff);
   }
   for (size_t i = 0; i < encoders_.size(); ++i) {
     // Allow more screen content to be detected as static.
@@ -710,14 +706,12 @@ uint32_t VP8EncoderImpl::MaxIntraTarget(uint32_t optimalBuffersize) {
 
   // Don't go below 3 times the per frame bandwidth.
   const uint32_t minIntraTh = 300;
-  return (targetPct < minIntraTh) ? minIntraTh: targetPct;
+  return (targetPct < minIntraTh) ? minIntraTh : targetPct;
 }
 
 int VP8EncoderImpl::Encode(const VideoFrame& frame,
                            const CodecSpecificInfo* codec_specific_info,
                            const std::vector<FrameType>* frame_types) {
-  TRACE_EVENT1("webrtc", "VP8::Encode", "timestamp", frame.timestamp());
-
   if (!inited_)
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   if (frame.IsZeroSize())
@@ -731,7 +725,7 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
       quality_scaler_enabled_ ? quality_scaler_.GetScaledFrame(frame) : frame;
 
   if (quality_scaler_enabled_ && (input_image.width() != codec_.width ||
-      input_image.height() != codec_.height)) {
+                                  input_image.height() != codec_.height)) {
     int ret = UpdateCodecFrameSize(input_image);
     if (ret < 0)
       return ret;
@@ -747,11 +741,11 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
   // Image in vpx_image_t format.
   // Input image is const. VP8's raw image is not defined as const.
   raw_images_[0].planes[VPX_PLANE_Y] =
-     const_cast<uint8_t*>(input_image.buffer(kYPlane));
+      const_cast<uint8_t*>(input_image.buffer(kYPlane));
   raw_images_[0].planes[VPX_PLANE_U] =
-     const_cast<uint8_t*>(input_image.buffer(kUPlane));
+      const_cast<uint8_t*>(input_image.buffer(kUPlane));
   raw_images_[0].planes[VPX_PLANE_V] =
-     const_cast<uint8_t*>(input_image.buffer(kVPlane));
+      const_cast<uint8_t*>(input_image.buffer(kVPlane));
 
   raw_images_[0].stride[VPX_PLANE_Y] = input_image.stride(kYPlane);
   raw_images_[0].stride[VPX_PLANE_U] = input_image.stride(kUPlane);
@@ -760,17 +754,17 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
   for (size_t i = 1; i < encoders_.size(); ++i) {
     // Scale the image down a number of times by downsampling factor
     libyuv::I420Scale(
-        raw_images_[i-1].planes[VPX_PLANE_Y],
-        raw_images_[i-1].stride[VPX_PLANE_Y],
-        raw_images_[i-1].planes[VPX_PLANE_U],
-        raw_images_[i-1].stride[VPX_PLANE_U],
-        raw_images_[i-1].planes[VPX_PLANE_V],
-        raw_images_[i-1].stride[VPX_PLANE_V],
-        raw_images_[i-1].d_w, raw_images_[i-1].d_h,
-        raw_images_[i].planes[VPX_PLANE_Y], raw_images_[i].stride[VPX_PLANE_Y],
-        raw_images_[i].planes[VPX_PLANE_U], raw_images_[i].stride[VPX_PLANE_U],
-        raw_images_[i].planes[VPX_PLANE_V], raw_images_[i].stride[VPX_PLANE_V],
-        raw_images_[i].d_w, raw_images_[i].d_h, libyuv::kFilterBilinear);
+        raw_images_[i - 1].planes[VPX_PLANE_Y],
+        raw_images_[i - 1].stride[VPX_PLANE_Y],
+        raw_images_[i - 1].planes[VPX_PLANE_U],
+        raw_images_[i - 1].stride[VPX_PLANE_U],
+        raw_images_[i - 1].planes[VPX_PLANE_V],
+        raw_images_[i - 1].stride[VPX_PLANE_V], raw_images_[i - 1].d_w,
+        raw_images_[i - 1].d_h, raw_images_[i].planes[VPX_PLANE_Y],
+        raw_images_[i].stride[VPX_PLANE_Y], raw_images_[i].planes[VPX_PLANE_U],
+        raw_images_[i].stride[VPX_PLANE_U], raw_images_[i].planes[VPX_PLANE_V],
+        raw_images_[i].stride[VPX_PLANE_V], raw_images_[i].d_w,
+        raw_images_[i].d_h, libyuv::kFilterBilinear);
   }
   vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
   for (size_t i = 0; i < encoders_.size(); ++i) {
@@ -805,8 +799,8 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
   if (send_key_frame) {
     // Adapt the size of the key frame when in screenshare with 1 temporal
     // layer.
-    if (encoders_.size() == 1 && codec_.mode == kScreensharing
-        && codec_.codecSpecific.VP8.numberOfTemporalLayers <= 1) {
+    if (encoders_.size() == 1 && codec_.mode == kScreensharing &&
+        codec_.codecSpecific.VP8.numberOfTemporalLayers <= 1) {
       const uint32_t forceKeyFrameIntraTh = 100;
       vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
                         forceKeyFrameIntraTh);
@@ -818,13 +812,12 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
     }
     std::fill(key_frame_request_.begin(), key_frame_request_.end(), false);
   } else if (codec_specific_info &&
-      codec_specific_info->codecType == kVideoCodecVP8) {
+             codec_specific_info->codecType == kVideoCodecVP8) {
     if (feedback_mode_) {
       // Handle RPSI and SLI messages and set up the appropriate encode flags.
       bool sendRefresh = false;
       if (codec_specific_info->codecSpecific.VP8.hasReceivedRPSI) {
-        rps_.ReceivedRPSI(
-            codec_specific_info->codecSpecific.VP8.pictureIdRPSI);
+        rps_.ReceivedRPSI(codec_specific_info->codecSpecific.VP8.pictureIdRPSI);
       }
       if (codec_specific_info->codecSpecific.VP8.hasReceivedSLI) {
         sendRefresh = rps_.ReceivedSLI(input_image.timestamp());
@@ -876,8 +869,7 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
     }
 
     vpx_codec_control(&encoders_[i], VP8E_SET_FRAME_FLAGS, flags[stream_idx]);
-    vpx_codec_control(&encoders_[i],
-                      VP8E_SET_TEMPORAL_LAYER_ID,
+    vpx_codec_control(&encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
                       temporal_layers_[stream_idx]->CurrentLayerId());
   }
   // TODO(holmer): Ideally the duration should be the timestamp diff of this
@@ -895,7 +887,7 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
   // Reset specific intra frame thresholds, following the key frame.
   if (send_key_frame) {
     vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
-        rc_max_intra_target_);
+                      rc_max_intra_target_);
   }
   if (error)
     return WEBRTC_VIDEO_CODEC_ERROR;
@@ -913,8 +905,7 @@ int VP8EncoderImpl::UpdateCodecFrameSize(const VideoFrame& input_image) {
     codec_.simulcastStream[0].height = input_image.height();
   }
   // Update the cpu_speed setting for resolution change.
-  vpx_codec_control(&(encoders_[0]),
-                    VP8E_SET_CPUUSED,
+  vpx_codec_control(&(encoders_[0]), VP8E_SET_CPUUSED,
                     SetCpuSpeed(codec_.width, codec_.height));
   raw_images_[0].w = codec_.width;
   raw_images_[0].h = codec_.height;
@@ -947,13 +938,12 @@ void VP8EncoderImpl::PopulateCodecSpecific(
   }
   vp8Info->simulcastIdx = stream_idx;
   vp8Info->keyIdx = kNoKeyIdx;  // TODO(hlundin) populate this
-  vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) ?
-      true : false;
+  vp8Info->nonReference =
+      (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) ? true : false;
   bool base_layer_sync_point = (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ||
-                                only_predicting_from_key_frame;
+                               only_predicting_from_key_frame;
   temporal_layers_[stream_idx]->PopulateCodecSpecific(base_layer_sync_point,
-                                                      vp8Info,
-                                                      timestamp);
+                                                      vp8Info, timestamp);
   // Prepare next.
   picture_id_[stream_idx] = (picture_id_[stream_idx] + 1) & 0x7FFF;
 }
@@ -966,27 +956,26 @@ int VP8EncoderImpl::GetEncodedPartitions(const VideoFrame& input_image,
   int stream_idx = static_cast<int>(encoders_.size()) - 1;
   int result = WEBRTC_VIDEO_CODEC_OK;
   for (size_t encoder_idx = 0; encoder_idx < encoders_.size();
-      ++encoder_idx, --stream_idx) {
+       ++encoder_idx, --stream_idx) {
     vpx_codec_iter_t iter = NULL;
     int part_idx = 0;
     encoded_images_[encoder_idx]._length = 0;
     encoded_images_[encoder_idx]._frameType = kVideoFrameDelta;
     RTPFragmentationHeader frag_info;
     // token_partitions_ is number of bits used.
-    frag_info.VerifyAndAllocateFragmentationHeader((1 << token_partitions_)
-                                                   + 1);
+    frag_info.VerifyAndAllocateFragmentationHeader((1 << token_partitions_) +
+                                                   1);
     CodecSpecificInfo codec_specific;
-    const vpx_codec_cx_pkt_t *pkt = NULL;
-    while ((pkt = vpx_codec_get_cx_data(&encoders_[encoder_idx],
-                                        &iter)) != NULL) {
+    const vpx_codec_cx_pkt_t* pkt = NULL;
+    while ((pkt = vpx_codec_get_cx_data(&encoders_[encoder_idx], &iter)) !=
+           NULL) {
       switch (pkt->kind) {
         case VPX_CODEC_CX_FRAME_PKT: {
           uint32_t length = encoded_images_[encoder_idx]._length;
           memcpy(&encoded_images_[encoder_idx]._buffer[length],
-                 pkt->data.frame.buf,
-                 pkt->data.frame.sz);
+                 pkt->data.frame.buf, pkt->data.frame.sz);
           frag_info.fragmentationOffset[part_idx] = length;
-          frag_info.fragmentationLength[part_idx] =  pkt->data.frame.sz;
+          frag_info.fragmentationLength[part_idx] = pkt->data.frame.sz;
           frag_info.fragmentationPlType[part_idx] = 0;  // not known here
           frag_info.fragmentationTimeDiff[part_idx] = 0;
           encoded_images_[encoder_idx]._length += pkt->data.frame.sz;
@@ -1063,7 +1052,6 @@ int VP8EncoderImpl::RegisterEncodeCompleteCallback(
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-
 VP8DecoderImpl::VP8DecoderImpl()
     : decode_complete_callback_(NULL),
       inited_(false),
@@ -1075,8 +1063,7 @@ VP8DecoderImpl::VP8DecoderImpl()
       propagation_cnt_(-1),
       last_frame_width_(0),
       last_frame_height_(0),
-      key_frame_required_(true) {
-}
+      key_frame_required_(true) {}
 
 VP8DecoderImpl::~VP8DecoderImpl() {
   inited_ = true;  // in order to do the actual release
@@ -1092,8 +1079,7 @@ int VP8DecoderImpl::Reset() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int VP8DecoderImpl::InitDecode(const VideoCodec* inst,
-                                       int number_of_cores) {
+int VP8DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
   int ret_val = Release();
   if (ret_val < 0) {
     return ret_val;
@@ -1104,12 +1090,12 @@ int VP8DecoderImpl::InitDecode(const VideoCodec* inst,
   if (inst && inst->codecType == kVideoCodecVP8) {
     feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
   }
-  vpx_codec_dec_cfg_t  cfg;
+  vpx_codec_dec_cfg_t cfg;
   // Setting number of threads to a constant value (1)
   cfg.threads = 1;
   cfg.h = cfg.w = 0;  // set after decode
 
-vpx_codec_flags_t flags = 0;
+  vpx_codec_flags_t flags = 0;
 #if !defined(WEBRTC_ARCH_ARM) && !defined(WEBRTC_ARCH_ARM64)
   flags = VPX_CODEC_USE_POSTPROC;
 #ifdef INDEPENDENT_PARTITIONS
@@ -1134,10 +1120,10 @@ vpx_codec_flags_t flags = 0;
 }
 
 int VP8DecoderImpl::Decode(const EncodedImage& input_image,
-                                   bool missing_frames,
-                                   const RTPFragmentationHeader* fragmentation,
-                                   const CodecSpecificInfo* codec_specific_info,
-                                   int64_t /*render_time_ms*/) {
+                           bool missing_frames,
+                           const RTPFragmentationHeader* fragmentation,
+                           const CodecSpecificInfo* codec_specific_info,
+                           int64_t /*render_time_ms*/) {
   if (!inited_) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
@@ -1188,9 +1174,9 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
     if (input_image._frameType == kVideoFrameKey &&
         input_image._completeFrame) {
       propagation_cnt_ = -1;
-    // Start count on first loss.
+      // Start count on first loss.
     } else if ((!input_image._completeFrame || missing_frames) &&
-        propagation_cnt_ == -1) {
+               propagation_cnt_ == -1) {
       propagation_cnt_ = 0;
     }
     if (propagation_cnt_ >= 0) {
@@ -1242,15 +1228,15 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
   if (input_image._frameType == kVideoFrameKey && input_image._buffer != NULL) {
     const uint32_t bytes_to_copy = input_image._length;
     if (last_keyframe_._size < bytes_to_copy) {
-      delete [] last_keyframe_._buffer;
+      delete[] last_keyframe_._buffer;
       last_keyframe_._buffer = NULL;
       last_keyframe_._size = 0;
     }
     uint8_t* temp_buffer = last_keyframe_._buffer;  // Save buffer ptr.
-    uint32_t temp_size = last_keyframe_._size;  // Save size.
-    last_keyframe_ = input_image;  // Shallow copy.
-    last_keyframe_._buffer = temp_buffer;  // Restore buffer ptr.
-    last_keyframe_._size = temp_size;  // Restore buffer size.
+    uint32_t temp_size = last_keyframe_._size;      // Save size.
+    last_keyframe_ = input_image;                   // Shallow copy.
+    last_keyframe_._buffer = temp_buffer;           // Restore buffer ptr.
+    last_keyframe_._size = temp_size;               // Restore buffer size.
     if (!last_keyframe_._buffer) {
       // Allocate memory.
       last_keyframe_._size = bytes_to_copy;
@@ -1300,7 +1286,8 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
     }
     if (picture_id > -1) {
       if (((reference_updates & VP8_GOLD_FRAME) ||
-          (reference_updates & VP8_ALTR_FRAME)) && !corrupted) {
+           (reference_updates & VP8_ALTR_FRAME)) &&
+          !corrupted) {
         decode_complete_callback_->ReceivedDecodedReferenceFrame(picture_id);
       }
       decode_complete_callback_->ReceivedDecodedFrame(picture_id);
@@ -1323,14 +1310,10 @@ int VP8DecoderImpl::DecodePartitions(
     const EncodedImage& input_image,
     const RTPFragmentationHeader* fragmentation) {
   for (int i = 0; i < fragmentation->fragmentationVectorSize; ++i) {
-    const uint8_t* partition = input_image._buffer +
-        fragmentation->fragmentationOffset[i];
-    const uint32_t partition_length =
-        fragmentation->fragmentationLength[i];
-    if (vpx_codec_decode(decoder_,
-                         partition,
-                         partition_length,
-                         0,
+    const uint8_t* partition =
+        input_image._buffer + fragmentation->fragmentationOffset[i];
+    const uint32_t partition_length = fragmentation->fragmentationLength[i];
+    if (vpx_codec_decode(decoder_, partition, partition_length, 0,
                          VPX_DL_REALTIME)) {
       return WEBRTC_VIDEO_CODEC_ERROR;
     }
@@ -1343,8 +1326,8 @@ int VP8DecoderImpl::DecodePartitions(
 }
 
 int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
-                                        uint32_t timestamp,
-                                        int64_t ntp_time_ms) {
+                                uint32_t timestamp,
+                                int64_t ntp_time_ms) {
   if (img == NULL) {
     // Decoder OK and NULL image => No show frame
     return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
@@ -1354,14 +1337,13 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
   // Allocate memory for decoded image.
   VideoFrame decoded_image(buffer_pool_.CreateBuffer(img->d_w, img->d_h),
                            timestamp, 0, kVideoRotation_0);
-  libyuv::I420Copy(
-      img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
-      img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
-      img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
-      decoded_image.buffer(kYPlane), decoded_image.stride(kYPlane),
-      decoded_image.buffer(kUPlane), decoded_image.stride(kUPlane),
-      decoded_image.buffer(kVPlane), decoded_image.stride(kVPlane),
-      img->d_w, img->d_h);
+  libyuv::I420Copy(img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
+                   img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
+                   img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
+                   decoded_image.buffer(kYPlane), decoded_image.stride(kYPlane),
+                   decoded_image.buffer(kUPlane), decoded_image.stride(kUPlane),
+                   decoded_image.buffer(kVPlane), decoded_image.stride(kVPlane),
+                   img->d_w, img->d_h);
   decoded_image.set_ntp_time_ms(ntp_time_ms);
   int ret = decode_complete_callback_->Decoded(decoded_image);
   if (ret != 0)
@@ -1380,7 +1362,7 @@ int VP8DecoderImpl::RegisterDecodeCompleteCallback(
 
 int VP8DecoderImpl::Release() {
   if (last_keyframe_._buffer != NULL) {
-    delete [] last_keyframe_._buffer;
+    delete[] last_keyframe_._buffer;
     last_keyframe_._buffer = NULL;
   }
   if (decoder_ != NULL) {
@@ -1400,15 +1382,19 @@ int VP8DecoderImpl::Release() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
+const char* VP8DecoderImpl::ImplementationName() const {
+  return "libvpx";
+}
+
 int VP8DecoderImpl::CopyReference(VP8DecoderImpl* copy) {
   // The type of frame to copy should be set in ref_frame_->frame_type
   // before the call to this function.
-  if (vpx_codec_control(decoder_, VP8_COPY_REFERENCE, ref_frame_)
-      != VPX_CODEC_OK) {
+  if (vpx_codec_control(decoder_, VP8_COPY_REFERENCE, ref_frame_) !=
+      VPX_CODEC_OK) {
     return -1;
   }
-  if (vpx_codec_control(copy->decoder_, VP8_SET_REFERENCE, ref_frame_)
-      != VPX_CODEC_OK) {
+  if (vpx_codec_control(copy->decoder_, VP8_SET_REFERENCE, ref_frame_) !=
+      VPX_CODEC_OK) {
     return -1;
   }
   return 0;
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
index ba14ed5841..9d5fb713a4 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
@@ -22,12 +22,12 @@
 #include "vpx/vp8cx.h"
 #include "vpx/vp8dx.h"
 
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
-#include "webrtc/modules/video_coding/utility/include/quality_scaler.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/quality_scaler.h"
 #include "webrtc/video_frame.h"
 
 namespace webrtc {
@@ -58,8 +58,11 @@ class VP8EncoderImpl : public VP8Encoder {
 
   void OnDroppedFrame() override {}
 
+  const char* ImplementationName() const override;
+
  private:
-  void SetupTemporalLayers(int num_streams, int num_temporal_layers,
+  void SetupTemporalLayers(int num_streams,
+                           int num_temporal_layers,
                            const VideoCodec& codec);
 
   // Set the cpu_speed setting for encoder based on resolution and/or platform.
@@ -126,15 +129,17 @@ class VP8DecoderImpl : public VP8Decoder {
   int InitDecode(const VideoCodec* inst, int number_of_cores) override;
 
   int Decode(const EncodedImage& input_image,
-                     bool missing_frames,
-                     const RTPFragmentationHeader* fragmentation,
-                     const CodecSpecificInfo* codec_specific_info,
-                     int64_t /*render_time_ms*/) override;
+             bool missing_frames,
+             const RTPFragmentationHeader* fragmentation,
+             const CodecSpecificInfo* codec_specific_info,
+             int64_t /*render_time_ms*/) override;
 
   int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
   int Release() override;
   int Reset() override;
 
+  const char* ImplementationName() const override;
+
  private:
   // Copy reference image from this _decoder to the _decoder in copyTo. Set
   // which frame type to copy in _refFrame->frame_type before the call to
@@ -165,4 +170,3 @@ class VP8DecoderImpl : public VP8Decoder {
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
-
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc b/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
index 5843d83fa7..9e546653db 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
@@ -1,4 +1,4 @@
- /*
+/*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
@@ -9,8 +9,9 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_video/interface/video_image.h"
+#include "webrtc/common_video/include/video_image.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
@@ -22,8 +23,7 @@
 class Vp8SequenceCoderEncodeCallback : public webrtc::EncodedImageCallback {
  public:
   explicit Vp8SequenceCoderEncodeCallback(FILE* encoded_file)
-      : encoded_file_(encoded_file),
-        encoded_bytes_(0) {}
+      : encoded_file_(encoded_file), encoded_bytes_(0) {}
   ~Vp8SequenceCoderEncodeCallback();
   int Encoded(const webrtc::EncodedImage& encoded_image,
               const webrtc::CodecSpecificInfo* codecSpecificInfo,
@@ -31,6 +31,7 @@ class Vp8SequenceCoderEncodeCallback : public webrtc::EncodedImageCallback {
   // Returns the encoded image.
   webrtc::EncodedImage encoded_image() { return encoded_image_; }
   size_t encoded_bytes() { return encoded_bytes_; }
+
  private:
   webrtc::EncodedImage encoded_image_;
   FILE* encoded_file_;
@@ -38,7 +39,7 @@ class Vp8SequenceCoderEncodeCallback : public webrtc::EncodedImageCallback {
 };
 
 Vp8SequenceCoderEncodeCallback::~Vp8SequenceCoderEncodeCallback() {
-  delete [] encoded_image_._buffer;
+  delete[] encoded_image_._buffer;
   encoded_image_._buffer = NULL;
 }
 int Vp8SequenceCoderEncodeCallback::Encoded(
@@ -46,7 +47,7 @@ int Vp8SequenceCoderEncodeCallback::Encoded(
     const webrtc::CodecSpecificInfo* codecSpecificInfo,
     const webrtc::RTPFragmentationHeader* fragmentation) {
   if (encoded_image_._size < encoded_image._size) {
-    delete [] encoded_image_._buffer;
+    delete[] encoded_image_._buffer;
     encoded_image_._buffer = NULL;
     encoded_image_._buffer = new uint8_t[encoded_image._size];
     encoded_image_._size = encoded_image._size;
@@ -68,7 +69,11 @@ class Vp8SequenceCoderDecodeCallback : public webrtc::DecodedImageCallback {
  public:
   explicit Vp8SequenceCoderDecodeCallback(FILE* decoded_file)
       : decoded_file_(decoded_file) {}
-  int Decoded(webrtc::VideoFrame& frame);
+  int32_t Decoded(webrtc::VideoFrame& frame) override;
+  int32_t Decoded(webrtc::VideoFrame& frame, int64_t decode_time_ms) override {
+    RTC_NOTREACHED();
+    return -1;
+  }
   bool DecodeComplete();
 
  private:
@@ -80,16 +85,16 @@ int Vp8SequenceCoderDecodeCallback::Decoded(webrtc::VideoFrame& image) {
   return 0;
 }
 
-int SequenceCoder(webrtc::test::CommandLineParser& parser) {
-  int width = strtol((parser.GetFlag("w")).c_str(), NULL, 10);
-  int height = strtol((parser.GetFlag("h")).c_str(), NULL, 10);
-  int framerate = strtol((parser.GetFlag("f")).c_str(), NULL, 10);
+int SequenceCoder(webrtc::test::CommandLineParser* parser) {
+  int width = strtol((parser->GetFlag("w")).c_str(), NULL, 10);
+  int height = strtol((parser->GetFlag("h")).c_str(), NULL, 10);
+  int framerate = strtol((parser->GetFlag("f")).c_str(), NULL, 10);
 
   if (width <= 0 || height <= 0 || framerate <= 0) {
     fprintf(stderr, "Error: Resolution cannot be <= 0!\n");
     return -1;
   }
-  int target_bitrate = strtol((parser.GetFlag("b")).c_str(), NULL, 10);
+  int target_bitrate = strtol((parser->GetFlag("b")).c_str(), NULL, 10);
   if (target_bitrate <= 0) {
     fprintf(stderr, "Error: Bit-rate cannot be <= 0!\n");
     return -1;
@@ -97,20 +102,20 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
 
   // SetUp
   // Open input file.
-  std::string encoded_file_name = parser.GetFlag("encoded_file");
+  std::string encoded_file_name = parser->GetFlag("encoded_file");
   FILE* encoded_file = fopen(encoded_file_name.c_str(), "wb");
   if (encoded_file == NULL) {
     fprintf(stderr, "Error: Cannot open encoded file\n");
     return -1;
   }
-  std::string input_file_name = parser.GetFlag("input_file");
+  std::string input_file_name = parser->GetFlag("input_file");
   FILE* input_file = fopen(input_file_name.c_str(), "rb");
   if (input_file == NULL) {
     fprintf(stderr, "Error: Cannot open input file\n");
     return -1;
   }
   // Open output file.
-  std::string output_file_name = parser.GetFlag("output_file");
+  std::string output_file_name = parser->GetFlag("output_file");
   FILE* output_file = fopen(output_file_name.c_str(), "wb");
   if (output_file == NULL) {
     fprintf(stderr, "Error: Cannot open output file\n");
@@ -118,8 +123,8 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
   }
 
   // Get range of frames: will encode num_frames following start_frame).
-  int start_frame = strtol((parser.GetFlag("start_frame")).c_str(), NULL, 10);
-  int num_frames = strtol((parser.GetFlag("num_frames")).c_str(), NULL, 10);
+  int start_frame = strtol((parser->GetFlag("start_frame")).c_str(), NULL, 10);
+  int num_frames = strtol((parser->GetFlag("num_frames")).c_str(), NULL, 10);
 
   // Codec SetUp.
   webrtc::VideoCodec inst;
@@ -157,8 +162,8 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
   int frames_processed = 0;
   input_frame.CreateEmptyFrame(width, height, width, half_width, half_width);
   while (!feof(input_file) &&
-      (num_frames == -1 || frames_processed < num_frames)) {
-     if (fread(frame_buffer.get(), 1, length, input_file) != length)
+         (num_frames == -1 || frames_processed < num_frames)) {
+    if (fread(frame_buffer.get(), 1, length, input_file) != length)
       continue;
     if (frame_cnt >= start_frame) {
       webrtc::ConvertToI420(webrtc::kI420, frame_buffer.get(), 0, 0, width,
@@ -179,33 +184,35 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
   printf("Actual bitrate: %f kbps\n", actual_bit_rate / 1000);
   webrtc::test::QualityMetricsResult psnr_result, ssim_result;
   EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
-      input_file_name.c_str(), output_file_name.c_str(),
-      inst.width, inst.height,
-      &psnr_result, &ssim_result));
+                   input_file_name.c_str(), output_file_name.c_str(),
+                   inst.width, inst.height, &psnr_result, &ssim_result));
   printf("PSNR avg: %f[dB], min: %f[dB]\nSSIM avg: %f, min: %f\n",
-          psnr_result.average, psnr_result.min,
-          ssim_result.average, ssim_result.min);
+         psnr_result.average, psnr_result.min, ssim_result.average,
+         ssim_result.min);
   return frame_cnt;
 }
 
 int main(int argc, char** argv) {
   std::string program_name = argv[0];
-  std::string usage = "Encode and decodes a video sequence, and writes"
-  "results to a file.\n"
-  "Example usage:\n" + program_name + " functionality"
-  " --w=352 --h=288 --input_file=input.yuv --output_file=output.yuv "
-  " Command line flags:\n"
-  "  - width(int): The width of the input file. Default: 352\n"
-  "  - height(int): The height of the input file. Default: 288\n"
-  "  - input_file(string): The YUV file to encode."
-  "      Default: foreman.yuv\n"
-  "  - encoded_file(string): The vp8 encoded file (encoder output)."
-  "      Default: vp8_encoded.vp8\n"
-  "  - output_file(string): The yuv decoded file (decoder output)."
-  "      Default: vp8_decoded.yuv\n."
-  "  - start_frame - frame number in which encoding will begin. Default: 0"
-  "  - num_frames - Number of frames to be processed. "
-  "      Default: -1 (entire sequence).";
+  std::string usage =
+      "Encode and decodes a video sequence, and writes"
+      "results to a file.\n"
+      "Example usage:\n" +
+      program_name +
+      " functionality"
+      " --w=352 --h=288 --input_file=input.yuv --output_file=output.yuv "
+      " Command line flags:\n"
+      "  - width(int): The width of the input file. Default: 352\n"
+      "  - height(int): The height of the input file. Default: 288\n"
+      "  - input_file(string): The YUV file to encode."
+      "      Default: foreman.yuv\n"
+      "  - encoded_file(string): The vp8 encoded file (encoder output)."
+      "      Default: vp8_encoded.vp8\n"
+      "  - output_file(string): The yuv decoded file (decoder output)."
+      "      Default: vp8_decoded.yuv\n."
+      "  - start_frame - frame number in which encoding will begin. Default: 0"
+      "  - num_frames - Number of frames to be processed. "
+      "      Default: -1 (entire sequence).";
 
   webrtc::test::CommandLineParser parser;
 
@@ -223,8 +230,8 @@ int main(int argc, char** argv) {
   parser.SetFlag("output_file", webrtc::test::OutputPath() + "vp8_decoded.yuv");
   parser.SetFlag("encoded_file",
                  webrtc::test::OutputPath() + "vp8_encoded.vp8");
-  parser.SetFlag("input_file", webrtc::test::ResourcePath("foreman_cif",
-                                                          "yuv"));
+  parser.SetFlag("input_file",
+                 webrtc::test::ResourcePath("foreman_cif", "yuv"));
   parser.SetFlag("help", "false");
 
   parser.ProcessFlags();
@@ -234,5 +241,5 @@ int main(int argc, char** argv) {
   }
   parser.PrintEnteredFlags();
 
-  return SequenceCoder(parser);
+  return SequenceCoder(&parser);
 }
diff --git a/webrtc/modules/video_coding/codecs/vp9/include/vp9.h b/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
index cd77f72dcb..3bcbe46b3a 100644
--- a/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
+++ b/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
@@ -12,7 +12,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
@@ -23,7 +23,6 @@ class VP9Encoder : public VideoEncoder {
   virtual ~VP9Encoder() {}
 };
 
-
 class VP9Decoder : public VideoDecoder {
  public:
   static VP9Decoder* Create();
diff --git a/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc b/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
new file mode 100644
index 0000000000..c7ed78a192
--- /dev/null
+++ b/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
@@ -0,0 +1,93 @@
+/* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+*
+*  Use of this source code is governed by a BSD-style license
+*  that can be found in the LICENSE file in the root of the source
+*  tree. An additional intellectual property rights grant can be found
+*  in the file PATENTS.  All contributing project authors may
+*  be found in the AUTHORS file in the root of the source tree.
+*/
+
+#include <algorithm>
+#include "webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h"
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+ScreenshareLayersVP9::ScreenshareLayersVP9(uint8_t num_layers)
+    : num_layers_(num_layers),
+      start_layer_(0),
+      last_timestamp_(0),
+      timestamp_initialized_(false) {
+  RTC_DCHECK_GT(num_layers, 0);
+  RTC_DCHECK_LE(num_layers, kMaxVp9NumberOfSpatialLayers);
+  memset(bits_used_, 0, sizeof(bits_used_));
+  memset(threshold_kbps_, 0, sizeof(threshold_kbps_));
+}
+
+uint8_t ScreenshareLayersVP9::GetStartLayer() const {
+  return start_layer_;
+}
+
+void ScreenshareLayersVP9::ConfigureBitrate(int threshold_kbps,
+                                            uint8_t layer_id) {
+  // The upper layer is always the layer we spill frames
+  // to when the bitrate becomes to high, therefore setting
+  // a max limit is not allowed. The top layer bitrate is
+  // never used either so configuring it makes no difference.
+  RTC_DCHECK_LT(layer_id, num_layers_ - 1);
+  threshold_kbps_[layer_id] = threshold_kbps;
+}
+
+void ScreenshareLayersVP9::LayerFrameEncoded(unsigned int size_bytes,
+                                             uint8_t layer_id) {
+  RTC_DCHECK_LT(layer_id, num_layers_);
+  bits_used_[layer_id] += size_bytes * 8;
+}
+
+VP9EncoderImpl::SuperFrameRefSettings
+ScreenshareLayersVP9::GetSuperFrameSettings(uint32_t timestamp,
+                                            bool is_keyframe) {
+  VP9EncoderImpl::SuperFrameRefSettings settings;
+  if (!timestamp_initialized_) {
+    last_timestamp_ = timestamp;
+    timestamp_initialized_ = true;
+  }
+  float time_diff = (timestamp - last_timestamp_) / 90.f;
+  float total_bits_used = 0;
+  float total_threshold_kbps = 0;
+  start_layer_ = 0;
+
+  // Up to (num_layers - 1) because we only have
+  // (num_layers - 1) thresholds to check.
+  for (int layer_id = 0; layer_id < num_layers_ - 1; ++layer_id) {
+    bits_used_[layer_id] = std::max(
+        0.f, bits_used_[layer_id] - time_diff * threshold_kbps_[layer_id]);
+    total_bits_used += bits_used_[layer_id];
+    total_threshold_kbps += threshold_kbps_[layer_id];
+
+    // If this is a keyframe then there should be no
+    // references to any previous frames.
+    if (!is_keyframe) {
+      settings.layer[layer_id].ref_buf1 = layer_id;
+      if (total_bits_used > total_threshold_kbps * 1000)
+        start_layer_ = layer_id + 1;
+    }
+
+    settings.layer[layer_id].upd_buf = layer_id;
+  }
+  // Since the above loop does not iterate over the last layer
+  // the reference of the last layer has to be set after the loop,
+  // and if this is a keyframe there should be no references to
+  // any previous frames.
+  if (!is_keyframe)
+    settings.layer[num_layers_ - 1].ref_buf1 = num_layers_ - 1;
+
+  settings.layer[num_layers_ - 1].upd_buf = num_layers_ - 1;
+  settings.is_keyframe = is_keyframe;
+  settings.start_layer = start_layer_;
+  settings.stop_layer = num_layers_ - 1;
+  last_timestamp_ = timestamp;
+  return settings;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h b/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h
new file mode 100644
index 0000000000..5a901ae359
--- /dev/null
+++ b/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h
@@ -0,0 +1,66 @@
+/* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+*
+*  Use of this source code is governed by a BSD-style license
+*  that can be found in the LICENSE file in the root of the source
+*  tree. An additional intellectual property rights grant can be found
+*  in the file PATENTS.  All contributing project authors may
+*  be found in the AUTHORS file in the root of the source tree.
+*/
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_SCREENSHARE_LAYERS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_SCREENSHARE_LAYERS_H_
+
+#include "webrtc/modules/video_coding/codecs/vp9/vp9_impl.h"
+
+namespace webrtc {
+
+class ScreenshareLayersVP9 {
+ public:
+  explicit ScreenshareLayersVP9(uint8_t num_layers);
+
+  // The target bitrate for layer with id layer_id.
+  void ConfigureBitrate(int threshold_kbps, uint8_t layer_id);
+
+  // The current start layer.
+  uint8_t GetStartLayer() const;
+
+  // Update the layer with the size of the layer frame.
+  void LayerFrameEncoded(unsigned int size_bytes, uint8_t layer_id);
+
+  // Get the layer settings for the next superframe.
+  //
+  // In short, each time the GetSuperFrameSettings is called the
+  // bitrate of every layer is calculated and if the cummulative
+  // bitrate exceeds the configured cummulative bitrates
+  // (ConfigureBitrate to configure) up to and including that
+  // layer then the resulting encoding settings for the
+  // superframe will only encode layers above that layer.
+  VP9EncoderImpl::SuperFrameRefSettings GetSuperFrameSettings(
+      uint32_t timestamp,
+      bool is_keyframe);
+
+ private:
+  // How many layers that are used.
+  uint8_t num_layers_;
+
+  // The index of the first layer to encode.
+  uint8_t start_layer_;
+
+  // Cummulative target kbps for the different layers.
+  float threshold_kbps_[kMaxVp9NumberOfSpatialLayers - 1];
+
+  // How many bits that has been used for a certain layer. Increased in
+  // FrameEncoded() by the size of the encoded frame and decreased in
+  // GetSuperFrameSettings() depending on the time between frames.
+  float bits_used_[kMaxVp9NumberOfSpatialLayers];
+
+  // Timestamp of last frame.
+  uint32_t last_timestamp_;
+
+  // If the last_timestamp_ has been set.
+  bool timestamp_initialized_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_SCREENSHARE_LAYERS_H_
diff --git a/webrtc/modules/video_coding/codecs/vp9/screenshare_layers_unittest.cc b/webrtc/modules/video_coding/codecs/vp9/screenshare_layers_unittest.cc
new file mode 100644
index 0000000000..5eb7b237ac
--- /dev/null
+++ b/webrtc/modules/video_coding/codecs/vp9/screenshare_layers_unittest.cc
@@ -0,0 +1,323 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "vpx/vp8cx.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h"
+#include "webrtc/modules/video_coding/codecs/vp9/vp9_impl.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+typedef VP9EncoderImpl::SuperFrameRefSettings Settings;
+
+const uint32_t kTickFrequency = 90000;
+
+class ScreenshareLayerTestVP9 : public ::testing::Test {
+ protected:
+  ScreenshareLayerTestVP9() : clock_(0) {}
+  virtual ~ScreenshareLayerTestVP9() {}
+
+  void InitScreenshareLayers(int layers) {
+    layers_.reset(new ScreenshareLayersVP9(layers));
+  }
+
+  void ConfigureBitrateForLayer(int kbps, uint8_t layer_id) {
+    layers_->ConfigureBitrate(kbps, layer_id);
+  }
+
+  void AdvanceTime(int64_t milliseconds) {
+    clock_.AdvanceTimeMilliseconds(milliseconds);
+  }
+
+  void AddKilobitsToLayer(int kilobits, uint8_t layer_id) {
+    layers_->LayerFrameEncoded(kilobits * 1000 / 8, layer_id);
+  }
+
+  void EqualRefsForLayer(const Settings& actual, uint8_t layer_id) {
+    EXPECT_EQ(expected_.layer[layer_id].upd_buf,
+              actual.layer[layer_id].upd_buf);
+    EXPECT_EQ(expected_.layer[layer_id].ref_buf1,
+              actual.layer[layer_id].ref_buf1);
+    EXPECT_EQ(expected_.layer[layer_id].ref_buf2,
+              actual.layer[layer_id].ref_buf2);
+    EXPECT_EQ(expected_.layer[layer_id].ref_buf3,
+              actual.layer[layer_id].ref_buf3);
+  }
+
+  void EqualRefs(const Settings& actual) {
+    for (unsigned int layer_id = 0; layer_id < kMaxVp9NumberOfSpatialLayers;
+         ++layer_id) {
+      EqualRefsForLayer(actual, layer_id);
+    }
+  }
+
+  void EqualStartStopKeyframe(const Settings& actual) {
+    EXPECT_EQ(expected_.start_layer, actual.start_layer);
+    EXPECT_EQ(expected_.stop_layer, actual.stop_layer);
+    EXPECT_EQ(expected_.is_keyframe, actual.is_keyframe);
+  }
+
+  // Check that the settings returned by GetSuperFrameSettings() is
+  // equal to the expected_ settings.
+  void EqualToExpected() {
+    uint32_t frame_timestamp_ =
+        clock_.TimeInMilliseconds() * (kTickFrequency / 1000);
+    Settings actual =
+        layers_->GetSuperFrameSettings(frame_timestamp_, expected_.is_keyframe);
+    EqualRefs(actual);
+    EqualStartStopKeyframe(actual);
+  }
+
+  Settings expected_;
+  SimulatedClock clock_;
+  rtc::scoped_ptr<ScreenshareLayersVP9> layers_;
+};
+
+TEST_F(ScreenshareLayerTestVP9, NoRefsOnKeyFrame) {
+  const int kNumLayers = kMaxVp9NumberOfSpatialLayers;
+  InitScreenshareLayers(kNumLayers);
+  expected_.start_layer = 0;
+  expected_.stop_layer = kNumLayers - 1;
+
+  for (int l = 0; l < kNumLayers; ++l) {
+    expected_.layer[l].upd_buf = l;
+  }
+  expected_.is_keyframe = true;
+  EqualToExpected();
+
+  for (int l = 0; l < kNumLayers; ++l) {
+    expected_.layer[l].ref_buf1 = l;
+  }
+  expected_.is_keyframe = false;
+  EqualToExpected();
+}
+
+// Test if it is possible to send at a high bitrate (over the threshold)
+// after a longer period of low bitrate. This should not be possible.
+TEST_F(ScreenshareLayerTestVP9, DontAccumelateAvailableBitsOverTime) {
+  InitScreenshareLayers(2);
+  ConfigureBitrateForLayer(100, 0);
+
+  expected_.layer[0].upd_buf = 0;
+  expected_.layer[0].ref_buf1 = 0;
+  expected_.layer[1].upd_buf = 1;
+  expected_.layer[1].ref_buf1 = 1;
+  expected_.start_layer = 0;
+  expected_.stop_layer = 1;
+
+  // Send 10 frames at a low bitrate (50 kbps)
+  for (int i = 0; i < 10; ++i) {
+    AdvanceTime(200);
+    EqualToExpected();
+    AddKilobitsToLayer(10, 0);
+  }
+
+  AdvanceTime(200);
+  EqualToExpected();
+  AddKilobitsToLayer(301, 0);
+
+  // Send 10 frames at a high bitrate (200 kbps)
+  expected_.start_layer = 1;
+  for (int i = 0; i < 10; ++i) {
+    AdvanceTime(200);
+    EqualToExpected();
+    AddKilobitsToLayer(40, 1);
+  }
+}
+
+// Test if used bits are accumelated over layers, as they should;
+TEST_F(ScreenshareLayerTestVP9, AccumelateUsedBitsOverLayers) {
+  const int kNumLayers = kMaxVp9NumberOfSpatialLayers;
+  InitScreenshareLayers(kNumLayers);
+  for (int l = 0; l < kNumLayers - 1; ++l)
+    ConfigureBitrateForLayer(100, l);
+  for (int l = 0; l < kNumLayers; ++l) {
+    expected_.layer[l].upd_buf = l;
+    expected_.layer[l].ref_buf1 = l;
+  }
+
+  expected_.start_layer = 0;
+  expected_.stop_layer = kNumLayers - 1;
+  EqualToExpected();
+
+  for (int layer = 0; layer < kNumLayers - 1; ++layer) {
+    expected_.start_layer = layer;
+    EqualToExpected();
+    AddKilobitsToLayer(101, layer);
+  }
+}
+
+// General testing of the bitrate controller.
+TEST_F(ScreenshareLayerTestVP9, 2LayerBitrate) {
+  InitScreenshareLayers(2);
+  ConfigureBitrateForLayer(100, 0);
+
+  expected_.layer[0].upd_buf = 0;
+  expected_.layer[1].upd_buf = 1;
+  expected_.layer[0].ref_buf1 = -1;
+  expected_.layer[1].ref_buf1 = -1;
+  expected_.start_layer = 0;
+  expected_.stop_layer = 1;
+
+  expected_.is_keyframe = true;
+  EqualToExpected();
+  AddKilobitsToLayer(100, 0);
+
+  expected_.layer[0].ref_buf1 = 0;
+  expected_.layer[1].ref_buf1 = 1;
+  expected_.is_keyframe = false;
+  AdvanceTime(199);
+  EqualToExpected();
+  AddKilobitsToLayer(100, 0);
+
+  expected_.start_layer = 1;
+  for (int frame = 0; frame < 3; ++frame) {
+    AdvanceTime(200);
+    EqualToExpected();
+    AddKilobitsToLayer(100, 1);
+  }
+
+  // Just before enough bits become available for L0 @0.999 seconds.
+  AdvanceTime(199);
+  EqualToExpected();
+  AddKilobitsToLayer(100, 1);
+
+  // Just after enough bits become available for L0 @1.0001 seconds.
+  expected_.start_layer = 0;
+  AdvanceTime(2);
+  EqualToExpected();
+  AddKilobitsToLayer(100, 0);
+
+  // Keyframes always encode all layers, even if it is over budget.
+  expected_.layer[0].ref_buf1 = -1;
+  expected_.layer[1].ref_buf1 = -1;
+  expected_.is_keyframe = true;
+  AdvanceTime(499);
+  EqualToExpected();
+  expected_.layer[0].ref_buf1 = 0;
+  expected_.layer[1].ref_buf1 = 1;
+  expected_.start_layer = 1;
+  expected_.is_keyframe = false;
+  EqualToExpected();
+  AddKilobitsToLayer(100, 0);
+
+  // 400 kb in L0 --> @3 second mark to fall below the threshold..
+  // just before @2.999 seconds.
+  expected_.is_keyframe = false;
+  AdvanceTime(1499);
+  EqualToExpected();
+  AddKilobitsToLayer(100, 1);
+
+  // just after @3.001 seconds.
+  expected_.start_layer = 0;
+  AdvanceTime(2);
+  EqualToExpected();
+  AddKilobitsToLayer(100, 0);
+}
+
+// General testing of the bitrate controller.
+TEST_F(ScreenshareLayerTestVP9, 3LayerBitrate) {
+  InitScreenshareLayers(3);
+  ConfigureBitrateForLayer(100, 0);
+  ConfigureBitrateForLayer(100, 1);
+
+  for (int l = 0; l < 3; ++l) {
+    expected_.layer[l].upd_buf = l;
+    expected_.layer[l].ref_buf1 = l;
+  }
+  expected_.start_layer = 0;
+  expected_.stop_layer = 2;
+
+  EqualToExpected();
+  AddKilobitsToLayer(105, 0);
+  AddKilobitsToLayer(30, 1);
+
+  AdvanceTime(199);
+  EqualToExpected();
+  AddKilobitsToLayer(105, 0);
+  AddKilobitsToLayer(30, 1);
+
+  expected_.start_layer = 1;
+  AdvanceTime(200);
+  EqualToExpected();
+  AddKilobitsToLayer(130, 1);
+
+  expected_.start_layer = 2;
+  AdvanceTime(200);
+  EqualToExpected();
+
+  // 400 kb in L1 --> @1.0 second mark to fall below threshold.
+  // 210 kb in L0 --> @1.1 second mark to fall below threshold.
+  // Just before L1 @0.999 seconds.
+  AdvanceTime(399);
+  EqualToExpected();
+
+  // Just after L1 @1.001 seconds.
+  expected_.start_layer = 1;
+  AdvanceTime(2);
+  EqualToExpected();
+
+  // Just before L0 @1.099 seconds.
+  AdvanceTime(99);
+  EqualToExpected();
+
+  // Just after L0 @1.101 seconds.
+  expected_.start_layer = 0;
+  AdvanceTime(2);
+  EqualToExpected();
+
+  // @1.1 seconds
+  AdvanceTime(99);
+  EqualToExpected();
+  AddKilobitsToLayer(200, 1);
+
+  expected_.is_keyframe = true;
+  for (int l = 0; l < 3; ++l)
+    expected_.layer[l].ref_buf1 = -1;
+  AdvanceTime(200);
+  EqualToExpected();
+
+  expected_.is_keyframe = false;
+  expected_.start_layer = 2;
+  for (int l = 0; l < 3; ++l)
+    expected_.layer[l].ref_buf1 = l;
+  AdvanceTime(200);
+  EqualToExpected();
+}
+
+// Test that the bitrate calculations are
+// correct when the timestamp wrap.
+TEST_F(ScreenshareLayerTestVP9, TimestampWrap) {
+  InitScreenshareLayers(2);
+  ConfigureBitrateForLayer(100, 0);
+
+  expected_.layer[0].upd_buf = 0;
+  expected_.layer[0].ref_buf1 = 0;
+  expected_.layer[1].upd_buf = 1;
+  expected_.layer[1].ref_buf1 = 1;
+  expected_.start_layer = 0;
+  expected_.stop_layer = 1;
+
+  // Advance time to just before the timestamp wraps.
+  AdvanceTime(std::numeric_limits<uint32_t>::max() / (kTickFrequency / 1000));
+  EqualToExpected();
+  AddKilobitsToLayer(200, 0);
+
+  // Wrap
+  expected_.start_layer = 1;
+  AdvanceTime(1);
+  EqualToExpected();
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9.gyp b/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
index 752521c5cb..8993d79bd7 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
@@ -14,30 +14,26 @@
     {
       'target_name': 'webrtc_vp9',
       'type': 'static_library',
-      'dependencies': [
-        '<(webrtc_root)/common_video/common_video.gyp:common_video',
-        '<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
       'conditions': [
         ['build_libvpx==1', {
           'dependencies': [
             '<(libvpx_dir)/libvpx.gyp:libvpx_new',
           ],
         }],
-        ['build_vp9==1', {
-          'sources': [
-            'include/vp9.h',
-            'vp9_frame_buffer_pool.cc',
-            'vp9_frame_buffer_pool.h',
-            'vp9_impl.cc',
-            'vp9_impl.h',
-          ],
-        }, {
-          'sources': [
-            'vp9_dummy_impl.cc',
-          ],
-        }],
+      ],
+      'dependencies': [
+        '<(webrtc_root)/common_video/common_video.gyp:common_video',
+        '<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+      ],
+      'sources': [
+        'include/vp9.h',
+        'screenshare_layers.cc',
+        'screenshare_layers.h',
+        'vp9_frame_buffer_pool.cc',
+        'vp9_frame_buffer_pool.h',
+        'vp9_impl.cc',
+        'vp9_impl.h',
       ],
     },
   ],
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_dummy_impl.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_dummy_impl.cc
deleted file mode 100644
index 491ccbe79c..0000000000
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_dummy_impl.cc
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- *
- */
-
-// This file contains an implementation of empty webrtc VP9 encoder/decoder
-// factories so it is possible to build webrtc without linking with vp9.
-#include "webrtc/modules/video_coding/codecs/vp9/vp9_impl.h"
-
-namespace webrtc {
-VP9Encoder* VP9Encoder::Create() { return nullptr; }
-VP9Decoder* VP9Decoder::Create() { return nullptr; }
-}
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
index bedbe68ca8..62c05d34fa 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
@@ -16,7 +16,7 @@
 #include "vpx/vpx_frame_buffer.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/base/logging.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
index 0ca7eeabe9..e554795519 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -21,36 +21,31 @@
 #include "vpx/vp8cx.h"
 #include "vpx/vp8dx.h"
 
-#include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/keep_ref_until_done.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
-namespace {
-
-// VP9DecoderImpl::ReturnFrame helper function used with WrappedI420Buffer.
-static void WrappedI420BufferNoLongerUsedCb(
-    webrtc::Vp9FrameBufferPool::Vp9FrameBuffer* img_buffer) {
-  img_buffer->Release();
-}
-
-}  // anonymous namespace
-
 namespace webrtc {
 
 // Only positive speeds, range for real-time coding currently is: 5 - 8.
 // Lower means slower/better quality, higher means fastest/lower quality.
 int GetCpuSpeed(int width, int height) {
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  return 8;
+#else
   // For smaller resolutions, use lower speed setting (get some coding gain at
   // the cost of increased encoding complexity).
   if (width * height <= 352 * 288)
     return 5;
   else
     return 7;
+#endif
 }
 
 VP9Encoder* VP9Encoder::Create() {
@@ -59,7 +54,7 @@ VP9Encoder* VP9Encoder::Create() {
 
 void VP9EncoderImpl::EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
                                                       void* user_data) {
-  VP9EncoderImpl* enc = (VP9EncoderImpl*)(user_data);
+  VP9EncoderImpl* enc = static_cast<VP9EncoderImpl*>(user_data);
   enc->GetEncodedLayerFrame(pkt);
 }
 
@@ -76,9 +71,12 @@ VP9EncoderImpl::VP9EncoderImpl()
       raw_(NULL),
       input_image_(NULL),
       tl0_pic_idx_(0),
-      gof_idx_(0),
+      frames_since_kf_(0),
       num_temporal_layers_(0),
-      num_spatial_layers_(0) {
+      num_spatial_layers_(0),
+      frames_encoded_(0),
+      // Use two spatial when screensharing with flexible mode.
+      spatial_layer_(new ScreenshareLayersVP9(2)) {
   memset(&codec_, 0, sizeof(codec_));
   uint32_t seed = static_cast<uint32_t>(TickTime::MillisecondTimestamp());
   srand(seed);
@@ -90,7 +88,7 @@ VP9EncoderImpl::~VP9EncoderImpl() {
 
 int VP9EncoderImpl::Release() {
   if (encoded_image_._buffer != NULL) {
-    delete [] encoded_image_._buffer;
+    delete[] encoded_image_._buffer;
     encoded_image_._buffer = NULL;
   }
   if (encoder_ != NULL) {
@@ -112,42 +110,72 @@ int VP9EncoderImpl::Release() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
+bool VP9EncoderImpl::ExplicitlyConfiguredSpatialLayers() const {
+  // We check target_bitrate_bps of the 0th layer to see if the spatial layers
+  // (i.e. bitrates) were explicitly configured.
+  return num_spatial_layers_ > 1 &&
+         codec_.spatialLayers[0].target_bitrate_bps > 0;
+}
+
 bool VP9EncoderImpl::SetSvcRates() {
-  float rate_ratio[VPX_MAX_LAYERS] = {0};
-  float total = 0;
   uint8_t i = 0;
 
-  for (i = 0; i < num_spatial_layers_; ++i) {
-    if (svc_internal_.svc_params.scaling_factor_num[i] <= 0 ||
-        svc_internal_.svc_params.scaling_factor_den[i] <= 0) {
+  if (ExplicitlyConfiguredSpatialLayers()) {
+    if (num_temporal_layers_ > 1) {
+      LOG(LS_ERROR) << "Multiple temporal layers when manually specifying "
+                       "spatial layers not implemented yet!";
       return false;
     }
-    rate_ratio[i] = static_cast<float>(
-        svc_internal_.svc_params.scaling_factor_num[i]) /
-        svc_internal_.svc_params.scaling_factor_den[i];
-    total += rate_ratio[i];
-  }
-
-  for (i = 0; i < num_spatial_layers_; ++i) {
-    config_->ss_target_bitrate[i] = static_cast<unsigned int>(
-        config_->rc_target_bitrate * rate_ratio[i] / total);
-    if (num_temporal_layers_ == 1) {
-      config_->layer_target_bitrate[i] = config_->ss_target_bitrate[i];
-    } else if (num_temporal_layers_ == 2) {
-      config_->layer_target_bitrate[i * num_temporal_layers_] =
-          config_->ss_target_bitrate[i] * 2 / 3;
-      config_->layer_target_bitrate[i * num_temporal_layers_ + 1] =
-          config_->ss_target_bitrate[i];
-    } else if (num_temporal_layers_ == 3) {
-      config_->layer_target_bitrate[i * num_temporal_layers_] =
-          config_->ss_target_bitrate[i] / 2;
-      config_->layer_target_bitrate[i * num_temporal_layers_ + 1] =
-          config_->layer_target_bitrate[i * num_temporal_layers_] +
-          (config_->ss_target_bitrate[i] / 4);
-      config_->layer_target_bitrate[i * num_temporal_layers_ + 2] =
-          config_->ss_target_bitrate[i];
-    } else {
-      return false;
+    int total_bitrate_bps = 0;
+    for (i = 0; i < num_spatial_layers_; ++i)
+      total_bitrate_bps += codec_.spatialLayers[i].target_bitrate_bps;
+    // If total bitrate differs now from what has been specified at the
+    // beginning, update the bitrates in the same ratio as before.
+    for (i = 0; i < num_spatial_layers_; ++i) {
+      config_->ss_target_bitrate[i] = config_->layer_target_bitrate[i] =
+          static_cast<int>(static_cast<int64_t>(config_->rc_target_bitrate) *
+                           codec_.spatialLayers[i].target_bitrate_bps /
+                           total_bitrate_bps);
+    }
+  } else {
+    float rate_ratio[VPX_MAX_LAYERS] = {0};
+    float total = 0;
+
+    for (i = 0; i < num_spatial_layers_; ++i) {
+      if (svc_internal_.svc_params.scaling_factor_num[i] <= 0 ||
+          svc_internal_.svc_params.scaling_factor_den[i] <= 0) {
+        LOG(LS_ERROR) << "Scaling factors not specified!";
+        return false;
+      }
+      rate_ratio[i] =
+          static_cast<float>(svc_internal_.svc_params.scaling_factor_num[i]) /
+          svc_internal_.svc_params.scaling_factor_den[i];
+      total += rate_ratio[i];
+    }
+
+    for (i = 0; i < num_spatial_layers_; ++i) {
+      config_->ss_target_bitrate[i] = static_cast<unsigned int>(
+          config_->rc_target_bitrate * rate_ratio[i] / total);
+      if (num_temporal_layers_ == 1) {
+        config_->layer_target_bitrate[i] = config_->ss_target_bitrate[i];
+      } else if (num_temporal_layers_ == 2) {
+        config_->layer_target_bitrate[i * num_temporal_layers_] =
+            config_->ss_target_bitrate[i] * 2 / 3;
+        config_->layer_target_bitrate[i * num_temporal_layers_ + 1] =
+            config_->ss_target_bitrate[i];
+      } else if (num_temporal_layers_ == 3) {
+        config_->layer_target_bitrate[i * num_temporal_layers_] =
+            config_->ss_target_bitrate[i] / 2;
+        config_->layer_target_bitrate[i * num_temporal_layers_ + 1] =
+            config_->layer_target_bitrate[i * num_temporal_layers_] +
+            (config_->ss_target_bitrate[i] / 4);
+        config_->layer_target_bitrate[i * num_temporal_layers_ + 2] =
+            config_->ss_target_bitrate[i];
+      } else {
+        LOG(LS_ERROR) << "Unsupported number of temporal layers: "
+                      << num_temporal_layers_;
+        return false;
+      }
     }
   }
 
@@ -178,6 +206,7 @@ int VP9EncoderImpl::SetRates(uint32_t new_bitrate_kbit,
   }
   config_->rc_target_bitrate = new_bitrate_kbit;
   codec_.maxFramerate = new_framerate;
+  spatial_layer_->ConfigureBitrate(new_bitrate_kbit, 0);
 
   if (!SetSvcRates()) {
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
@@ -216,6 +245,7 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
   if (inst->codecSpecific.VP9.numberOfSpatialLayers > 2) {
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   }
+
   int retVal = Release();
   if (retVal < 0) {
     return retVal;
@@ -237,10 +267,10 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
     num_temporal_layers_ = 1;
 
   // Random start 16 bits is enough.
-  picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;
+  picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;  // NOLINT
   // Allocate memory for encoded image
   if (encoded_image_._buffer != NULL) {
-    delete [] encoded_image_._buffer;
+    delete[] encoded_image_._buffer;
   }
   encoded_image_._size = CalcBufferSize(kI420, codec_.width, codec_.height);
   encoded_image_._buffer = new uint8_t[encoded_image_._size];
@@ -248,8 +278,8 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
   // Creating a wrapper to the image - setting image data to NULL. Actual
   // pointer will be set in encode. Setting align to 1, as it is meaningless
   // (actual memory is not allocated).
-  raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height,
-                      1, NULL);
+  raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height, 1,
+                      NULL);
   // Populate encoder configuration with default values.
   if (vpx_codec_enc_config_default(vpx_codec_vp9_cx(), config_, 0)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
@@ -264,8 +294,8 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
   config_->g_lag_in_frames = 0;  // 0- no frame lagging
   config_->g_threads = 1;
   // Rate control settings.
-  config_->rc_dropframe_thresh = inst->codecSpecific.VP9.frameDroppingOn ?
-      30 : 0;
+  config_->rc_dropframe_thresh =
+      inst->codecSpecific.VP9.frameDroppingOn ? 30 : 0;
   config_->rc_end_usage = VPX_CBR;
   config_->g_pass = VPX_RC_ONE_PASS;
   config_->rc_min_quantizer = 2;
@@ -277,24 +307,32 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
   config_->rc_buf_sz = 1000;
   // Set the maximum target size of any key-frame.
   rc_max_intra_target_ = MaxIntraTarget(config_->rc_buf_optimal_sz);
-  if (inst->codecSpecific.VP9.keyFrameInterval  > 0) {
+  if (inst->codecSpecific.VP9.keyFrameInterval > 0) {
     config_->kf_mode = VPX_KF_AUTO;
     config_->kf_max_dist = inst->codecSpecific.VP9.keyFrameInterval;
+    // Needs to be set (in svc mode) to get correct periodic key frame interval
+    // (will have no effect in non-svc).
+    config_->kf_min_dist = config_->kf_max_dist;
   } else {
     config_->kf_mode = VPX_KF_DISABLED;
   }
-  config_->rc_resize_allowed = inst->codecSpecific.VP9.automaticResizeOn ?
-      1 : 0;
+  config_->rc_resize_allowed =
+      inst->codecSpecific.VP9.automaticResizeOn ? 1 : 0;
   // Determine number of threads based on the image size and #cores.
-  config_->g_threads = NumberOfThreads(config_->g_w,
-                                       config_->g_h,
-                                       number_of_cores);
+  config_->g_threads =
+      NumberOfThreads(config_->g_w, config_->g_h, number_of_cores);
 
   cpu_speed_ = GetCpuSpeed(config_->g_w, config_->g_h);
 
   // TODO(asapersson): Check configuration of temporal switch up and increase
   // pattern length.
-  if (num_temporal_layers_ == 1) {
+  is_flexible_mode_ = inst->codecSpecific.VP9.flexibleMode;
+  if (is_flexible_mode_) {
+    config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_BYPASS;
+    config_->ts_number_layers = num_temporal_layers_;
+    if (codec_.mode == kScreensharing)
+      spatial_layer_->ConfigureBitrate(inst->startBitrate, 0);
+  } else if (num_temporal_layers_ == 1) {
     gof_.SetGofInfoVP9(kTemporalStructureMode1);
     config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_NOLAYERING;
     config_->ts_number_layers = 1;
@@ -326,7 +364,7 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   }
 
-  tl0_pic_idx_ = static_cast<uint8_t>(rand());
+  tl0_pic_idx_ = static_cast<uint8_t>(rand());  // NOLINT
 
   return InitAndSetControlSettings(inst);
 }
@@ -347,16 +385,28 @@ int VP9EncoderImpl::NumberOfThreads(int width,
 }
 
 int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
-  config_->ss_number_layers = num_spatial_layers_;
-
-  int scaling_factor_num = 256;
-  for (int i = num_spatial_layers_ - 1; i >= 0; --i) {
+  // Set QP-min/max per spatial and temporal layer.
+  int tot_num_layers = num_spatial_layers_ * num_temporal_layers_;
+  for (int i = 0; i < tot_num_layers; ++i) {
     svc_internal_.svc_params.max_quantizers[i] = config_->rc_max_quantizer;
     svc_internal_.svc_params.min_quantizers[i] = config_->rc_min_quantizer;
-    // 1:2 scaling in each dimension.
-    svc_internal_.svc_params.scaling_factor_num[i] = scaling_factor_num;
-    svc_internal_.svc_params.scaling_factor_den[i] = 256;
-    scaling_factor_num /= 2;
+  }
+  config_->ss_number_layers = num_spatial_layers_;
+  if (ExplicitlyConfiguredSpatialLayers()) {
+    for (int i = 0; i < num_spatial_layers_; ++i) {
+      const auto& layer = codec_.spatialLayers[i];
+      svc_internal_.svc_params.scaling_factor_num[i] = layer.scaling_factor_num;
+      svc_internal_.svc_params.scaling_factor_den[i] = layer.scaling_factor_den;
+    }
+  } else {
+    int scaling_factor_num = 256;
+    for (int i = num_spatial_layers_ - 1; i >= 0; --i) {
+      // 1:2 scaling in each dimension.
+      svc_internal_.svc_params.scaling_factor_num[i] = scaling_factor_num;
+      svc_internal_.svc_params.scaling_factor_den[i] = 256;
+      if (codec_.mode != kScreensharing)
+        scaling_factor_num /= 2;
+    }
   }
 
   if (!SetSvcRates()) {
@@ -381,8 +431,10 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
   }
   // Register callback for getting each spatial layer.
   vpx_codec_priv_output_cx_pkt_cb_pair_t cbp = {
-      VP9EncoderImpl::EncoderOutputCodedPacketCallback, (void*)(this)};
-  vpx_codec_control(encoder_, VP9E_REGISTER_CX_CALLBACK, (void*)(&cbp));
+      VP9EncoderImpl::EncoderOutputCodedPacketCallback,
+      reinterpret_cast<void*>(this)};
+  vpx_codec_control(encoder_, VP9E_REGISTER_CX_CALLBACK,
+                    reinterpret_cast<void*>(&cbp));
 
   // Control function to set the number of column tiles in encoding a frame, in
   // log2 unit: e.g., 0 = 1 tile column, 1 = 2 tile columns, 2 = 4 tile columns.
@@ -417,7 +469,7 @@ uint32_t VP9EncoderImpl::MaxIntraTarget(uint32_t optimal_buffer_size) {
       optimal_buffer_size * scale_par * codec_.maxFramerate / 10;
   // Don't go below 3 times the per frame bandwidth.
   const uint32_t min_intra_size = 300;
-  return (target_pct < min_intra_size) ? min_intra_size: target_pct;
+  return (target_pct < min_intra_size) ? min_intra_size : target_pct;
 }
 
 int VP9EncoderImpl::Encode(const VideoFrame& input_image,
@@ -455,12 +507,35 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
   raw_->stride[VPX_PLANE_U] = input_image.stride(kUPlane);
   raw_->stride[VPX_PLANE_V] = input_image.stride(kVPlane);
 
-  int flags = 0;
+  vpx_enc_frame_flags_t flags = 0;
   bool send_keyframe = (frame_type == kVideoFrameKey);
   if (send_keyframe) {
     // Key frame request from caller.
     flags = VPX_EFLAG_FORCE_KF;
   }
+
+  if (is_flexible_mode_) {
+    SuperFrameRefSettings settings;
+
+    // These structs are copied when calling vpx_codec_control,
+    // therefore it is ok for them to go out of scope.
+    vpx_svc_ref_frame_config enc_layer_conf;
+    vpx_svc_layer_id layer_id;
+
+    if (codec_.mode == kRealtimeVideo) {
+      // Real time video not yet implemented in flexible mode.
+      RTC_NOTREACHED();
+    } else {
+      settings = spatial_layer_->GetSuperFrameSettings(input_image.timestamp(),
+                                                       send_keyframe);
+    }
+    enc_layer_conf = GenerateRefsAndFlags(settings);
+    layer_id.temporal_layer_id = 0;
+    layer_id.spatial_layer_id = settings.start_layer;
+    vpx_codec_control(encoder_, VP9E_SET_SVC_LAYER_ID, &layer_id);
+    vpx_codec_control(encoder_, VP9E_SET_SVC_REF_FRAME_CONFIG, &enc_layer_conf);
+  }
+
   assert(codec_.maxFramerate > 0);
   uint32_t duration = 90000 / codec_.maxFramerate;
   if (vpx_codec_encode(encoder_, raw_, timestamp_, duration, flags,
@@ -473,12 +548,12 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
 }
 
 void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
-                                       const vpx_codec_cx_pkt& pkt,
-                                       uint32_t timestamp) {
+                                           const vpx_codec_cx_pkt& pkt,
+                                           uint32_t timestamp) {
   assert(codec_specific != NULL);
   codec_specific->codecType = kVideoCodecVP9;
-  CodecSpecificInfoVP9 *vp9_info = &(codec_specific->codecSpecific.VP9);
-  // TODO(asapersson): Set correct values.
+  CodecSpecificInfoVP9* vp9_info = &(codec_specific->codecSpecific.VP9);
+  // TODO(asapersson): Set correct value.
   vp9_info->inter_pic_predicted =
       (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? false : true;
   vp9_info->flexible_mode = codec_.codecSpecific.VP9.flexibleMode;
@@ -486,9 +561,6 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
                                  !codec_.codecSpecific.VP9.flexibleMode)
                                     ? true
                                     : false;
-  if (pkt.data.frame.flags & VPX_FRAME_IS_KEY) {
-    gof_idx_ = 0;
-  }
 
   vpx_svc_layer_id_t layer_id = {0};
   vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
@@ -511,25 +583,31 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
     vp9_info->ss_data_available = false;
   }
 
-  if (vp9_info->flexible_mode) {
-    vp9_info->gof_idx = kNoGofIdx;
+  // TODO(asapersson): this info has to be obtained from the encoder.
+  vp9_info->temporal_up_switch = false;
+
+  bool is_first_frame = false;
+  if (is_flexible_mode_) {
+    is_first_frame =
+        layer_id.spatial_layer_id == spatial_layer_->GetStartLayer();
   } else {
-    vp9_info->gof_idx =
-        static_cast<uint8_t>(gof_idx_++ % gof_.num_frames_in_gof);
+    is_first_frame = layer_id.spatial_layer_id == 0;
   }
 
-  // TODO(asapersson): this info has to be obtained from the encoder.
-  vp9_info->temporal_up_switch = true;
-
-  if (layer_id.spatial_layer_id == 0) {
+  if (is_first_frame) {
     picture_id_ = (picture_id_ + 1) & 0x7FFF;
     // TODO(asapersson): this info has to be obtained from the encoder.
     vp9_info->inter_layer_predicted = false;
+    ++frames_since_kf_;
   } else {
     // TODO(asapersson): this info has to be obtained from the encoder.
     vp9_info->inter_layer_predicted = true;
   }
 
+  if (pkt.data.frame.flags & VPX_FRAME_IS_KEY) {
+    frames_since_kf_ = 0;
+  }
+
   vp9_info->picture_id = picture_id_;
 
   if (!vp9_info->flexible_mode) {
@@ -542,6 +620,20 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
   // Always populate this, so that the packetizer can properly set the marker
   // bit.
   vp9_info->num_spatial_layers = num_spatial_layers_;
+
+  vp9_info->num_ref_pics = 0;
+  if (vp9_info->flexible_mode) {
+    vp9_info->gof_idx = kNoGofIdx;
+    vp9_info->num_ref_pics = num_ref_pics_[layer_id.spatial_layer_id];
+    for (int i = 0; i < num_ref_pics_[layer_id.spatial_layer_id]; ++i) {
+      vp9_info->p_diff[i] = p_diff_[layer_id.spatial_layer_id][i];
+    }
+  } else {
+    vp9_info->gof_idx =
+        static_cast<uint8_t>(frames_since_kf_ % gof_.num_frames_in_gof);
+    vp9_info->temporal_up_switch = gof_.temporal_up_switch[vp9_info->gof_idx];
+  }
+
   if (vp9_info->ss_data_available) {
     vp9_info->spatial_layer_resolution_present = true;
     for (size_t i = 0; i < vp9_info->num_spatial_layers; ++i) {
@@ -577,6 +669,14 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
   frag_info.fragmentationPlType[part_idx] = 0;
   frag_info.fragmentationTimeDiff[part_idx] = 0;
   encoded_image_._length += static_cast<uint32_t>(pkt->data.frame.sz);
+
+  vpx_svc_layer_id_t layer_id = {0};
+  vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
+  if (is_flexible_mode_ && codec_.mode == kScreensharing)
+    spatial_layer_->LayerFrameEncoded(
+        static_cast<unsigned int>(encoded_image_._length),
+        layer_id.spatial_layer_id);
+
   assert(encoded_image_._length <= encoded_image_._size);
 
   // End of frame.
@@ -598,6 +698,108 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
+vpx_svc_ref_frame_config VP9EncoderImpl::GenerateRefsAndFlags(
+    const SuperFrameRefSettings& settings) {
+  static const vpx_enc_frame_flags_t kAllFlags =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_LAST |
+      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_GF;
+  vpx_svc_ref_frame_config sf_conf = {};
+  if (settings.is_keyframe) {
+    // Used later on to make sure we don't make any invalid references.
+    memset(buffer_updated_at_frame_, -1, sizeof(buffer_updated_at_frame_));
+    for (int layer = settings.start_layer; layer <= settings.stop_layer;
+         ++layer) {
+      num_ref_pics_[layer] = 0;
+      buffer_updated_at_frame_[settings.layer[layer].upd_buf] = frames_encoded_;
+      // When encoding a keyframe only the alt_fb_idx is used
+      // to specify which layer ends up in which buffer.
+      sf_conf.alt_fb_idx[layer] = settings.layer[layer].upd_buf;
+    }
+  } else {
+    for (int layer_idx = settings.start_layer; layer_idx <= settings.stop_layer;
+         ++layer_idx) {
+      vpx_enc_frame_flags_t layer_flags = kAllFlags;
+      num_ref_pics_[layer_idx] = 0;
+      int8_t refs[3] = {settings.layer[layer_idx].ref_buf1,
+                        settings.layer[layer_idx].ref_buf2,
+                        settings.layer[layer_idx].ref_buf3};
+
+      for (unsigned int ref_idx = 0; ref_idx < kMaxVp9RefPics; ++ref_idx) {
+        if (refs[ref_idx] == -1)
+          continue;
+
+        RTC_DCHECK_GE(refs[ref_idx], 0);
+        RTC_DCHECK_LE(refs[ref_idx], 7);
+        // Easier to remove flags from all flags rather than having to
+        // build the flags from 0.
+        switch (num_ref_pics_[layer_idx]) {
+          case 0: {
+            sf_conf.lst_fb_idx[layer_idx] = refs[ref_idx];
+            layer_flags &= ~VP8_EFLAG_NO_REF_LAST;
+            break;
+          }
+          case 1: {
+            sf_conf.gld_fb_idx[layer_idx] = refs[ref_idx];
+            layer_flags &= ~VP8_EFLAG_NO_REF_GF;
+            break;
+          }
+          case 2: {
+            sf_conf.alt_fb_idx[layer_idx] = refs[ref_idx];
+            layer_flags &= ~VP8_EFLAG_NO_REF_ARF;
+            break;
+          }
+        }
+        // Make sure we don't reference a buffer that hasn't been
+        // used at all or hasn't been used since a keyframe.
+        RTC_DCHECK_NE(buffer_updated_at_frame_[refs[ref_idx]], -1);
+
+        p_diff_[layer_idx][num_ref_pics_[layer_idx]] =
+            frames_encoded_ - buffer_updated_at_frame_[refs[ref_idx]];
+        num_ref_pics_[layer_idx]++;
+      }
+
+      bool upd_buf_same_as_a_ref = false;
+      if (settings.layer[layer_idx].upd_buf != -1) {
+        for (unsigned int ref_idx = 0; ref_idx < kMaxVp9RefPics; ++ref_idx) {
+          if (settings.layer[layer_idx].upd_buf == refs[ref_idx]) {
+            switch (ref_idx) {
+              case 0: {
+                layer_flags &= ~VP8_EFLAG_NO_UPD_LAST;
+                break;
+              }
+              case 1: {
+                layer_flags &= ~VP8_EFLAG_NO_UPD_GF;
+                break;
+              }
+              case 2: {
+                layer_flags &= ~VP8_EFLAG_NO_UPD_ARF;
+                break;
+              }
+            }
+            upd_buf_same_as_a_ref = true;
+            break;
+          }
+        }
+        if (!upd_buf_same_as_a_ref) {
+          // If we have three references and a buffer is specified to be
+          // updated, then that buffer must be the same as one of the
+          // three references.
+          RTC_CHECK_LT(num_ref_pics_[layer_idx], kMaxVp9RefPics);
+
+          sf_conf.alt_fb_idx[layer_idx] = settings.layer[layer_idx].upd_buf;
+          layer_flags ^= VP8_EFLAG_NO_UPD_ARF;
+        }
+
+        int updated_buffer = settings.layer[layer_idx].upd_buf;
+        buffer_updated_at_frame_[updated_buffer] = frames_encoded_;
+        sf_conf.frame_flags[layer_idx] = layer_flags;
+      }
+    }
+  }
+  ++frames_encoded_;
+  return sf_conf;
+}
+
 int VP9EncoderImpl::SetChannelParameters(uint32_t packet_loss, int64_t rtt) {
   return WEBRTC_VIDEO_CODEC_OK;
 }
@@ -608,6 +810,10 @@ int VP9EncoderImpl::RegisterEncodeCompleteCallback(
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
+const char* VP9EncoderImpl::ImplementationName() const {
+  return "libvpx";
+}
+
 VP9Decoder* VP9Decoder::Create() {
   return new VP9DecoderImpl();
 }
@@ -652,7 +858,7 @@ int VP9DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
   if (decoder_ == NULL) {
     decoder_ = new vpx_codec_ctx_t;
   }
-  vpx_codec_dec_cfg_t  cfg;
+  vpx_codec_dec_cfg_t cfg;
   // Setting number of threads to a constant value (1)
   cfg.threads = 1;
   cfg.h = cfg.w = 0;  // set after decode
@@ -705,10 +911,8 @@ int VP9DecoderImpl::Decode(const EncodedImage& input_image,
   }
   // During decode libvpx may get and release buffers from |frame_buffer_pool_|.
   // In practice libvpx keeps a few (~3-4) buffers alive at a time.
-  if (vpx_codec_decode(decoder_,
-                       buffer,
-                       static_cast<unsigned int>(input_image._length),
-                       0,
+  if (vpx_codec_decode(decoder_, buffer,
+                       static_cast<unsigned int>(input_image._length), 0,
                        VPX_DL_REALTIME)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
@@ -730,24 +934,22 @@ int VP9DecoderImpl::ReturnFrame(const vpx_image_t* img, uint32_t timestamp) {
   }
 
   // This buffer contains all of |img|'s image data, a reference counted
-  // Vp9FrameBuffer. Performing AddRef/Release ensures it is not released and
-  // recycled during use (libvpx is done with the buffers after a few
+  // Vp9FrameBuffer. (libvpx is done with the buffers after a few
   // vpx_codec_decode calls or vpx_codec_destroy).
   Vp9FrameBufferPool::Vp9FrameBuffer* img_buffer =
       static_cast<Vp9FrameBufferPool::Vp9FrameBuffer*>(img->fb_priv);
-  img_buffer->AddRef();
   // The buffer can be used directly by the VideoFrame (without copy) by
   // using a WrappedI420Buffer.
   rtc::scoped_refptr<WrappedI420Buffer> img_wrapped_buffer(
       new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
-          img->d_w, img->d_h,
-          img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
-          img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
-          img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
+          img->d_w, img->d_h, img->planes[VPX_PLANE_Y],
+          img->stride[VPX_PLANE_Y], img->planes[VPX_PLANE_U],
+          img->stride[VPX_PLANE_U], img->planes[VPX_PLANE_V],
+          img->stride[VPX_PLANE_V],
           // WrappedI420Buffer's mechanism for allowing the release of its frame
           // buffer is through a callback function. This is where we should
           // release |img_buffer|.
-          rtc::Bind(&WrappedI420BufferNoLongerUsedCb, img_buffer)));
+          rtc::KeepRefUntilDone(img_buffer)));
 
   VideoFrame decoded_image;
   decoded_image.set_video_frame_buffer(img_wrapped_buffer);
@@ -781,4 +983,9 @@ int VP9DecoderImpl::Release() {
   inited_ = false;
   return WEBRTC_VIDEO_CODEC_OK;
 }
+
+const char* VP9DecoderImpl::ImplementationName() const {
+  return "libvpx";
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
index f9c123079e..bfa4540304 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
@@ -9,8 +9,10 @@
  *
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_IMPL_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
+
+#include <vector>
 
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
@@ -21,6 +23,8 @@
 
 namespace webrtc {
 
+class ScreenshareLayersVP9;
+
 class VP9EncoderImpl : public VP9Encoder {
  public:
   VP9EncoderImpl();
@@ -45,6 +49,22 @@ class VP9EncoderImpl : public VP9Encoder {
 
   void OnDroppedFrame() override {}
 
+  const char* ImplementationName() const override;
+
+  struct LayerFrameRefSettings {
+    int8_t upd_buf = -1;   // -1 - no update,    0..7 - update buffer 0..7
+    int8_t ref_buf1 = -1;  // -1 - no reference, 0..7 - reference buffer 0..7
+    int8_t ref_buf2 = -1;  // -1 - no reference, 0..7 - reference buffer 0..7
+    int8_t ref_buf3 = -1;  // -1 - no reference, 0..7 - reference buffer 0..7
+  };
+
+  struct SuperFrameRefSettings {
+    LayerFrameRefSettings layer[kMaxVp9NumberOfSpatialLayers];
+    uint8_t start_layer = 0;  // The first spatial layer to be encoded.
+    uint8_t stop_layer = 0;   // The last spatial layer to be encoded.
+    bool is_keyframe = false;
+  };
+
  private:
   // Determine number of encoder threads to use.
   int NumberOfThreads(int width, int height, int number_of_cores);
@@ -56,8 +76,18 @@ class VP9EncoderImpl : public VP9Encoder {
                              const vpx_codec_cx_pkt& pkt,
                              uint32_t timestamp);
 
+  bool ExplicitlyConfiguredSpatialLayers() const;
   bool SetSvcRates();
 
+  // Used for flexible mode to set the flags and buffer references used
+  // by the encoder. Also calculates the references used by the RTP
+  // packetizer.
+  //
+  // Has to be called for every frame (keyframes included) to update the
+  // state used to calculate references.
+  vpx_svc_ref_frame_config GenerateRefsAndFlags(
+      const SuperFrameRefSettings& settings);
+
   virtual int GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt);
 
   // Callback function for outputting packets per spatial layer.
@@ -88,11 +118,18 @@ class VP9EncoderImpl : public VP9Encoder {
   GofInfoVP9 gof_;       // Contains each frame's temporal information for
                          // non-flexible mode.
   uint8_t tl0_pic_idx_;  // Only used in non-flexible mode.
-  size_t gof_idx_;       // Only used in non-flexible mode.
+  size_t frames_since_kf_;
   uint8_t num_temporal_layers_;
   uint8_t num_spatial_layers_;
-};
 
+  // Used for flexible mode.
+  bool is_flexible_mode_;
+  int64_t buffer_updated_at_frame_[kNumVp9Buffers];
+  int64_t frames_encoded_;
+  uint8_t num_ref_pics_[kMaxVp9NumberOfSpatialLayers];
+  uint8_t p_diff_[kMaxVp9NumberOfSpatialLayers][kMaxVp9RefPics];
+  rtc::scoped_ptr<ScreenshareLayersVP9> spatial_layer_;
+};
 
 class VP9DecoderImpl : public VP9Decoder {
  public:
@@ -114,6 +151,8 @@ class VP9DecoderImpl : public VP9Decoder {
 
   int Reset() override;
 
+  const char* ImplementationName() const override;
+
  private:
   int ReturnFrame(const vpx_image_t* img, uint32_t timeStamp);
 
@@ -127,4 +166,4 @@ class VP9DecoderImpl : public VP9Decoder {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_IMPL_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
diff --git a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc b/webrtc/modules/video_coding/content_metrics_processing.cc
index 757ffb0e46..0c3a6dbc6c 100644
--- a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
+++ b/webrtc/modules/video_coding/content_metrics_processing.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/content_metrics_processing.h"
+#include "webrtc/modules/video_coding/content_metrics_processing.h"
 
 #include <math.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 //////////////////////////////////
@@ -38,15 +38,15 @@ int VCMContentMetricsProcessing::Reset() {
   recursive_avg_->Reset();
   uniform_avg_->Reset();
   frame_cnt_uniform_avg_ = 0;
-  avg_motion_level_  = 0.0f;
+  avg_motion_level_ = 0.0f;
   avg_spatial_level_ = 0.0f;
   return VCM_OK;
 }
 
 void VCMContentMetricsProcessing::UpdateFrameRate(uint32_t frameRate) {
   // Update factor for recursive averaging.
-  recursive_avg_factor_ = static_cast<float> (1000.0f) /
-      static_cast<float>(frameRate *  kQmMinIntervalMs);
+  recursive_avg_factor_ = static_cast<float>(1000.0f) /
+                          static_cast<float>(frameRate * kQmMinIntervalMs);
 }
 
 VideoContentMetrics* VCMContentMetricsProcessing::LongTermAvgData() {
@@ -58,10 +58,10 @@ VideoContentMetrics* VCMContentMetricsProcessing::ShortTermAvgData() {
     return NULL;
   }
   // Two metrics are used: motion and spatial level.
-  uniform_avg_->motion_magnitude = avg_motion_level_ /
-      static_cast<float>(frame_cnt_uniform_avg_);
-  uniform_avg_->spatial_pred_err = avg_spatial_level_ /
-      static_cast<float>(frame_cnt_uniform_avg_);
+  uniform_avg_->motion_magnitude =
+      avg_motion_level_ / static_cast<float>(frame_cnt_uniform_avg_);
+  uniform_avg_->spatial_pred_err =
+      avg_spatial_level_ / static_cast<float>(frame_cnt_uniform_avg_);
   return uniform_avg_;
 }
 
@@ -73,7 +73,7 @@ void VCMContentMetricsProcessing::ResetShortTermAvgData() {
 }
 
 int VCMContentMetricsProcessing::UpdateContentData(
-    const VideoContentMetrics *contentMetrics) {
+    const VideoContentMetrics* contentMetrics) {
   if (contentMetrics == NULL) {
     return VCM_OK;
   }
@@ -81,7 +81,7 @@ int VCMContentMetricsProcessing::UpdateContentData(
 }
 
 int VCMContentMetricsProcessing::ProcessContent(
-    const VideoContentMetrics *contentMetrics) {
+    const VideoContentMetrics* contentMetrics) {
   // Update the recursive averaged metrics: average is over longer window
   // of time: over QmMinIntervalMs ms.
   UpdateRecursiveAvg(contentMetrics);
@@ -92,34 +92,33 @@ int VCMContentMetricsProcessing::ProcessContent(
 }
 
 void VCMContentMetricsProcessing::UpdateUniformAvg(
-    const VideoContentMetrics *contentMetrics) {
+    const VideoContentMetrics* contentMetrics) {
   // Update frame counter.
   frame_cnt_uniform_avg_ += 1;
   // Update averaged metrics: motion and spatial level are used.
   avg_motion_level_ += contentMetrics->motion_magnitude;
-  avg_spatial_level_ +=  contentMetrics->spatial_pred_err;
+  avg_spatial_level_ += contentMetrics->spatial_pred_err;
   return;
 }
 
 void VCMContentMetricsProcessing::UpdateRecursiveAvg(
-    const VideoContentMetrics *contentMetrics) {
-
+    const VideoContentMetrics* contentMetrics) {
   // Spatial metrics: 2x2, 1x2(H), 2x1(V).
-  recursive_avg_->spatial_pred_err = (1 - recursive_avg_factor_) *
-      recursive_avg_->spatial_pred_err +
+  recursive_avg_->spatial_pred_err =
+      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err +
       recursive_avg_factor_ * contentMetrics->spatial_pred_err;
 
-  recursive_avg_->spatial_pred_err_h = (1 - recursive_avg_factor_) *
-      recursive_avg_->spatial_pred_err_h +
+  recursive_avg_->spatial_pred_err_h =
+      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err_h +
       recursive_avg_factor_ * contentMetrics->spatial_pred_err_h;
 
-  recursive_avg_->spatial_pred_err_v = (1 - recursive_avg_factor_) *
-      recursive_avg_->spatial_pred_err_v +
+  recursive_avg_->spatial_pred_err_v =
+      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err_v +
       recursive_avg_factor_ * contentMetrics->spatial_pred_err_v;
 
   // Motion metric: Derived from NFD (normalized frame difference).
-  recursive_avg_->motion_magnitude = (1 - recursive_avg_factor_) *
-      recursive_avg_->motion_magnitude +
+  recursive_avg_->motion_magnitude =
+      (1 - recursive_avg_factor_) * recursive_avg_->motion_magnitude +
       recursive_avg_factor_ * contentMetrics->motion_magnitude;
 }
-}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/content_metrics_processing.h b/webrtc/modules/video_coding/content_metrics_processing.h
index 3517f757d4..3f67ec19c9 100644
--- a/webrtc/modules/video_coding/main/source/content_metrics_processing.h
+++ b/webrtc/modules/video_coding/content_metrics_processing.h
@@ -18,14 +18,10 @@ namespace webrtc {
 struct VideoContentMetrics;
 
 // QM interval time (in ms)
-enum {
-  kQmMinIntervalMs = 10000
-};
+enum { kQmMinIntervalMs = 10000 };
 
 // Flag for NFD metric vs motion metric
-enum {
-  kNfdMetric = 1
-};
+enum { kNfdMetric = 1 };
 
 /**********************************/
 /* Content Metrics Processing     */
@@ -36,7 +32,7 @@ class VCMContentMetricsProcessing {
   ~VCMContentMetricsProcessing();
 
   // Update class with latest metrics.
-  int UpdateContentData(const VideoContentMetrics *contentMetrics);
+  int UpdateContentData(const VideoContentMetrics* contentMetrics);
 
   // Reset the short-term averaged content data.
   void ResetShortTermAvgData();
@@ -57,13 +53,13 @@ class VCMContentMetricsProcessing {
 
  private:
   // Compute working average.
-  int ProcessContent(const VideoContentMetrics *contentMetrics);
+  int ProcessContent(const VideoContentMetrics* contentMetrics);
 
   // Update the recursive averaged metrics: longer time average (~5/10 secs).
-  void UpdateRecursiveAvg(const VideoContentMetrics *contentMetrics);
+  void UpdateRecursiveAvg(const VideoContentMetrics* contentMetrics);
 
   // Update the uniform averaged metrics: shorter time average (~RTCP report).
-  void UpdateUniformAvg(const VideoContentMetrics *contentMetrics);
+  void UpdateUniformAvg(const VideoContentMetrics* contentMetrics);
 
   VideoContentMetrics* recursive_avg_;
   VideoContentMetrics* uniform_avg_;
diff --git a/webrtc/modules/video_coding/main/source/decoding_state.cc b/webrtc/modules/video_coding/decoding_state.cc
index cc92f1c83f..89be9b66c1 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state.cc
+++ b/webrtc/modules/video_coding/decoding_state.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/decoding_state.h"
+#include "webrtc/modules/video_coding/decoding_state.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 namespace webrtc {
 
@@ -24,7 +24,9 @@ VCMDecodingState::VCMDecodingState()
       temporal_id_(kNoTemporalIdx),
       tl0_pic_id_(kNoTl0PicIdx),
       full_sync_(true),
-      in_initial_state_(true) {}
+      in_initial_state_(true) {
+  memset(frame_decoded_, 0, sizeof(frame_decoded_));
+}
 
 VCMDecodingState::~VCMDecodingState() {}
 
@@ -37,6 +39,7 @@ void VCMDecodingState::Reset() {
   tl0_pic_id_ = kNoTl0PicIdx;
   full_sync_ = true;
   in_initial_state_ = true;
+  memset(frame_decoded_, 0, sizeof(frame_decoded_));
 }
 
 uint32_t VCMDecodingState::time_stamp() const {
@@ -63,12 +66,33 @@ bool VCMDecodingState::IsOldPacket(const VCMPacket* packet) const {
 
 void VCMDecodingState::SetState(const VCMFrameBuffer* frame) {
   assert(frame != NULL && frame->GetHighSeqNum() >= 0);
-  UpdateSyncState(frame);
+  if (!UsingFlexibleMode(frame))
+    UpdateSyncState(frame);
   sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum());
   time_stamp_ = frame->TimeStamp();
   picture_id_ = frame->PictureId();
   temporal_id_ = frame->TemporalId();
   tl0_pic_id_ = frame->Tl0PicId();
+
+  if (UsingFlexibleMode(frame)) {
+    uint16_t frame_index = picture_id_ % kFrameDecodedLength;
+    if (in_initial_state_) {
+      frame_decoded_cleared_to_ = frame_index;
+    } else if (frame->FrameType() == kVideoFrameKey) {
+      memset(frame_decoded_, 0, sizeof(frame_decoded_));
+      frame_decoded_cleared_to_ = frame_index;
+    } else {
+      if (AheadOfFramesDecodedClearedTo(frame_index)) {
+        while (frame_decoded_cleared_to_ != frame_index) {
+          frame_decoded_cleared_to_ =
+              (frame_decoded_cleared_to_ + 1) % kFrameDecodedLength;
+          frame_decoded_[frame_decoded_cleared_to_] = false;
+        }
+      }
+    }
+    frame_decoded_[frame_index] = true;
+  }
+
   in_initial_state_ = false;
 }
 
@@ -80,6 +104,8 @@ void VCMDecodingState::CopyFrom(const VCMDecodingState& state) {
   tl0_pic_id_ = state.tl0_pic_id_;
   full_sync_ = state.full_sync_;
   in_initial_state_ = state.in_initial_state_;
+  frame_decoded_cleared_to_ = state.frame_decoded_cleared_to_;
+  memcpy(frame_decoded_, state.frame_decoded_, sizeof(frame_decoded_));
 }
 
 bool VCMDecodingState::UpdateEmptyFrame(const VCMFrameBuffer* frame) {
@@ -140,8 +166,8 @@ void VCMDecodingState::UpdateSyncState(const VCMFrameBuffer* frame) {
         full_sync_ = ContinuousPictureId(frame->PictureId());
       }
     } else {
-      full_sync_ = ContinuousSeqNum(static_cast<uint16_t>(
-          frame->GetLowSeqNum()));
+      full_sync_ =
+          ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
     }
   }
 }
@@ -173,7 +199,11 @@ bool VCMDecodingState::ContinuousFrame(const VCMFrameBuffer* frame) const {
   if (!full_sync_ && !frame->LayerSync())
     return false;
   if (UsingPictureId(frame)) {
-    return ContinuousPictureId(frame->PictureId());
+    if (UsingFlexibleMode(frame)) {
+      return ContinuousFrameRefs(frame);
+    } else {
+      return ContinuousPictureId(frame->PictureId());
+    }
   } else {
     return ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
   }
@@ -199,8 +229,7 @@ bool VCMDecodingState::ContinuousSeqNum(uint16_t seq_num) const {
   return seq_num == static_cast<uint16_t>(sequence_num_ + 1);
 }
 
-bool VCMDecodingState::ContinuousLayer(int temporal_id,
-                                       int tl0_pic_id) const {
+bool VCMDecodingState::ContinuousLayer(int temporal_id, int tl0_pic_id) const {
   // First, check if applicable.
   if (temporal_id == kNoTemporalIdx || tl0_pic_id == kNoTl0PicIdx)
     return false;
@@ -216,8 +245,41 @@ bool VCMDecodingState::ContinuousLayer(int temporal_id,
   return (static_cast<uint8_t>(tl0_pic_id_ + 1) == tl0_pic_id);
 }
 
+bool VCMDecodingState::ContinuousFrameRefs(const VCMFrameBuffer* frame) const {
+  uint8_t num_refs = frame->CodecSpecific()->codecSpecific.VP9.num_ref_pics;
+  for (uint8_t r = 0; r < num_refs; ++r) {
+    uint16_t frame_ref = frame->PictureId() -
+                         frame->CodecSpecific()->codecSpecific.VP9.p_diff[r];
+    uint16_t frame_index = frame_ref % kFrameDecodedLength;
+    if (AheadOfFramesDecodedClearedTo(frame_index) ||
+        !frame_decoded_[frame_index]) {
+      return false;
+    }
+  }
+  return true;
+}
+
 bool VCMDecodingState::UsingPictureId(const VCMFrameBuffer* frame) const {
   return (frame->PictureId() != kNoPictureId && picture_id_ != kNoPictureId);
 }
 
+bool VCMDecodingState::UsingFlexibleMode(const VCMFrameBuffer* frame) const {
+  return frame->CodecSpecific()->codecType == kVideoCodecVP9 &&
+         frame->CodecSpecific()->codecSpecific.VP9.flexible_mode;
+}
+
+// TODO(philipel): change how check work, this check practially
+// limits the max p_diff to 64.
+bool VCMDecodingState::AheadOfFramesDecodedClearedTo(uint16_t index) const {
+  // No way of knowing for sure if we are actually ahead of
+  // frame_decoded_cleared_to_. We just make the assumption
+  // that we are not trying to reference back to a very old
+  // index, but instead are referencing a newer index.
+  uint16_t diff =
+      index > frame_decoded_cleared_to_
+          ? kFrameDecodedLength - (index - frame_decoded_cleared_to_)
+          : frame_decoded_cleared_to_ - index;
+  return diff > kFrameDecodedLength / 2;
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/decoding_state.h b/webrtc/modules/video_coding/decoding_state.h
index 99ee335195..f4ea8ae081 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state.h
+++ b/webrtc/modules/video_coding/decoding_state.h
@@ -21,6 +21,11 @@ class VCMPacket;
 
 class VCMDecodingState {
  public:
+  // The max number of bits used to reference back
+  // to a previous frame when using flexible mode.
+  static const uint16_t kNumRefBits = 7;
+  static const uint16_t kFrameDecodedLength = 1 << kNumRefBits;
+
   VCMDecodingState();
   ~VCMDecodingState();
   // Check for old frame
@@ -52,17 +57,24 @@ class VCMDecodingState {
   bool ContinuousPictureId(int picture_id) const;
   bool ContinuousSeqNum(uint16_t seq_num) const;
   bool ContinuousLayer(int temporal_id, int tl0_pic_id) const;
+  bool ContinuousFrameRefs(const VCMFrameBuffer* frame) const;
   bool UsingPictureId(const VCMFrameBuffer* frame) const;
+  bool UsingFlexibleMode(const VCMFrameBuffer* frame) const;
+  bool AheadOfFramesDecodedClearedTo(uint16_t index) const;
 
   // Keep state of last decoded frame.
   // TODO(mikhal/stefan): create designated classes to handle these types.
-  uint16_t    sequence_num_;
-  uint32_t    time_stamp_;
-  int         picture_id_;
-  int         temporal_id_;
-  int         tl0_pic_id_;
-  bool        full_sync_;  // Sync flag when temporal layers are used.
-  bool        in_initial_state_;
+  uint16_t sequence_num_;
+  uint32_t time_stamp_;
+  int picture_id_;
+  int temporal_id_;
+  int tl0_pic_id_;
+  bool full_sync_;  // Sync flag when temporal layers are used.
+  bool in_initial_state_;
+
+  // Used to check references in flexible mode.
+  bool frame_decoded_[kFrameDecodedLength];
+  uint16_t frame_decoded_cleared_to_;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc b/webrtc/modules/video_coding/decoding_state_unittest.cc
index feae701a65..5f5d0d38b1 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
+++ b/webrtc/modules/video_coding/decoding_state_unittest.cc
@@ -11,11 +11,11 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/decoding_state.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/decoding_state.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 namespace webrtc {
 
@@ -446,4 +446,254 @@ TEST(TestDecodingState, PictureIdRepeat) {
   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
 }
 
+TEST(TestDecodingState, FrameContinuityFlexibleModeKeyFrame) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.isFirstPacket = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.codecSpecificHeader.codec = kRtpVideoVp9;
+
+  RTPVideoHeaderVP9& vp9_hdr = packet.codecSpecificHeader.codecHeader.VP9;
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Key frame again
+  vp9_hdr.picture_id = 11;
+  frame.Reset();
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = 12;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, FrameContinuityFlexibleModeOutOfOrderFrames) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.isFirstPacket = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.codecSpecificHeader.codec = kRtpVideoVp9;
+
+  RTPVideoHeaderVP9& vp9_hdr = packet.codecSpecificHeader.codecHeader.VP9;
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 10, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = 15;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Out of order, last id 15, this id 12, ref to 10, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 12;
+  vp9_hdr.pid_diff[0] = 2;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref 10, 12, 15, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 3;
+  vp9_hdr.pid_diff[0] = 10;
+  vp9_hdr.pid_diff[1] = 8;
+  vp9_hdr.pid_diff[2] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, FrameContinuityFlexibleModeGeneral) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.isFirstPacket = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.codecSpecificHeader.codec = kRtpVideoVp9;
+
+  RTPVideoHeaderVP9& vp9_hdr = packet.codecSpecificHeader.codecHeader.VP9;
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+
+  // Delta frame as first frame
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Key frame then delta frame
+  frame.Reset();
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  dec_state.SetState(&frame);
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.picture_id = 15;
+  vp9_hdr.pid_diff[0] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 16;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Ref to 15, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 16;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11 and 15, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 2;
+  vp9_hdr.pid_diff[0] = 9;
+  vp9_hdr.pid_diff[1] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Ref to 10, 15 and 16, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 22;
+  vp9_hdr.num_ref_pics = 3;
+  vp9_hdr.pid_diff[0] = 12;
+  vp9_hdr.pid_diff[1] = 7;
+  vp9_hdr.pid_diff[2] = 6;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Key Frame, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameKey;
+  vp9_hdr.picture_id = VCMDecodingState::kFrameDecodedLength - 2;
+  vp9_hdr.num_ref_pics = 0;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame at last index, ref to KF, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = VCMDecodingState::kFrameDecodedLength - 1;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping buffer length, ref to last index, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 0;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping start frame, ref to 0, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 20;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping start frame, ref to 10, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 23;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 13;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Key frame, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameKey;
+  vp9_hdr.picture_id = 25;
+  vp9_hdr.num_ref_pics = 0;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to KF, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = 26;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to frame previous to KF, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 30;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 30;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.cc b/webrtc/modules/video_coding/encoded_frame.cc
index d86704d632..261074ae73 100644
--- a/webrtc/modules/video_coding/main/source/encoded_frame.cc
+++ b/webrtc/modules/video_coding/encoded_frame.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/generic_encoder.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
 
 namespace webrtc {
 
@@ -24,7 +24,7 @@ VCMEncodedFrame::VCMEncodedFrame()
       _fragmentation(),
       _rotation(kVideoRotation_0),
       _rotation_set(false) {
-    _codecSpecificInfo.codecType = kVideoCodecUnknown;
+  _codecSpecificInfo.codecType = kVideoCodecUnknown;
 }
 
 VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
@@ -36,15 +36,14 @@ VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
       _fragmentation(),
       _rotation(kVideoRotation_0),
       _rotation_set(false) {
-    _codecSpecificInfo.codecType = kVideoCodecUnknown;
-    _buffer = NULL;
-    _size = 0;
-    _length = 0;
-    if (rhs._buffer != NULL)
-    {
-        VerifyAndAllocate(rhs._length);
-        memcpy(_buffer, rhs._buffer, rhs._length);
-    }
+  _codecSpecificInfo.codecType = kVideoCodecUnknown;
+  _buffer = NULL;
+  _size = 0;
+  _length = 0;
+  if (rhs._buffer != NULL) {
+    VerifyAndAllocate(rhs._length);
+    memcpy(_buffer, rhs._buffer, rhs._length);
+  }
 }
 
 VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
@@ -60,49 +59,43 @@ VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
   _buffer = NULL;
   _size = 0;
   _length = 0;
-  if (rhs._buffer != NULL)
-  {
-      VerifyAndAllocate(rhs._length);
-      memcpy(_buffer, rhs._buffer, rhs._length);
-      _length = rhs._length;
+  if (rhs._buffer != NULL) {
+    VerifyAndAllocate(rhs._length);
+    memcpy(_buffer, rhs._buffer, rhs._length);
+    _length = rhs._length;
   }
   _fragmentation.CopyFrom(rhs._fragmentation);
 }
 
-VCMEncodedFrame::~VCMEncodedFrame()
-{
-    Free();
+VCMEncodedFrame::~VCMEncodedFrame() {
+  Free();
 }
 
-void VCMEncodedFrame::Free()
-{
-    Reset();
-    if (_buffer != NULL)
-    {
-        delete [] _buffer;
-        _buffer = NULL;
-    }
+void VCMEncodedFrame::Free() {
+  Reset();
+  if (_buffer != NULL) {
+    delete[] _buffer;
+    _buffer = NULL;
+  }
 }
 
-void VCMEncodedFrame::Reset()
-{
-    _renderTimeMs = -1;
-    _timeStamp = 0;
-    _payloadType = 0;
-    _frameType = kVideoFrameDelta;
-    _encodedWidth = 0;
-    _encodedHeight = 0;
-    _completeFrame = false;
-    _missingFrame = false;
-    _length = 0;
-    _codecSpecificInfo.codecType = kVideoCodecUnknown;
-    _codec = kVideoCodecUnknown;
-    _rotation = kVideoRotation_0;
-    _rotation_set = false;
+void VCMEncodedFrame::Reset() {
+  _renderTimeMs = -1;
+  _timeStamp = 0;
+  _payloadType = 0;
+  _frameType = kVideoFrameDelta;
+  _encodedWidth = 0;
+  _encodedHeight = 0;
+  _completeFrame = false;
+  _missingFrame = false;
+  _length = 0;
+  _codecSpecificInfo.codecType = kVideoCodecUnknown;
+  _codec = kVideoCodecUnknown;
+  _rotation = kVideoRotation_0;
+  _rotation_set = false;
 }
 
-void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
-{
+void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header) {
   if (header) {
     switch (header->codec) {
       case kRtpVideoVp8: {
@@ -147,6 +140,12 @@ void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
             header->codecHeader.VP9.inter_pic_predicted;
         _codecSpecificInfo.codecSpecific.VP9.flexible_mode =
             header->codecHeader.VP9.flexible_mode;
+        _codecSpecificInfo.codecSpecific.VP9.num_ref_pics =
+            header->codecHeader.VP9.num_ref_pics;
+        for (uint8_t r = 0; r < header->codecHeader.VP9.num_ref_pics; ++r) {
+          _codecSpecificInfo.codecSpecific.VP9.p_diff[r] =
+              header->codecHeader.VP9.pid_diff[r];
+        }
         _codecSpecificInfo.codecSpecific.VP9.ss_data_available =
             header->codecHeader.VP9.ss_data_available;
         if (header->codecHeader.VP9.picture_id != kNoPictureId) {
@@ -209,21 +208,18 @@ const RTPFragmentationHeader* VCMEncodedFrame::FragmentationHeader() const {
   return &_fragmentation;
 }
 
-void VCMEncodedFrame::VerifyAndAllocate(size_t minimumSize)
-{
-    if(minimumSize > _size)
-    {
-        // create buffer of sufficient size
-        uint8_t* newBuffer = new uint8_t[minimumSize];
-        if(_buffer)
-        {
-            // copy old data
-            memcpy(newBuffer, _buffer, _size);
-            delete [] _buffer;
-        }
-        _buffer = newBuffer;
-        _size = minimumSize;
+void VCMEncodedFrame::VerifyAndAllocate(size_t minimumSize) {
+  if (minimumSize > _size) {
+    // create buffer of sufficient size
+    uint8_t* newBuffer = new uint8_t[minimumSize];
+    if (_buffer) {
+      // copy old data
+      memcpy(newBuffer, _buffer, _size);
+      delete[] _buffer;
     }
+    _buffer = newBuffer;
+    _size = minimumSize;
+  }
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/encoded_frame.h b/webrtc/modules/video_coding/encoded_frame.h
new file mode 100644
index 0000000000..9034200980
--- /dev/null
+++ b/webrtc/modules/video_coding/encoded_frame.h
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/common_video/include/video_image.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+
+namespace webrtc {
+
+class VCMEncodedFrame : protected EncodedImage {
+ public:
+  VCMEncodedFrame();
+  explicit VCMEncodedFrame(const webrtc::EncodedImage& rhs);
+  VCMEncodedFrame(const VCMEncodedFrame& rhs);
+
+  ~VCMEncodedFrame();
+  /**
+  *   Delete VideoFrame and resets members to zero
+  */
+  void Free();
+  /**
+  *   Set render time in milliseconds
+  */
+  void SetRenderTime(const int64_t renderTimeMs) {
+    _renderTimeMs = renderTimeMs;
+  }
+
+  /**
+  *   Set the encoded frame size
+  */
+  void SetEncodedSize(uint32_t width, uint32_t height) {
+    _encodedWidth = width;
+    _encodedHeight = height;
+  }
+  /**
+  *   Get the encoded image
+  */
+  const webrtc::EncodedImage& EncodedImage() const {
+    return static_cast<const webrtc::EncodedImage&>(*this);
+  }
+  /**
+  *   Get pointer to frame buffer
+  */
+  const uint8_t* Buffer() const { return _buffer; }
+  /**
+  *   Get frame length
+  */
+  size_t Length() const { return _length; }
+  /**
+  *   Get frame timestamp (90kHz)
+  */
+  uint32_t TimeStamp() const { return _timeStamp; }
+  /**
+  *   Get render time in milliseconds
+  */
+  int64_t RenderTimeMs() const { return _renderTimeMs; }
+  /**
+  *   Get frame type
+  */
+  webrtc::FrameType FrameType() const { return _frameType; }
+  /**
+  *   Get frame rotation
+  */
+  VideoRotation rotation() const { return _rotation; }
+  /**
+  *   True if this frame is complete, false otherwise
+  */
+  bool Complete() const { return _completeFrame; }
+  /**
+  *   True if there's a frame missing before this frame
+  */
+  bool MissingFrame() const { return _missingFrame; }
+  /**
+  *   Payload type of the encoded payload
+  */
+  uint8_t PayloadType() const { return _payloadType; }
+  /**
+  *   Get codec specific info.
+  *   The returned pointer is only valid as long as the VCMEncodedFrame
+  *   is valid. Also, VCMEncodedFrame owns the pointer and will delete
+  *   the object.
+  */
+  const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
+
+  const RTPFragmentationHeader* FragmentationHeader() const;
+
+ protected:
+  /**
+  * Verifies that current allocated buffer size is larger than or equal to the
+  * input size.
+  * If the current buffer size is smaller, a new allocation is made and the old
+  * buffer data
+  * is copied to the new buffer.
+  * Buffer size is updated to minimumSize.
+  */
+  void VerifyAndAllocate(size_t minimumSize);
+
+  void Reset();
+
+  void CopyCodecSpecific(const RTPVideoHeader* header);
+
+  int64_t _renderTimeMs;
+  uint8_t _payloadType;
+  bool _missingFrame;
+  CodecSpecificInfo _codecSpecificInfo;
+  webrtc::VideoCodecType _codec;
+  RTPFragmentationHeader _fragmentation;
+  VideoRotation _rotation;
+
+  // Video rotation is only set along with the last packet for each frame
+  // (same as marker bit). This |_rotation_set| is only for debugging purpose
+  // to ensure we don't set it twice for a frame.
+  bool _rotation_set;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
diff --git a/webrtc/modules/video_coding/fec_tables_xor.h b/webrtc/modules/video_coding/fec_tables_xor.h
new file mode 100644
index 0000000000..fa5bd7bde4
--- /dev/null
+++ b/webrtc/modules/video_coding/fec_tables_xor.h
@@ -0,0 +1,459 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FEC_TABLES_XOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FEC_TABLES_XOR_H_
+
+// This is a private header for media_opt_util.cc.
+// It should not be included by other files.
+
+namespace webrtc {
+
+// Table for Protection factor (code rate) of delta frames, for the XOR FEC.
+// Input is the packet loss and an effective rate (bits/frame).
+// Output is array kCodeRateXORTable[k], where k = rate_i*129 + loss_j;
+// loss_j = 0,1,..128, and rate_i varies over some range.
+static const int kSizeCodeRateXORTable = 6450;
+static const unsigned char kCodeRateXORTable[kSizeCodeRateXORTable] = {
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   11,  11,  11,  11,  11,  11,  11,  11,  11,  11,  11,  11,
+    11,  11,  11,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  51,  51,  51,  51,  51,  51,  51,  51,  51,
+    51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,
+    51,  51,  51,  51,  51,  51,  51,  51,  51,  0,   0,   0,   0,   0,   0,
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   8,   8,   8,
+    8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   30,  30,  30,
+    30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  56,  56,  56,
+    56,  56,  56,  56,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,
+    87,  87,  87,  87,  87,  87,  87,  87,  87,  78,  78,  78,  78,  78,  78,
+    78,  78,  78,  0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   6,   6,   6,   23,  23,  23,  23,  23,  23,  23,  23,  23,
+    23,  23,  23,  23,  23,  23,  44,  44,  44,  44,  44,  44,  50,  50,  50,
+    50,  50,  50,  50,  50,  50,  68,  68,  68,  68,  68,  68,  68,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  105, 105, 105, 105, 105, 105,
+    105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105,
+    105, 105, 105, 88,  88,  88,  88,  88,  88,  88,  88,  88,  0,   0,   0,
+    0,   0,   0,   0,   0,   0,   5,   5,   5,   5,   5,   5,   19,  19,  19,
+    36,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,
+    55,  55,  55,  55,  55,  55,  69,  69,  69,  69,  69,  69,  69,  69,  69,
+    75,  75,  80,  80,  80,  80,  80,  97,  97,  97,  97,  97,  97,  97,  97,
+    97,  97,  102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102,
+    102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102,
+    102, 102, 102, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116,
+    116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 100, 100, 100,
+    100, 100, 100, 100, 100, 100, 0,   0,   0,   0,   0,   0,   0,   0,   4,
+    16,  16,  16,  16,  16,  16,  30,  35,  35,  47,  58,  58,  58,  58,  58,
+    58,  58,  58,  58,  58,  58,  58,  58,  58,  63,  63,  63,  63,  63,  63,
+    77,  77,  77,  77,  77,  77,  77,  82,  82,  82,  82,  94,  94,  94,  94,
+    94,  105, 105, 105, 105, 110, 110, 110, 110, 110, 110, 122, 122, 122, 122,
+    122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122,
+    122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 115, 115, 115, 115, 115, 115, 115, 115, 115,
+    0,   0,   0,   0,   0,   0,   0,   4,   14,  27,  27,  27,  27,  27,  31,
+    41,  52,  52,  56,  69,  69,  69,  69,  69,  69,  69,  69,  69,  69,  69,
+    69,  69,  69,  69,  69,  69,  69,  69,  69,  79,  79,  79,  79,  83,  83,
+    83,  94,  94,  94,  94,  106, 106, 106, 106, 106, 115, 115, 115, 115, 125,
+    125, 125, 125, 125, 125, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   0,   0,   3,   3,
+    3,   17,  28,  38,  38,  38,  38,  38,  47,  51,  63,  63,  63,  72,  72,
+    72,  72,  72,  72,  72,  76,  76,  76,  76,  80,  80,  80,  80,  80,  80,
+    80,  80,  80,  84,  84,  84,  84,  93,  93,  93,  105, 105, 105, 105, 114,
+    114, 114, 114, 114, 124, 124, 124, 124, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   0,   0,   0,   12,  12,  12,  35,  43,  47,  47,  47,
+    47,  47,  58,  58,  66,  66,  66,  70,  70,  70,  70,  70,  73,  73,  82,
+    82,  82,  86,  94,  94,  94,  94,  94,  94,  94,  94,  94,  94,  94,  94,
+    94,  105, 105, 105, 114, 114, 114, 114, 117, 117, 117, 117, 117, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   0,
+    0,   24,  24,  24,  49,  53,  53,  53,  53,  53,  53,  61,  61,  64,  64,
+    64,  64,  70,  70,  70,  70,  78,  78,  88,  88,  88,  96,  106, 106, 106,
+    106, 106, 106, 106, 106, 106, 106, 112, 112, 112, 120, 120, 120, 124, 124,
+    124, 124, 124, 124, 124, 124, 124, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   0,   0,   5,   36,  36,  36,  55,  55,
+    55,  55,  55,  55,  55,  58,  58,  58,  58,  58,  64,  78,  78,  78,  78,
+    87,  87,  94,  94,  94,  103, 110, 110, 110, 110, 110, 110, 110, 110, 116,
+    116, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   0,   0,   18,  43,  43,  43,  53,  53,  53,  53,  53,  53,  53,  53,
+    58,  58,  58,  58,  71,  87,  87,  87,  87,  94,  94,  97,  97,  97,  109,
+    111, 111, 111, 111, 111, 111, 111, 111, 125, 125, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   0,   31,  46,  46,
+    46,  48,  48,  48,  48,  48,  48,  48,  48,  66,  66,  66,  66,  80,  93,
+    93,  93,  93,  95,  95,  95,  95,  100, 115, 115, 115, 115, 115, 115, 115,
+    115, 115, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   0,   4,   40,  45,  45,  45,  45,  45,  45,  45,  45,
+    49,  49,  49,  74,  74,  74,  74,  86,  90,  90,  90,  90,  95,  95,  95,
+    95,  106, 120, 120, 120, 120, 120, 120, 120, 120, 120, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   14,
+    42,  42,  42,  42,  42,  42,  42,  42,  46,  56,  56,  56,  80,  80,  80,
+    80,  84,  84,  84,  84,  88,  99,  99,  99,  99,  111, 122, 122, 122, 122,
+    122, 122, 122, 122, 122, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   17,  35,  35,  52,  59,  59,  70,  70,  76,  76,  76,
+    99,  105, 105, 105, 105, 105, 111, 111, 111, 111, 111, 111, 111, 121, 127,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   17,  34,
+    36,  51,  61,  62,  70,  70,  80,  80,  80,  93,  103, 103, 103, 103, 103,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
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+    127, 127, 127, 127, 127, 127, 0,   16,  33,  39,  50,  59,  65,  72,  72,
+    82,  82,  82,  91,  100, 100, 100, 100, 100, 109, 109, 109, 109, 109, 121,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   16,  32,  43,  48,  54,  66,  75,  75,  81,  83,  83,  92,  97,  97,
+    97,  99,  99,  105, 105, 105, 105, 105, 123, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   15,  31,  46,  47,  49,
+    69,  77,  77,  81,  85,  85,  93,  95,  95,  95,  100, 100, 102, 102, 102,
+    102, 102, 120, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   15,  30,  46,  48,  48,  70,  75,  79,  82,  87,  87,
+    92,  94,  94,  94,  103, 103, 103, 103, 103, 104, 104, 115, 120, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   15,  30,
+    45,  50,  50,  68,  70,  80,  85,  89,  89,  90,  95,  95,  95,  104, 104,
+    104, 104, 104, 109, 109, 112, 114, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   14,  29,  44,  54,  54,  64,  64,  83,
+    87,  88,  88,  88,  98,  98,  98,  103, 103, 103, 103, 103, 113, 113, 113,
+    113, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   14,  29,  43,  56,  56,  61,  61,  84,  85,  88,  88,  88,  100, 100,
+    100, 102, 102, 102, 102, 102, 113, 116, 116, 116, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   14,  28,  42,  57,  57,
+    62,  62,  80,  80,  91,  91,  91,  100, 100, 100, 100, 100, 100, 100, 100,
+    109, 119, 119, 119, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   14,  28,  42,  56,  56,  65,  66,  76,  76,  92,  92,
+    92,  97,  97,  97,  101, 101, 101, 101, 101, 106, 121, 121, 121, 126, 126,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   13,  27,
+    41,  55,  55,  67,  72,  74,  74,  90,  90,  90,  91,  91,  91,  105, 105,
+    105, 105, 105, 107, 122, 122, 122, 123, 123, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   13,  27,  40,  54,  54,  67,  76,  76,
+    76,  85,  85,  85,  85,  85,  85,  112, 112, 112, 112, 112, 112, 121, 121,
+    121, 121, 121, 126, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_FEC_TABLES_XOR_H_
diff --git a/webrtc/modules/video_coding/frame_buffer.cc b/webrtc/modules/video_coding/frame_buffer.cc
new file mode 100644
index 0000000000..b6ddeda4e7
--- /dev/null
+++ b/webrtc/modules/video_coding/frame_buffer.cc
@@ -0,0 +1,270 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/frame_buffer.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/packet.h"
+
+namespace webrtc {
+
+VCMFrameBuffer::VCMFrameBuffer()
+    : _state(kStateEmpty), _nackCount(0), _latestPacketTimeMs(-1) {}
+
+VCMFrameBuffer::~VCMFrameBuffer() {}
+
+VCMFrameBuffer::VCMFrameBuffer(const VCMFrameBuffer& rhs)
+    : VCMEncodedFrame(rhs),
+      _state(rhs._state),
+      _sessionInfo(),
+      _nackCount(rhs._nackCount),
+      _latestPacketTimeMs(rhs._latestPacketTimeMs) {
+  _sessionInfo = rhs._sessionInfo;
+  _sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
+}
+
+webrtc::FrameType VCMFrameBuffer::FrameType() const {
+  return _sessionInfo.FrameType();
+}
+
+int32_t VCMFrameBuffer::GetLowSeqNum() const {
+  return _sessionInfo.LowSequenceNumber();
+}
+
+int32_t VCMFrameBuffer::GetHighSeqNum() const {
+  return _sessionInfo.HighSequenceNumber();
+}
+
+int VCMFrameBuffer::PictureId() const {
+  return _sessionInfo.PictureId();
+}
+
+int VCMFrameBuffer::TemporalId() const {
+  return _sessionInfo.TemporalId();
+}
+
+bool VCMFrameBuffer::LayerSync() const {
+  return _sessionInfo.LayerSync();
+}
+
+int VCMFrameBuffer::Tl0PicId() const {
+  return _sessionInfo.Tl0PicId();
+}
+
+bool VCMFrameBuffer::NonReference() const {
+  return _sessionInfo.NonReference();
+}
+
+void VCMFrameBuffer::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
+  _sessionInfo.SetGofInfo(gof_info, idx);
+  // TODO(asapersson): Consider adding hdr->VP9.ref_picture_id for testing.
+  _codecSpecificInfo.codecSpecific.VP9.temporal_idx =
+      gof_info.temporal_idx[idx];
+  _codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
+      gof_info.temporal_up_switch[idx];
+}
+
+bool VCMFrameBuffer::IsSessionComplete() const {
+  return _sessionInfo.complete();
+}
+
+// Insert packet
+VCMFrameBufferEnum VCMFrameBuffer::InsertPacket(
+    const VCMPacket& packet,
+    int64_t timeInMs,
+    VCMDecodeErrorMode decode_error_mode,
+    const FrameData& frame_data) {
+  assert(!(NULL == packet.dataPtr && packet.sizeBytes > 0));
+  if (packet.dataPtr != NULL) {
+    _payloadType = packet.payloadType;
+  }
+
+  if (kStateEmpty == _state) {
+    // First packet (empty and/or media) inserted into this frame.
+    // store some info and set some initial values.
+    _timeStamp = packet.timestamp;
+    // We only take the ntp timestamp of the first packet of a frame.
+    ntp_time_ms_ = packet.ntp_time_ms_;
+    _codec = packet.codec;
+    if (packet.frameType != kEmptyFrame) {
+      // first media packet
+      SetState(kStateIncomplete);
+    }
+  }
+
+  uint32_t requiredSizeBytes =
+      Length() + packet.sizeBytes +
+      (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
+  if (requiredSizeBytes >= _size) {
+    const uint8_t* prevBuffer = _buffer;
+    const uint32_t increments =
+        requiredSizeBytes / kBufferIncStepSizeBytes +
+        (requiredSizeBytes % kBufferIncStepSizeBytes > 0);
+    const uint32_t newSize = _size + increments * kBufferIncStepSizeBytes;
+    if (newSize > kMaxJBFrameSizeBytes) {
+      LOG(LS_ERROR) << "Failed to insert packet due to frame being too "
+                       "big.";
+      return kSizeError;
+    }
+    VerifyAndAllocate(newSize);
+    _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
+  }
+
+  if (packet.width > 0 && packet.height > 0) {
+    _encodedWidth = packet.width;
+    _encodedHeight = packet.height;
+  }
+
+  // Don't copy payload specific data for empty packets (e.g padding packets).
+  if (packet.sizeBytes > 0)
+    CopyCodecSpecific(&packet.codecSpecificHeader);
+
+  int retVal =
+      _sessionInfo.InsertPacket(packet, _buffer, decode_error_mode, frame_data);
+  if (retVal == -1) {
+    return kSizeError;
+  } else if (retVal == -2) {
+    return kDuplicatePacket;
+  } else if (retVal == -3) {
+    return kOutOfBoundsPacket;
+  }
+  // update length
+  _length = Length() + static_cast<uint32_t>(retVal);
+
+  _latestPacketTimeMs = timeInMs;
+
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+  // ts_126114v120700p.pdf Section 7.4.5.
+  // The MTSI client shall add the payload bytes as defined in this clause
+  // onto the last RTP packet in each group of packets which make up a key
+  // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
+  // (HEVC)).
+  if (packet.markerBit) {
+    RTC_DCHECK(!_rotation_set);
+    _rotation = packet.codecSpecificHeader.rotation;
+    _rotation_set = true;
+  }
+
+  if (_sessionInfo.complete()) {
+    SetState(kStateComplete);
+    return kCompleteSession;
+  } else if (_sessionInfo.decodable()) {
+    SetState(kStateDecodable);
+    return kDecodableSession;
+  }
+  return kIncomplete;
+}
+
+int64_t VCMFrameBuffer::LatestPacketTimeMs() const {
+  return _latestPacketTimeMs;
+}
+
+void VCMFrameBuffer::IncrementNackCount() {
+  _nackCount++;
+}
+
+int16_t VCMFrameBuffer::GetNackCount() const {
+  return _nackCount;
+}
+
+bool VCMFrameBuffer::HaveFirstPacket() const {
+  return _sessionInfo.HaveFirstPacket();
+}
+
+bool VCMFrameBuffer::HaveLastPacket() const {
+  return _sessionInfo.HaveLastPacket();
+}
+
+int VCMFrameBuffer::NumPackets() const {
+  return _sessionInfo.NumPackets();
+}
+
+void VCMFrameBuffer::Reset() {
+  _length = 0;
+  _timeStamp = 0;
+  _sessionInfo.Reset();
+  _payloadType = 0;
+  _nackCount = 0;
+  _latestPacketTimeMs = -1;
+  _state = kStateEmpty;
+  VCMEncodedFrame::Reset();
+}
+
+// Set state of frame
+void VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state) {
+  if (_state == state) {
+    return;
+  }
+  switch (state) {
+    case kStateIncomplete:
+      // we can go to this state from state kStateEmpty
+      assert(_state == kStateEmpty);
+
+      // Do nothing, we received a packet
+      break;
+
+    case kStateComplete:
+      assert(_state == kStateEmpty || _state == kStateIncomplete ||
+             _state == kStateDecodable);
+
+      break;
+
+    case kStateEmpty:
+      // Should only be set to empty through Reset().
+      assert(false);
+      break;
+
+    case kStateDecodable:
+      assert(_state == kStateEmpty || _state == kStateIncomplete);
+      break;
+  }
+  _state = state;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum VCMFrameBuffer::GetState() const {
+  return _state;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum VCMFrameBuffer::GetState(uint32_t& timeStamp) const {
+  timeStamp = TimeStamp();
+  return GetState();
+}
+
+bool VCMFrameBuffer::IsRetransmitted() const {
+  return _sessionInfo.session_nack();
+}
+
+void VCMFrameBuffer::PrepareForDecode(bool continuous) {
+#ifdef INDEPENDENT_PARTITIONS
+  if (_codec == kVideoCodecVP8) {
+    _length = _sessionInfo.BuildVP8FragmentationHeader(_buffer, _length,
+                                                       &_fragmentation);
+  } else {
+    size_t bytes_removed = _sessionInfo.MakeDecodable();
+    _length -= bytes_removed;
+  }
+#else
+  size_t bytes_removed = _sessionInfo.MakeDecodable();
+  _length -= bytes_removed;
+#endif
+  // Transfer frame information to EncodedFrame and create any codec
+  // specific information.
+  _frameType = _sessionInfo.FrameType();
+  _completeFrame = _sessionInfo.complete();
+  _missingFrame = !continuous;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/frame_buffer.h b/webrtc/modules/video_coding/frame_buffer.h
index ab4ff6574e..f5a707efe4 100644
--- a/webrtc/modules/video_coding/main/source/frame_buffer.h
+++ b/webrtc/modules/video_coding/frame_buffer.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/session_info.h"
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/session_info.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -89,4 +89,4 @@ class VCMFrameBuffer : public VCMEncodedFrame {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
diff --git a/webrtc/modules/video_coding/generic_decoder.cc b/webrtc/modules/video_coding/generic_decoder.cc
new file mode 100644
index 0000000000..5cbe0f5ba0
--- /dev/null
+++ b/webrtc/modules/video_coding/generic_decoder.cc
@@ -0,0 +1,192 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/generic_decoder.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming* timing,
+                                                 Clock* clock)
+    : _critSect(CriticalSectionWrapper::CreateCriticalSection()),
+      _clock(clock),
+      _receiveCallback(NULL),
+      _timing(timing),
+      _timestampMap(kDecoderFrameMemoryLength),
+      _lastReceivedPictureID(0) {}
+
+VCMDecodedFrameCallback::~VCMDecodedFrameCallback() {
+  delete _critSect;
+}
+
+void VCMDecodedFrameCallback::SetUserReceiveCallback(
+    VCMReceiveCallback* receiveCallback) {
+  CriticalSectionScoped cs(_critSect);
+  _receiveCallback = receiveCallback;
+}
+
+VCMReceiveCallback* VCMDecodedFrameCallback::UserReceiveCallback() {
+  CriticalSectionScoped cs(_critSect);
+  return _receiveCallback;
+}
+
+int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage) {
+  return Decoded(decodedImage, -1);
+}
+
+int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
+                                         int64_t decode_time_ms) {
+  TRACE_EVENT_INSTANT1("webrtc", "VCMDecodedFrameCallback::Decoded",
+                       "timestamp", decodedImage.timestamp());
+  // TODO(holmer): We should improve this so that we can handle multiple
+  // callbacks from one call to Decode().
+  VCMFrameInformation* frameInfo;
+  VCMReceiveCallback* callback;
+  {
+    CriticalSectionScoped cs(_critSect);
+    frameInfo = _timestampMap.Pop(decodedImage.timestamp());
+    callback = _receiveCallback;
+  }
+
+  if (frameInfo == NULL) {
+    LOG(LS_WARNING) << "Too many frames backed up in the decoder, dropping "
+                       "this one.";
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  const int64_t now_ms = _clock->TimeInMilliseconds();
+  if (decode_time_ms < 0) {
+    decode_time_ms =
+        static_cast<int32_t>(now_ms - frameInfo->decodeStartTimeMs);
+  }
+  _timing->StopDecodeTimer(decodedImage.timestamp(), decode_time_ms, now_ms,
+                           frameInfo->renderTimeMs);
+
+  if (callback != NULL) {
+    decodedImage.set_render_time_ms(frameInfo->renderTimeMs);
+    decodedImage.set_rotation(frameInfo->rotation);
+    callback->FrameToRender(decodedImage);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(
+    const uint64_t pictureId) {
+  CriticalSectionScoped cs(_critSect);
+  if (_receiveCallback != NULL) {
+    return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
+  }
+  return -1;
+}
+
+int32_t VCMDecodedFrameCallback::ReceivedDecodedFrame(
+    const uint64_t pictureId) {
+  _lastReceivedPictureID = pictureId;
+  return 0;
+}
+
+uint64_t VCMDecodedFrameCallback::LastReceivedPictureID() const {
+  return _lastReceivedPictureID;
+}
+
+void VCMDecodedFrameCallback::OnDecoderImplementationName(
+    const char* implementation_name) {
+  CriticalSectionScoped cs(_critSect);
+  if (_receiveCallback)
+    _receiveCallback->OnDecoderImplementationName(implementation_name);
+}
+
+void VCMDecodedFrameCallback::Map(uint32_t timestamp,
+                                  VCMFrameInformation* frameInfo) {
+  CriticalSectionScoped cs(_critSect);
+  _timestampMap.Add(timestamp, frameInfo);
+}
+
+int32_t VCMDecodedFrameCallback::Pop(uint32_t timestamp) {
+  CriticalSectionScoped cs(_critSect);
+  if (_timestampMap.Pop(timestamp) == NULL) {
+    return VCM_GENERAL_ERROR;
+  }
+  return VCM_OK;
+}
+
+VCMGenericDecoder::VCMGenericDecoder(VideoDecoder* decoder, bool isExternal)
+    : _callback(NULL),
+      _frameInfos(),
+      _nextFrameInfoIdx(0),
+      _decoder(decoder),
+      _codecType(kVideoCodecUnknown),
+      _isExternal(isExternal),
+      _keyFrameDecoded(false) {}
+
+VCMGenericDecoder::~VCMGenericDecoder() {}
+
+int32_t VCMGenericDecoder::InitDecode(const VideoCodec* settings,
+                                      int32_t numberOfCores) {
+  TRACE_EVENT0("webrtc", "VCMGenericDecoder::InitDecode");
+  _codecType = settings->codecType;
+
+  return _decoder->InitDecode(settings, numberOfCores);
+}
+
+int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame, int64_t nowMs) {
+    TRACE_EVENT1("webrtc", "VCMGenericDecoder::Decode", "timestamp",
+                 frame.EncodedImage()._timeStamp);
+    _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
+    _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
+    _frameInfos[_nextFrameInfoIdx].rotation = frame.rotation();
+    _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
+
+    _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
+    int32_t ret = _decoder->Decode(frame.EncodedImage(), frame.MissingFrame(),
+                                   frame.FragmentationHeader(),
+                                   frame.CodecSpecific(), frame.RenderTimeMs());
+
+    _callback->OnDecoderImplementationName(_decoder->ImplementationName());
+    if (ret < WEBRTC_VIDEO_CODEC_OK) {
+        LOG(LS_WARNING) << "Failed to decode frame with timestamp "
+                        << frame.TimeStamp() << ", error code: " << ret;
+        _callback->Pop(frame.TimeStamp());
+        return ret;
+    } else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
+               ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI) {
+        // No output
+        _callback->Pop(frame.TimeStamp());
+    }
+    return ret;
+}
+
+int32_t VCMGenericDecoder::Release() {
+  return _decoder->Release();
+}
+
+int32_t VCMGenericDecoder::Reset() {
+  return _decoder->Reset();
+}
+
+int32_t VCMGenericDecoder::RegisterDecodeCompleteCallback(
+    VCMDecodedFrameCallback* callback) {
+  _callback = callback;
+  return _decoder->RegisterDecodeCompleteCallback(callback);
+}
+
+bool VCMGenericDecoder::External() const {
+  return _isExternal;
+}
+
+bool VCMGenericDecoder::PrefersLateDecoding() const {
+  return _decoder->PrefersLateDecoding();
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/generic_decoder.h b/webrtc/modules/video_coding/generic_decoder.h
new file mode 100644
index 0000000000..67ceabfc53
--- /dev/null
+++ b/webrtc/modules/video_coding/generic_decoder.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/timestamp_map.h"
+#include "webrtc/modules/video_coding/timing.h"
+
+namespace webrtc {
+
+class VCMReceiveCallback;
+
+enum { kDecoderFrameMemoryLength = 10 };
+
+struct VCMFrameInformation {
+  int64_t renderTimeMs;
+  int64_t decodeStartTimeMs;
+  void* userData;
+  VideoRotation rotation;
+};
+
+class VCMDecodedFrameCallback : public DecodedImageCallback {
+ public:
+  VCMDecodedFrameCallback(VCMTiming* timing, Clock* clock);
+    virtual ~VCMDecodedFrameCallback();
+    void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
+    VCMReceiveCallback* UserReceiveCallback();
+
+    virtual int32_t Decoded(VideoFrame& decodedImage);  // NOLINT
+    virtual int32_t Decoded(VideoFrame& decodedImage,   // NOLINT
+                            int64_t decode_time_ms);
+    virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);
+    virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId);
+
+    uint64_t LastReceivedPictureID() const;
+    void OnDecoderImplementationName(const char* implementation_name);
+
+    void Map(uint32_t timestamp, VCMFrameInformation* frameInfo);
+    int32_t Pop(uint32_t timestamp);
+
+ private:
+    // Protect |_receiveCallback| and |_timestampMap|.
+    CriticalSectionWrapper* _critSect;
+    Clock* _clock;
+    VCMReceiveCallback* _receiveCallback GUARDED_BY(_critSect);
+    VCMTiming* _timing;
+    VCMTimestampMap _timestampMap GUARDED_BY(_critSect);
+    uint64_t _lastReceivedPictureID;
+};
+
+class VCMGenericDecoder {
+  friend class VCMCodecDataBase;
+
+ public:
+  explicit VCMGenericDecoder(VideoDecoder* decoder, bool isExternal = false);
+  ~VCMGenericDecoder();
+
+  /**
+  * Initialize the decoder with the information from the VideoCodec
+  */
+  int32_t InitDecode(const VideoCodec* settings, int32_t numberOfCores);
+
+  /**
+  * Decode to a raw I420 frame,
+  *
+  * inputVideoBuffer reference to encoded video frame
+  */
+  int32_t Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
+
+  /**
+  * Free the decoder memory
+  */
+  int32_t Release();
+
+  /**
+  * Reset the decoder state, prepare for a new call
+  */
+  int32_t Reset();
+
+  /**
+  * Set decode callback. Deregistering while decoding is illegal.
+  */
+  int32_t RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
+
+  bool External() const;
+  bool PrefersLateDecoding() const;
+
+ private:
+  VCMDecodedFrameCallback* _callback;
+  VCMFrameInformation _frameInfos[kDecoderFrameMemoryLength];
+  uint32_t _nextFrameInfoIdx;
+  VideoDecoder* const _decoder;
+  VideoCodecType _codecType;
+  bool _isExternal;
+  bool _keyFrameDecoded;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
diff --git a/webrtc/modules/video_coding/main/source/generic_encoder.cc b/webrtc/modules/video_coding/generic_encoder.cc
index de196040f0..c7444ce99f 100644
--- a/webrtc/modules/video_coding/main/source/generic_encoder.cc
+++ b/webrtc/modules/video_coding/generic_encoder.cc
@@ -8,13 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/modules/video_coding/generic_encoder.h"
+
+#include <vector>
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 namespace {
@@ -27,8 +31,7 @@ void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
       rtp->codec = kRtpVideoVp8;
       rtp->codecHeader.VP8.InitRTPVideoHeaderVP8();
       rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
-      rtp->codecHeader.VP8.nonReference =
-          info->codecSpecific.VP8.nonReference;
+      rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference;
       rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
       rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync;
       rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx;
@@ -54,11 +57,9 @@ void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
       rtp->codecHeader.VP9.inter_layer_predicted =
           info->codecSpecific.VP9.inter_layer_predicted;
       rtp->codecHeader.VP9.gof_idx = info->codecSpecific.VP9.gof_idx;
-
-      // Packetizer needs to know the number of spatial layers to correctly set
-      // the marker bit, even when the number won't be written in the packet.
       rtp->codecHeader.VP9.num_spatial_layers =
           info->codecSpecific.VP9.num_spatial_layers;
+
       if (info->codecSpecific.VP9.ss_data_available) {
         rtp->codecHeader.VP9.spatial_layer_resolution_present =
             info->codecSpecific.VP9.spatial_layer_resolution_present;
@@ -71,6 +72,10 @@ void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
         }
         rtp->codecHeader.VP9.gof.CopyGofInfoVP9(info->codecSpecific.VP9.gof);
       }
+
+      rtp->codecHeader.VP9.num_ref_pics = info->codecSpecific.VP9.num_ref_pics;
+      for (int i = 0; i < info->codecSpecific.VP9.num_ref_pics; ++i)
+        rtp->codecHeader.VP9.pid_diff[i] = info->codecSpecific.VP9.p_diff[i];
       return;
     }
     case kVideoCodecH264:
@@ -86,7 +91,7 @@ void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
 }
 }  // namespace
 
-//#define DEBUG_ENCODER_BIT_STREAM
+// #define DEBUG_ENCODER_BIT_STREAM
 
 VCMGenericEncoder::VCMGenericEncoder(
     VideoEncoder* encoder,
@@ -110,6 +115,7 @@ int32_t VCMGenericEncoder::Release() {
 int32_t VCMGenericEncoder::InitEncode(const VideoCodec* settings,
                                       int32_t numberOfCores,
                                       size_t maxPayloadSize) {
+  TRACE_EVENT0("webrtc", "VCMGenericEncoder::InitEncode");
   {
     rtc::CritScope lock(&params_lock_);
     encoder_params_.target_bitrate = settings->startBitrate * 1000;
@@ -130,6 +136,9 @@ int32_t VCMGenericEncoder::InitEncode(const VideoCodec* settings,
 int32_t VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
                                   const CodecSpecificInfo* codecSpecificInfo,
                                   const std::vector<FrameType>& frameTypes) {
+  TRACE_EVENT1("webrtc", "VCMGenericEncoder::Encode", "timestamp",
+               inputFrame.timestamp());
+
   for (FrameType frame_type : frameTypes)
     RTC_DCHECK(frame_type == kVideoFrameKey || frame_type == kVideoFrameDelta);
 
@@ -143,6 +152,12 @@ int32_t VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
   vcm_encoded_frame_callback_->SetRotation(rotation_);
 
   int32_t result = encoder_->Encode(inputFrame, codecSpecificInfo, &frameTypes);
+
+  if (vcm_encoded_frame_callback_) {
+    vcm_encoded_frame_callback_->SignalLastEncoderImplementationUsed(
+        encoder_->ImplementationName());
+  }
+
   if (is_screenshare_ &&
       result == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT) {
     // Target bitrate exceeded, encoder state has been reset - try again.
@@ -182,10 +197,8 @@ EncoderParameters VCMGenericEncoder::GetEncoderParameters() const {
   return encoder_params_;
 }
 
-int32_t
-VCMGenericEncoder::SetPeriodicKeyFrames(bool enable)
-{
-    return encoder_->SetPeriodicKeyFrames(enable);
+int32_t VCMGenericEncoder::SetPeriodicKeyFrames(bool enable) {
+  return encoder_->SetPeriodicKeyFrames(enable);
 }
 
 int32_t VCMGenericEncoder::RequestFrame(
@@ -194,10 +207,8 @@ int32_t VCMGenericEncoder::RequestFrame(
   return encoder_->Encode(image, NULL, &frame_types);
 }
 
-bool
-VCMGenericEncoder::InternalSource() const
-{
-    return internal_source_;
+bool VCMGenericEncoder::InternalSource() const {
+  return internal_source_;
 }
 
 void VCMGenericEncoder::OnDroppedFrame() {
@@ -212,12 +223,12 @@ int VCMGenericEncoder::GetTargetFramerate() {
   return encoder_->GetTargetFramerate();
 }
 
- /***************************
-  * Callback Implementation
-  ***************************/
+/***************************
+ * Callback Implementation
+ ***************************/
 VCMEncodedFrameCallback::VCMEncodedFrameCallback(
     EncodedImageCallback* post_encode_callback)
-    : _sendCallback(),
+    : send_callback_(),
       _mediaOpt(NULL),
       _payloadType(0),
       _internalSource(false),
@@ -229,39 +240,37 @@ VCMEncodedFrameCallback::VCMEncodedFrameCallback(
 #endif
 {
 #ifdef DEBUG_ENCODER_BIT_STREAM
-    _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
+  _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
 #endif
 }
 
-VCMEncodedFrameCallback::~VCMEncodedFrameCallback()
-{
+VCMEncodedFrameCallback::~VCMEncodedFrameCallback() {
 #ifdef DEBUG_ENCODER_BIT_STREAM
-    fclose(_bitStreamAfterEncoder);
+  fclose(_bitStreamAfterEncoder);
 #endif
 }
 
-int32_t
-VCMEncodedFrameCallback::SetTransportCallback(VCMPacketizationCallback* transport)
-{
-    _sendCallback = transport;
-    return VCM_OK;
+int32_t VCMEncodedFrameCallback::SetTransportCallback(
+    VCMPacketizationCallback* transport) {
+  send_callback_ = transport;
+  return VCM_OK;
 }
 
 int32_t VCMEncodedFrameCallback::Encoded(
-    const EncodedImage& encodedImage,
+    const EncodedImage& encoded_image,
     const CodecSpecificInfo* codecSpecificInfo,
     const RTPFragmentationHeader* fragmentationHeader) {
-  RTC_DCHECK(encodedImage._frameType == kVideoFrameKey ||
-             encodedImage._frameType == kVideoFrameDelta);
-  post_encode_callback_->Encoded(encodedImage, NULL, NULL);
+  TRACE_EVENT_INSTANT1("webrtc", "VCMEncodedFrameCallback::Encoded",
+                       "timestamp", encoded_image._timeStamp);
+  post_encode_callback_->Encoded(encoded_image, NULL, NULL);
 
-  if (_sendCallback == NULL) {
+  if (send_callback_ == NULL) {
     return VCM_UNINITIALIZED;
   }
 
 #ifdef DEBUG_ENCODER_BIT_STREAM
   if (_bitStreamAfterEncoder != NULL) {
-    fwrite(encodedImage._buffer, 1, encodedImage._length,
+    fwrite(encoded_image._buffer, 1, encoded_image._length,
            _bitStreamAfterEncoder);
   }
 #endif
@@ -274,25 +283,29 @@ int32_t VCMEncodedFrameCallback::Encoded(
   }
   rtpVideoHeader.rotation = _rotation;
 
-  int32_t callbackReturn = _sendCallback->SendData(
-      _payloadType, encodedImage, *fragmentationHeader, rtpVideoHeaderPtr);
+  int32_t callbackReturn = send_callback_->SendData(
+      _payloadType, encoded_image, *fragmentationHeader, rtpVideoHeaderPtr);
   if (callbackReturn < 0) {
     return callbackReturn;
   }
 
   if (_mediaOpt != NULL) {
-    _mediaOpt->UpdateWithEncodedData(encodedImage);
+    _mediaOpt->UpdateWithEncodedData(encoded_image);
     if (_internalSource)
       return _mediaOpt->DropFrame();  // Signal to encoder to drop next frame.
   }
   return VCM_OK;
 }
 
-void
-VCMEncodedFrameCallback::SetMediaOpt(
-    media_optimization::MediaOptimization *mediaOpt)
-{
-    _mediaOpt = mediaOpt;
+void VCMEncodedFrameCallback::SetMediaOpt(
+    media_optimization::MediaOptimization* mediaOpt) {
+  _mediaOpt = mediaOpt;
+}
+
+void VCMEncodedFrameCallback::SignalLastEncoderImplementationUsed(
+    const char* implementation_name) {
+  if (send_callback_)
+    send_callback_->OnEncoderImplementationName(implementation_name);
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/generic_encoder.h b/webrtc/modules/video_coding/generic_encoder.h
new file mode 100644
index 0000000000..f739edb44f
--- /dev/null
+++ b/webrtc/modules/video_coding/generic_encoder.h
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+
+#include <stdio.h>
+#include <vector>
+
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+namespace media_optimization {
+class MediaOptimization;
+}  // namespace media_optimization
+
+struct EncoderParameters {
+  uint32_t target_bitrate;
+  uint8_t loss_rate;
+  int64_t rtt;
+  uint32_t input_frame_rate;
+};
+
+/*************************************/
+/* VCMEncodeFrameCallback class     */
+/***********************************/
+class VCMEncodedFrameCallback : public EncodedImageCallback {
+ public:
+    explicit VCMEncodedFrameCallback(
+      EncodedImageCallback* post_encode_callback);
+    virtual ~VCMEncodedFrameCallback();
+
+    /*
+    * Callback implementation - codec encode complete
+    */
+    int32_t Encoded(
+        const EncodedImage& encodedImage,
+        const CodecSpecificInfo* codecSpecificInfo = NULL,
+        const RTPFragmentationHeader* fragmentationHeader = NULL);
+    /*
+    * Callback implementation - generic encoder encode complete
+    */
+    int32_t SetTransportCallback(VCMPacketizationCallback* transport);
+    /**
+    * Set media Optimization
+    */
+    void SetMediaOpt(media_optimization::MediaOptimization* mediaOpt);
+
+    void SetPayloadType(uint8_t payloadType) {
+      _payloadType = payloadType;
+    }
+
+    void SetInternalSource(bool internalSource) {
+      _internalSource = internalSource;
+    }
+
+    void SetRotation(VideoRotation rotation) { _rotation = rotation; }
+    void SignalLastEncoderImplementationUsed(
+        const char* encoder_implementation_name);
+
+ private:
+    VCMPacketizationCallback* send_callback_;
+    media_optimization::MediaOptimization* _mediaOpt;
+    uint8_t _payloadType;
+    bool _internalSource;
+    VideoRotation _rotation;
+
+    EncodedImageCallback* post_encode_callback_;
+
+#ifdef DEBUG_ENCODER_BIT_STREAM
+    FILE* _bitStreamAfterEncoder;
+#endif
+};  // end of VCMEncodeFrameCallback class
+
+/******************************/
+/* VCMGenericEncoder class    */
+/******************************/
+class VCMGenericEncoder {
+  friend class VCMCodecDataBase;
+
+ public:
+  VCMGenericEncoder(VideoEncoder* encoder,
+                    VideoEncoderRateObserver* rate_observer,
+                    VCMEncodedFrameCallback* encoded_frame_callback,
+                    bool internalSource);
+  ~VCMGenericEncoder();
+  /**
+  * Free encoder memory
+  */
+  int32_t Release();
+  /**
+  * Initialize the encoder with the information from the VideoCodec
+  */
+  int32_t InitEncode(const VideoCodec* settings,
+                     int32_t numberOfCores,
+                     size_t maxPayloadSize);
+  /**
+  * Encode raw image
+  * inputFrame        : Frame containing raw image
+  * codecSpecificInfo : Specific codec data
+  * cameraFrameRate   : Request or information from the remote side
+  * frameType         : The requested frame type to encode
+  */
+  int32_t Encode(const VideoFrame& inputFrame,
+                 const CodecSpecificInfo* codecSpecificInfo,
+                 const std::vector<FrameType>& frameTypes);
+
+  void SetEncoderParameters(const EncoderParameters& params);
+  EncoderParameters GetEncoderParameters() const;
+
+  int32_t SetPeriodicKeyFrames(bool enable);
+
+  int32_t RequestFrame(const std::vector<FrameType>& frame_types);
+
+  bool InternalSource() const;
+
+  void OnDroppedFrame();
+
+  bool SupportsNativeHandle() const;
+
+  int GetTargetFramerate();
+
+ private:
+  VideoEncoder* const encoder_;
+  VideoEncoderRateObserver* const rate_observer_;
+  VCMEncodedFrameCallback* const vcm_encoded_frame_callback_;
+  const bool internal_source_;
+  mutable rtc::CriticalSection params_lock_;
+  EncoderParameters encoder_params_ GUARDED_BY(params_lock_);
+  VideoRotation rotation_;
+  bool is_screenshare_;
+};  // end of VCMGenericEncoder class
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
diff --git a/webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h b/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
index 302d4a3a13..0185dae333 100644
--- a/webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h
+++ b/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VCM_CALLBACKS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VCM_CALLBACKS_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -20,16 +20,15 @@ namespace webrtc {
 class MockVCMFrameTypeCallback : public VCMFrameTypeCallback {
  public:
   MOCK_METHOD0(RequestKeyFrame, int32_t());
-  MOCK_METHOD1(SliceLossIndicationRequest,
-               int32_t(const uint64_t pictureId));
+  MOCK_METHOD1(SliceLossIndicationRequest, int32_t(const uint64_t pictureId));
 };
 
 class MockPacketRequestCallback : public VCMPacketRequestCallback {
  public:
-  MOCK_METHOD2(ResendPackets, int32_t(const uint16_t* sequenceNumbers,
-                                      uint16_t length));
+  MOCK_METHOD2(ResendPackets,
+               int32_t(const uint16_t* sequenceNumbers, uint16_t length));
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VCM_CALLBACKS_H_
diff --git a/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h b/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
new file mode 100644
index 0000000000..9cb4a83535
--- /dev/null
+++ b/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
+
+#include <string>
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class MockEncodedImageCallback : public EncodedImageCallback {
+ public:
+  MOCK_METHOD3(Encoded,
+               int32_t(const EncodedImage& encodedImage,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       const RTPFragmentationHeader* fragmentation));
+};
+
+class MockVideoEncoder : public VideoEncoder {
+ public:
+  MOCK_CONST_METHOD2(Version, int32_t(int8_t* version, int32_t length));
+  MOCK_METHOD3(InitEncode,
+               int32_t(const VideoCodec* codecSettings,
+                       int32_t numberOfCores,
+                       size_t maxPayloadSize));
+  MOCK_METHOD3(Encode,
+               int32_t(const VideoFrame& inputImage,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       const std::vector<FrameType>* frame_types));
+  MOCK_METHOD1(RegisterEncodeCompleteCallback,
+               int32_t(EncodedImageCallback* callback));
+  MOCK_METHOD0(Release, int32_t());
+  MOCK_METHOD0(Reset, int32_t());
+  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
+  MOCK_METHOD2(SetRates, int32_t(uint32_t newBitRate, uint32_t frameRate));
+  MOCK_METHOD1(SetPeriodicKeyFrames, int32_t(bool enable));
+};
+
+class MockDecodedImageCallback : public DecodedImageCallback {
+ public:
+  MOCK_METHOD1(Decoded, int32_t(VideoFrame& decodedImage));  // NOLINT
+  MOCK_METHOD2(Decoded,
+               int32_t(VideoFrame& decodedImage,  // NOLINT
+                       int64_t decode_time_ms));
+  MOCK_METHOD1(ReceivedDecodedReferenceFrame,
+               int32_t(const uint64_t pictureId));
+  MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
+};
+
+class MockVideoDecoder : public VideoDecoder {
+ public:
+  MOCK_METHOD2(InitDecode,
+               int32_t(const VideoCodec* codecSettings, int32_t numberOfCores));
+  MOCK_METHOD5(Decode,
+               int32_t(const EncodedImage& inputImage,
+                       bool missingFrames,
+                       const RTPFragmentationHeader* fragmentation,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       int64_t renderTimeMs));
+  MOCK_METHOD1(RegisterDecodeCompleteCallback,
+               int32_t(DecodedImageCallback* callback));
+  MOCK_METHOD0(Release, int32_t());
+  MOCK_METHOD0(Reset, int32_t());
+  MOCK_METHOD0(Copy, VideoDecoder*());
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
diff --git a/webrtc/modules/video_coding/include/video_codec_interface.h b/webrtc/modules/video_coding/include/video_codec_interface.h
new file mode 100644
index 0000000000..19303c0d67
--- /dev/null
+++ b/webrtc/modules/video_coding/include/video_codec_interface.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
+
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_decoder.h"
+#include "webrtc/video_encoder.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+
+class RTPFragmentationHeader;  // forward declaration
+
+// Note: if any pointers are added to this struct, it must be fitted
+// with a copy-constructor. See below.
+struct CodecSpecificInfoVP8 {
+  bool hasReceivedSLI;
+  uint8_t pictureIdSLI;
+  bool hasReceivedRPSI;
+  uint64_t pictureIdRPSI;
+  int16_t pictureId;  // Negative value to skip pictureId.
+  bool nonReference;
+  uint8_t simulcastIdx;
+  uint8_t temporalIdx;
+  bool layerSync;
+  int tl0PicIdx;  // Negative value to skip tl0PicIdx.
+  int8_t keyIdx;  // Negative value to skip keyIdx.
+};
+
+struct CodecSpecificInfoVP9 {
+  bool has_received_sli;
+  uint8_t picture_id_sli;
+  bool has_received_rpsi;
+  uint64_t picture_id_rpsi;
+  int16_t picture_id;  // Negative value to skip pictureId.
+
+  bool inter_pic_predicted;  // This layer frame is dependent on previously
+                             // coded frame(s).
+  bool flexible_mode;
+  bool ss_data_available;
+
+  int tl0_pic_idx;  // Negative value to skip tl0PicIdx.
+  uint8_t temporal_idx;
+  uint8_t spatial_idx;
+  bool temporal_up_switch;
+  bool inter_layer_predicted;  // Frame is dependent on directly lower spatial
+                               // layer frame.
+  uint8_t gof_idx;
+
+  // SS data.
+  size_t num_spatial_layers;  // Always populated.
+  bool spatial_layer_resolution_present;
+  uint16_t width[kMaxVp9NumberOfSpatialLayers];
+  uint16_t height[kMaxVp9NumberOfSpatialLayers];
+  GofInfoVP9 gof;
+
+  // Frame reference data.
+  uint8_t num_ref_pics;
+  uint8_t p_diff[kMaxVp9RefPics];
+};
+
+struct CodecSpecificInfoGeneric {
+  uint8_t simulcast_idx;
+};
+
+struct CodecSpecificInfoH264 {};
+
+union CodecSpecificInfoUnion {
+  CodecSpecificInfoGeneric generic;
+  CodecSpecificInfoVP8 VP8;
+  CodecSpecificInfoVP9 VP9;
+  CodecSpecificInfoH264 H264;
+};
+
+// Note: if any pointers are added to this struct or its sub-structs, it
+// must be fitted with a copy-constructor. This is because it is copied
+// in the copy-constructor of VCMEncodedFrame.
+struct CodecSpecificInfo {
+  VideoCodecType codecType;
+  CodecSpecificInfoUnion codecSpecific;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
diff --git a/webrtc/modules/video_coding/include/video_coding.h b/webrtc/modules/video_coding/include/video_coding.h
new file mode 100644
index 0000000000..c46896c823
--- /dev/null
+++ b/webrtc/modules/video_coding/include/video_coding.h
@@ -0,0 +1,519 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
+
+#if defined(WEBRTC_WIN)
+// This is a workaround on Windows due to the fact that some Windows
+// headers define CreateEvent as a macro to either CreateEventW or CreateEventA.
+// This can cause problems since we use that name as well and could
+// declare them as one thing here whereas in another place a windows header
+// may have been included and then implementing CreateEvent() causes compilation
+// errors.  So for consistency, we include the main windows header here.
+#include <windows.h>
+#endif
+
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+
+class Clock;
+class EncodedImageCallback;
+class VideoEncoder;
+class VideoDecoder;
+struct CodecSpecificInfo;
+
+class EventFactory {
+ public:
+  virtual ~EventFactory() {}
+
+  virtual EventWrapper* CreateEvent() = 0;
+};
+
+class EventFactoryImpl : public EventFactory {
+ public:
+  virtual ~EventFactoryImpl() {}
+
+  virtual EventWrapper* CreateEvent() { return EventWrapper::Create(); }
+};
+
+// Used to indicate which decode with errors mode should be used.
+enum VCMDecodeErrorMode {
+  kNoErrors,         // Never decode with errors. Video will freeze
+                     // if nack is disabled.
+  kSelectiveErrors,  // Frames that are determined decodable in
+                     // VCMSessionInfo may be decoded with missing
+                     // packets. As not all incomplete frames will be
+                     // decodable, video will freeze if nack is disabled.
+  kWithErrors        // Release frames as needed. Errors may be
+                     // introduced as some encoded frames may not be
+                     // complete.
+};
+
+class VideoCodingModule : public Module {
+ public:
+  enum SenderNackMode { kNackNone, kNackAll, kNackSelective };
+
+  enum ReceiverRobustness { kNone, kHardNack, kSoftNack, kReferenceSelection };
+
+  static VideoCodingModule* Create(
+      Clock* clock,
+      VideoEncoderRateObserver* encoder_rate_observer,
+      VCMQMSettingsCallback* qm_settings_callback);
+
+  static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory);
+
+  static void Destroy(VideoCodingModule* module);
+
+  // Get supported codec settings using codec type
+  //
+  // Input:
+  //      - codecType      : The codec type to get settings for
+  //      - codec          : Memory where the codec settings will be stored
+  //
+  // Return value     : VCM_OK,              on success
+  //                    VCM_PARAMETER_ERROR  if codec not supported
+  static void Codec(VideoCodecType codecType, VideoCodec* codec);
+
+  /*
+  *   Sender
+  */
+
+  // Registers a codec to be used for encoding. Calling this
+  // API multiple times overwrites any previously registered codecs.
+  //
+  // NOTE: Must be called on the thread that constructed the VCM instance.
+  //
+  // Input:
+  //      - sendCodec      : Settings for the codec to be registered.
+  //      - numberOfCores  : The number of cores the codec is allowed
+  //                         to use.
+  //      - maxPayloadSize : The maximum size each payload is allowed
+  //                                to have. Usually MTU - overhead.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterSendCodec(const VideoCodec* sendCodec,
+                                    uint32_t numberOfCores,
+                                    uint32_t maxPayloadSize) = 0;
+
+  // Register an external encoder object. This can not be used together with
+  // external decoder callbacks.
+  //
+  // Input:
+  //      - externalEncoder : Encoder object to be used for encoding frames
+  //      inserted
+  //                          with the AddVideoFrame API.
+  //      - payloadType     : The payload type bound which this encoder is bound
+  //      to.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  // TODO(pbos): Remove return type when unused elsewhere.
+  virtual int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                          uint8_t payloadType,
+                                          bool internalSource = false) = 0;
+
+  // API to get currently configured encoder target bitrate in bits/s.
+  //
+  // Return value      : 0,   on success.
+  //                     < 0, on error.
+  virtual int Bitrate(unsigned int* bitrate) const = 0;
+
+  // API to get currently configured encoder target frame rate.
+  //
+  // Return value      : 0,   on success.
+  //                     < 0, on error.
+  virtual int FrameRate(unsigned int* framerate) const = 0;
+
+  // Sets the parameters describing the send channel. These parameters are
+  // inputs to the
+  // Media Optimization inside the VCM and also specifies the target bit rate
+  // for the
+  // encoder. Bit rate used by NACK should already be compensated for by the
+  // user.
+  //
+  // Input:
+  //      - target_bitrate        : The target bitrate for VCM in bits/s.
+  //      - lossRate              : Fractions of lost packets the past second.
+  //                                (loss rate in percent = 100 * packetLoss /
+  //                                255)
+  //      - rtt                   : Current round-trip time in ms.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,         on error.
+  virtual int32_t SetChannelParameters(uint32_t target_bitrate,
+                                       uint8_t lossRate,
+                                       int64_t rtt) = 0;
+
+  // Sets the parameters describing the receive channel. These parameters are
+  // inputs to the
+  // Media Optimization inside the VCM.
+  //
+  // Input:
+  //      - rtt                   : Current round-trip time in ms.
+  //                                with the most amount available bandwidth in
+  //                                a conference
+  //                                scenario
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetReceiveChannelParameters(int64_t rtt) = 0;
+
+  // Register a transport callback which will be called to deliver the encoded
+  // data and
+  // side information.
+  //
+  // Input:
+  //      - transport  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterTransportCallback(
+      VCMPacketizationCallback* transport) = 0;
+
+  // Register video output information callback which will be called to deliver
+  // information
+  // about the video stream produced by the encoder, for instance the average
+  // frame rate and
+  // bit rate.
+  //
+  // Input:
+  //      - outputInformation  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterSendStatisticsCallback(
+      VCMSendStatisticsCallback* sendStats) = 0;
+
+  // Register a video protection callback which will be called to deliver
+  // the requested FEC rate and NACK status (on/off).
+  //
+  // Input:
+  //      - protection  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterProtectionCallback(
+      VCMProtectionCallback* protection) = 0;
+
+  // Enable or disable a video protection method.
+  //
+  // Input:
+  //      - videoProtection  : The method to enable or disable.
+  //      - enable           : True if the method should be enabled, false if
+  //                           it should be disabled.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetVideoProtection(VCMVideoProtection videoProtection,
+                                     bool enable) = 0;
+
+  // Add one raw video frame to the encoder. This function does all the
+  // necessary
+  // processing, then decides what frame type to encode, or if the frame should
+  // be
+  // dropped. If the frame should be encoded it passes the frame to the encoder
+  // before it returns.
+  //
+  // Input:
+  //      - videoFrame        : Video frame to encode.
+  //      - codecSpecificInfo : Extra codec information, e.g., pre-parsed
+  //      in-band signaling.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t AddVideoFrame(
+      const VideoFrame& videoFrame,
+      const VideoContentMetrics* contentMetrics = NULL,
+      const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
+
+  // Next frame encoded should be an intra frame (keyframe).
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t IntraFrameRequest(int stream_index) = 0;
+
+  // Frame Dropper enable. Can be used to disable the frame dropping when the
+  // encoder
+  // over-uses its bit rate. This API is designed to be used when the encoded
+  // frames
+  // are supposed to be stored to an AVI file, or when the I420 codec is used
+  // and the
+  // target bit rate shouldn't affect the frame rate.
+  //
+  // Input:
+  //      - enable            : True to enable the setting, false to disable it.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t EnableFrameDropper(bool enable) = 0;
+
+  /*
+  *   Receiver
+  */
+
+  // Register possible receive codecs, can be called multiple times for
+  // different codecs.
+  // The module will automatically switch between registered codecs depending on
+  // the
+  // payload type of incoming frames. The actual decoder will be created when
+  // needed.
+  //
+  // Input:
+  //      - receiveCodec      : Settings for the codec to be registered.
+  //      - numberOfCores     : Number of CPU cores that the decoder is allowed
+  //      to use.
+  //      - requireKeyFrame   : Set this to true if you don't want any delta
+  //      frames
+  //                            to be decoded until the first key frame has been
+  //                            decoded.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
+                                       int32_t numberOfCores,
+                                       bool requireKeyFrame = false) = 0;
+
+  // Register an externally defined decoder/renderer object. Can be a decoder
+  // only or a
+  // decoder coupled with a renderer. Note that RegisterReceiveCodec must be
+  // called to
+  // be used for decoding incoming streams.
+  //
+  // Input:
+  //      - externalDecoder        : The external decoder/renderer object.
+  //      - payloadType            : The payload type which this decoder should
+  //      be
+  //                                 registered to.
+  //
+  virtual void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                       uint8_t payloadType) = 0;
+
+  // Register a receive callback. Will be called whenever there is a new frame
+  // ready
+  // for rendering.
+  //
+  // Input:
+  //      - receiveCallback        : The callback object to be used by the
+  //      module when a
+  //                                 frame is ready for rendering.
+  //                                 De-register with a NULL pointer.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterReceiveCallback(
+      VCMReceiveCallback* receiveCallback) = 0;
+
+  // Register a receive statistics callback which will be called to deliver
+  // information
+  // about the video stream received by the receiving side of the VCM, for
+  // instance the
+  // average frame rate and bit rate.
+  //
+  // Input:
+  //      - receiveStats  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterReceiveStatisticsCallback(
+      VCMReceiveStatisticsCallback* receiveStats) = 0;
+
+  // Register a decoder timing callback which will be called to deliver
+  // information about the timing of the decoder in the receiving side of the
+  // VCM, for instance the current and maximum frame decode latency.
+  //
+  // Input:
+  //      - decoderTiming  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterDecoderTimingCallback(
+      VCMDecoderTimingCallback* decoderTiming) = 0;
+
+  // Register a frame type request callback. This callback will be called when
+  // the
+  // module needs to request specific frame types from the send side.
+  //
+  // Input:
+  //      - frameTypeCallback      : The callback object to be used by the
+  //      module when
+  //                                 requesting a specific type of frame from
+  //                                 the send side.
+  //                                 De-register with a NULL pointer.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterFrameTypeCallback(
+      VCMFrameTypeCallback* frameTypeCallback) = 0;
+
+  // Registers a callback which is called whenever the receive side of the VCM
+  // encounters holes in the packet sequence and needs packets to be
+  // retransmitted.
+  //
+  // Input:
+  //              - callback      : The callback to be registered in the VCM.
+  //
+  // Return value     : VCM_OK,     on success.
+  //                    <0,         on error.
+  virtual int32_t RegisterPacketRequestCallback(
+      VCMPacketRequestCallback* callback) = 0;
+
+  // Waits for the next frame in the jitter buffer to become complete
+  // (waits no longer than maxWaitTimeMs), then passes it to the decoder for
+  // decoding.
+  // Should be called as often as possible to get the most out of the decoder.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t Decode(uint16_t maxWaitTimeMs = 200) = 0;
+
+  // Registers a callback which conveys the size of the render buffer.
+  virtual int RegisterRenderBufferSizeCallback(
+      VCMRenderBufferSizeCallback* callback) = 0;
+
+  // Reset the decoder state to the initial state.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t ResetDecoder() = 0;
+
+  // API to get the codec which is currently used for decoding by the module.
+  //
+  // Input:
+  //      - currentReceiveCodec      : Settings for the codec to be registered.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
+
+  // API to get the codec type currently used for decoding by the module.
+  //
+  // Return value      : codecy type,            on success.
+  //                     kVideoCodecUnknown, on error or if no receive codec is
+  //                     registered
+  virtual VideoCodecType ReceiveCodec() const = 0;
+
+  // Insert a parsed packet into the receiver side of the module. Will be placed
+  // in the
+  // jitter buffer waiting for the frame to become complete. Returns as soon as
+  // the packet
+  // has been placed in the jitter buffer.
+  //
+  // Input:
+  //      - incomingPayload      : Payload of the packet.
+  //      - payloadLength        : Length of the payload.
+  //      - rtpInfo              : The parsed header.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t IncomingPacket(const uint8_t* incomingPayload,
+                                 size_t payloadLength,
+                                 const WebRtcRTPHeader& rtpInfo) = 0;
+
+  // Minimum playout delay (Used for lip-sync). This is the minimum delay
+  // required
+  // to sync with audio. Not included in  VideoCodingModule::Delay()
+  // Defaults to 0 ms.
+  //
+  // Input:
+  //      - minPlayoutDelayMs   : Additional delay in ms.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) = 0;
+
+  // Set the time required by the renderer to render a frame.
+  //
+  // Input:
+  //      - timeMS        : The time in ms required by the renderer to render a
+  //      frame.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetRenderDelay(uint32_t timeMS) = 0;
+
+  // The total delay desired by the VCM. Can be less than the minimum
+  // delay set with SetMinimumPlayoutDelay.
+  //
+  // Return value      : Total delay in ms, on success.
+  //                     < 0,               on error.
+  virtual int32_t Delay() const = 0;
+
+  // Returns the number of packets discarded by the jitter buffer due to being
+  // too late. This can include duplicated packets which arrived after the
+  // frame was sent to the decoder. Therefore packets which were prematurely
+  // NACKed will be counted.
+  virtual uint32_t DiscardedPackets() const = 0;
+
+  // Robustness APIs
+
+  // Set the receiver robustness mode. The mode decides how the receiver
+  // responds to losses in the stream. The type of counter-measure (soft or
+  // hard NACK, dual decoder, RPS, etc.) is selected through the
+  // robustnessMode parameter. The errorMode parameter decides if it is
+  // allowed to display frames corrupted by losses. Note that not all
+  // combinations of the two parameters are feasible. An error will be
+  // returned for invalid combinations.
+  // Input:
+  //      - robustnessMode : selected robustness mode.
+  //      - errorMode      : selected error mode.
+  //
+  // Return value      : VCM_OK, on success;
+  //                     < 0, on error.
+  virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
+                                        VCMDecodeErrorMode errorMode) = 0;
+
+  // Set the decode error mode. The mode decides which errors (if any) are
+  // allowed in decodable frames. Note that setting decode_error_mode to
+  // anything other than kWithErrors without enabling nack will cause
+  // long-term freezes (resulting from frequent key frame requests) if
+  // packet loss occurs.
+  virtual void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) = 0;
+
+  // Sets the maximum number of sequence numbers that we are allowed to NACK
+  // and the oldest sequence number that we will consider to NACK. If a
+  // sequence number older than |max_packet_age_to_nack| is missing
+  // a key frame will be requested. A key frame will also be requested if the
+  // time of incomplete or non-continuous frames in the jitter buffer is above
+  // |max_incomplete_time_ms|.
+  virtual void SetNackSettings(size_t max_nack_list_size,
+                               int max_packet_age_to_nack,
+                               int max_incomplete_time_ms) = 0;
+
+  // Setting a desired delay to the VCM receiver. Video rendering will be
+  // delayed by at least desired_delay_ms.
+  virtual int SetMinReceiverDelay(int desired_delay_ms) = 0;
+
+  // Lets the sender suspend video when the rate drops below
+  // |threshold_bps|, and turns back on when the rate goes back up above
+  // |threshold_bps| + |window_bps|.
+  virtual void SuspendBelowMinBitrate() = 0;
+
+  // Returns true if SuspendBelowMinBitrate is engaged and the video has been
+  // suspended due to bandwidth limitations; otherwise false.
+  virtual bool VideoSuspended() const = 0;
+
+  virtual void RegisterPreDecodeImageCallback(
+      EncodedImageCallback* observer) = 0;
+  virtual void RegisterPostEncodeImageCallback(
+      EncodedImageCallback* post_encode_callback) = 0;
+  // Releases pending decode calls, permitting faster thread shutdown.
+  virtual void TriggerDecoderShutdown() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
diff --git a/webrtc/modules/video_coding/main/interface/video_coding_defines.h b/webrtc/modules/video_coding/include/video_coding_defines.h
index fd38d64415..673a02b713 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding_defines.h
+++ b/webrtc/modules/video_coding/include/video_coding_defines.h
@@ -8,33 +8,33 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
-#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
 namespace webrtc {
 
 // Error codes
-#define VCM_FRAME_NOT_READY      3
-#define VCM_REQUEST_SLI          2
-#define VCM_MISSING_CALLBACK     1
-#define VCM_OK                   0
-#define VCM_GENERAL_ERROR       -1
-#define VCM_LEVEL_EXCEEDED      -2
-#define VCM_MEMORY              -3
-#define VCM_PARAMETER_ERROR     -4
-#define VCM_UNKNOWN_PAYLOAD     -5
-#define VCM_CODEC_ERROR         -6
-#define VCM_UNINITIALIZED       -7
+#define VCM_FRAME_NOT_READY 3
+#define VCM_REQUEST_SLI 2
+#define VCM_MISSING_CALLBACK 1
+#define VCM_OK 0
+#define VCM_GENERAL_ERROR -1
+#define VCM_LEVEL_EXCEEDED -2
+#define VCM_MEMORY -3
+#define VCM_PARAMETER_ERROR -4
+#define VCM_UNKNOWN_PAYLOAD -5
+#define VCM_CODEC_ERROR -6
+#define VCM_UNINITIALIZED -7
 #define VCM_NO_CODEC_REGISTERED -8
 #define VCM_JITTER_BUFFER_ERROR -9
-#define VCM_OLD_PACKET_ERROR    -10
-#define VCM_NO_FRAME_DECODED    -11
-#define VCM_ERROR_REQUEST_SLI   -12
-#define VCM_NOT_IMPLEMENTED     -20
+#define VCM_OLD_PACKET_ERROR -10
+#define VCM_NO_FRAME_DECODED -11
+#define VCM_ERROR_REQUEST_SLI -12
+#define VCM_NOT_IMPLEMENTED -20
 
 enum { kDefaultStartBitrateKbps = 300 };
 
@@ -62,40 +62,42 @@ class VCMPacketizationCallback {
                            const RTPFragmentationHeader& fragmentationHeader,
                            const RTPVideoHeader* rtpVideoHdr) = 0;
 
+  virtual void OnEncoderImplementationName(const char* implementation_name) {}
+
  protected:
-  virtual ~VCMPacketizationCallback() {
-  }
+  virtual ~VCMPacketizationCallback() {}
 };
 
-// Callback class used for passing decoded frames which are ready to be rendered.
+// Callback class used for passing decoded frames which are ready to be
+// rendered.
 class VCMReceiveCallback {
  public:
-  virtual int32_t FrameToRender(VideoFrame& videoFrame) = 0;
-  virtual int32_t ReceivedDecodedReferenceFrame(
-      const uint64_t pictureId) {
+  virtual int32_t FrameToRender(VideoFrame& videoFrame) = 0;  // NOLINT
+  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) {
     return -1;
   }
   // Called when the current receive codec changes.
   virtual void OnIncomingPayloadType(int payload_type) {}
+  virtual void OnDecoderImplementationName(const char* implementation_name) {}
 
  protected:
-  virtual ~VCMReceiveCallback() {
-  }
+  virtual ~VCMReceiveCallback() {}
 };
 
-// Callback class used for informing the user of the bit rate and frame rate produced by the
+// Callback class used for informing the user of the bit rate and frame rate
+// produced by the
 // encoder.
 class VCMSendStatisticsCallback {
  public:
   virtual int32_t SendStatistics(const uint32_t bitRate,
-                                       const uint32_t frameRate) = 0;
+                                 const uint32_t frameRate) = 0;
 
  protected:
-  virtual ~VCMSendStatisticsCallback() {
-  }
+  virtual ~VCMSendStatisticsCallback() {}
 };
 
-// Callback class used for informing the user of the incoming bit rate and frame rate.
+// Callback class used for informing the user of the incoming bit rate and frame
+// rate.
 class VCMReceiveStatisticsCallback {
  public:
   virtual void OnReceiveRatesUpdated(uint32_t bitRate, uint32_t frameRate) = 0;
@@ -103,8 +105,7 @@ class VCMReceiveStatisticsCallback {
   virtual void OnFrameCountsUpdated(const FrameCounts& frame_counts) = 0;
 
  protected:
-  virtual ~VCMReceiveStatisticsCallback() {
-  }
+  virtual ~VCMReceiveStatisticsCallback() {}
 };
 
 // Callback class used for informing the user of decode timing info.
@@ -133,8 +134,7 @@ class VCMProtectionCallback {
                                 uint32_t* sent_fec_rate_bps) = 0;
 
  protected:
-  virtual ~VCMProtectionCallback() {
-  }
+  virtual ~VCMProtectionCallback() {}
 };
 
 class VideoEncoderRateObserver {
@@ -143,31 +143,30 @@ class VideoEncoderRateObserver {
   virtual void OnSetRates(uint32_t bitrate_bps, int framerate) = 0;
 };
 
-// Callback class used for telling the user about what frame type needed to continue decoding.
+// Callback class used for telling the user about what frame type needed to
+// continue decoding.
 // Typically a key frame when the stream has been corrupted in some way.
 class VCMFrameTypeCallback {
  public:
   virtual int32_t RequestKeyFrame() = 0;
-  virtual int32_t SliceLossIndicationRequest(
-      const uint64_t pictureId) {
+  virtual int32_t SliceLossIndicationRequest(const uint64_t pictureId) {
     return -1;
   }
 
  protected:
-  virtual ~VCMFrameTypeCallback() {
-  }
+  virtual ~VCMFrameTypeCallback() {}
 };
 
-// Callback class used for telling the user about which packet sequence numbers are currently
+// Callback class used for telling the user about which packet sequence numbers
+// are currently
 // missing and need to be resent.
 class VCMPacketRequestCallback {
  public:
   virtual int32_t ResendPackets(const uint16_t* sequenceNumbers,
-                                      uint16_t length) = 0;
+                                uint16_t length) = 0;
 
  protected:
-  virtual ~VCMPacketRequestCallback() {
-  }
+  virtual ~VCMPacketRequestCallback() {}
 };
 
 // Callback used to inform the user of the the desired resolution
@@ -175,14 +174,13 @@ class VCMPacketRequestCallback {
 class VCMQMSettingsCallback {
  public:
   virtual int32_t SetVideoQMSettings(const uint32_t frameRate,
-                                           const uint32_t width,
-                                           const uint32_t height) = 0;
+                                     const uint32_t width,
+                                     const uint32_t height) = 0;
 
   virtual void SetTargetFramerate(int frame_rate) = 0;
 
  protected:
-  virtual ~VCMQMSettingsCallback() {
-  }
+  virtual ~VCMQMSettingsCallback() {}
 };
 
 // Callback class used for telling the user about the size (in time) of the
@@ -192,10 +190,9 @@ class VCMRenderBufferSizeCallback {
   virtual void RenderBufferSizeMs(int buffer_size_ms) = 0;
 
  protected:
-  virtual ~VCMRenderBufferSizeCallback() {
-  }
+  virtual ~VCMRenderBufferSizeCallback() {}
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
diff --git a/webrtc/modules/video_coding/include/video_error_codes.h b/webrtc/modules/video_coding/include/video_error_codes.h
new file mode 100644
index 0000000000..360aa87744
--- /dev/null
+++ b/webrtc/modules/video_coding/include/video_error_codes.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
+
+// NOTE: in sync with video_coding_module_defines.h
+
+// Define return values
+
+#define WEBRTC_VIDEO_CODEC_REQUEST_SLI 2
+#define WEBRTC_VIDEO_CODEC_NO_OUTPUT 1
+#define WEBRTC_VIDEO_CODEC_OK 0
+#define WEBRTC_VIDEO_CODEC_ERROR -1
+#define WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED -2
+#define WEBRTC_VIDEO_CODEC_MEMORY -3
+#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
+#define WEBRTC_VIDEO_CODEC_ERR_SIZE -5
+#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
+#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
+#define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
+#define WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE -13
+#define WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT -14
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
diff --git a/webrtc/modules/video_coding/inter_frame_delay.cc b/webrtc/modules/video_coding/inter_frame_delay.cc
new file mode 100644
index 0000000000..fb3b54d204
--- /dev/null
+++ b/webrtc/modules/video_coding/inter_frame_delay.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/inter_frame_delay.h"
+
+namespace webrtc {
+
+VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock) {
+  Reset(currentWallClock);
+}
+
+// Resets the delay estimate
+void VCMInterFrameDelay::Reset(int64_t currentWallClock) {
+  _zeroWallClock = currentWallClock;
+  _wrapArounds = 0;
+  _prevWallClock = 0;
+  _prevTimestamp = 0;
+  _dTS = 0;
+}
+
+// Calculates the delay of a frame with the given timestamp.
+// This method is called when the frame is complete.
+bool VCMInterFrameDelay::CalculateDelay(uint32_t timestamp,
+                                        int64_t* delay,
+                                        int64_t currentWallClock) {
+  if (_prevWallClock == 0) {
+    // First set of data, initialization, wait for next frame
+    _prevWallClock = currentWallClock;
+    _prevTimestamp = timestamp;
+    *delay = 0;
+    return true;
+  }
+
+  int32_t prevWrapArounds = _wrapArounds;
+  CheckForWrapArounds(timestamp);
+
+  // This will be -1 for backward wrap arounds and +1 for forward wrap arounds
+  int32_t wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
+
+  // Account for reordering in jitter variance estimate in the future?
+  // Note that this also captures incomplete frames which are grabbed
+  // for decoding after a later frame has been complete, i.e. real
+  // packet losses.
+  if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) ||
+      wrapAroundsSincePrev < 0) {
+    *delay = 0;
+    return false;
+  }
+
+  // Compute the compensated timestamp difference and convert it to ms and
+  // round it to closest integer.
+  _dTS = static_cast<int64_t>(
+      (timestamp + wrapAroundsSincePrev * (static_cast<int64_t>(1) << 32) -
+       _prevTimestamp) /
+          90.0 +
+      0.5);
+
+  // frameDelay is the difference of dT and dTS -- i.e. the difference of
+  // the wall clock time difference and the timestamp difference between
+  // two following frames.
+  *delay = static_cast<int64_t>(currentWallClock - _prevWallClock - _dTS);
+
+  _prevTimestamp = timestamp;
+  _prevWallClock = currentWallClock;
+
+  return true;
+}
+
+// Returns the current difference between incoming timestamps
+uint32_t VCMInterFrameDelay::CurrentTimeStampDiffMs() const {
+  if (_dTS < 0) {
+    return 0;
+  }
+  return static_cast<uint32_t>(_dTS);
+}
+
+// Investigates if the timestamp clock has overflowed since the last timestamp
+// and
+// keeps track of the number of wrap arounds since reset.
+void VCMInterFrameDelay::CheckForWrapArounds(uint32_t timestamp) {
+  if (timestamp < _prevTimestamp) {
+    // This difference will probably be less than -2^31 if we have had a wrap
+    // around
+    // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to
+    // a Word32,
+    // it should be positive.
+    if (static_cast<int32_t>(timestamp - _prevTimestamp) > 0) {
+      // Forward wrap around
+      _wrapArounds++;
+    }
+    // This difference will probably be less than -2^31 if we have had a
+    // backward
+    // wrap around.
+    // Since it is cast to a Word32, it should be positive.
+  } else if (static_cast<int32_t>(_prevTimestamp - timestamp) > 0) {
+    // Backward wrap around
+    _wrapArounds--;
+  }
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/inter_frame_delay.h b/webrtc/modules/video_coding/inter_frame_delay.h
new file mode 100644
index 0000000000..94b73908bb
--- /dev/null
+++ b/webrtc/modules/video_coding/inter_frame_delay.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class VCMInterFrameDelay {
+ public:
+  explicit VCMInterFrameDelay(int64_t currentWallClock);
+
+  // Resets the estimate. Zeros are given as parameters.
+  void Reset(int64_t currentWallClock);
+
+  // Calculates the delay of a frame with the given timestamp.
+  // This method is called when the frame is complete.
+  //
+  // Input:
+  //          - timestamp         : RTP timestamp of a received frame
+  //          - *delay            : Pointer to memory where the result should be
+  //          stored
+  //          - currentWallClock  : The current time in milliseconds.
+  //                                Should be -1 for normal operation, only used
+  //                                for testing.
+  // Return value                 : true if OK, false when reordered timestamps
+  bool CalculateDelay(uint32_t timestamp,
+                      int64_t* delay,
+                      int64_t currentWallClock);
+
+  // Returns the current difference between incoming timestamps
+  //
+  // Return value                 : Wrap-around compensated difference between
+  // incoming
+  //                                timestamps.
+  uint32_t CurrentTimeStampDiffMs() const;
+
+ private:
+  // Controls if the RTP timestamp counter has had a wrap around
+  // between the current and the previously received frame.
+  //
+  // Input:
+  //          - timestmap         : RTP timestamp of the current frame.
+  void CheckForWrapArounds(uint32_t timestamp);
+
+  int64_t _zeroWallClock;  // Local timestamp of the first video packet received
+  int32_t _wrapArounds;    // Number of wrapArounds detected
+  // The previous timestamp passed to the delay estimate
+  uint32_t _prevTimestamp;
+  // The previous wall clock timestamp used by the delay estimate
+  int64_t _prevWallClock;
+  // Wrap-around compensated difference between incoming timestamps
+  int64_t _dTS;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
diff --git a/webrtc/modules/video_coding/internal_defines.h b/webrtc/modules/video_coding/internal_defines.h
new file mode 100644
index 0000000000..e225726dea
--- /dev/null
+++ b/webrtc/modules/video_coding/internal_defines.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
+
+inline uint32_t MaskWord64ToUWord32(int64_t w64) {
+  return static_cast<uint32_t>(MASK_32_BITS(w64));
+}
+
+#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
+#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
+
+#define VCM_DEFAULT_CODEC_WIDTH 352
+#define VCM_DEFAULT_CODEC_HEIGHT 288
+#define VCM_DEFAULT_FRAME_RATE 30
+#define VCM_MIN_BITRATE 30
+#define VCM_FLUSH_INDICATOR 4
+
+#define VCM_NO_RECEIVER_ID 0
+
+inline int32_t VCMId(const int32_t vcmId, const int32_t receiverId = 0) {
+  return static_cast<int32_t>((vcmId << 16) + receiverId);
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/jitter_buffer.cc
index bfdd7867d9..640bcb4f22 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/jitter_buffer.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
 
 #include <assert.h>
 
@@ -15,19 +15,19 @@
 #include <utility>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/inter_frame_delay.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
+#include "webrtc/modules/video_coding/packet.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
@@ -38,6 +38,10 @@ static const uint32_t kSsCleanupIntervalSec = 60;
 // Use this rtt if no value has been reported.
 static const int64_t kDefaultRtt = 200;
 
+// Request a keyframe if no continuous frame has been received for this
+// number of milliseconds and NACKs are disabled.
+static const int64_t kMaxDiscontinuousFramesTime = 1000;
+
 typedef std::pair<uint32_t, VCMFrameBuffer*> FrameListPair;
 
 bool IsKeyFrame(FrameListPair pair) {
@@ -89,7 +93,7 @@ int FrameList::RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
 }
 
 void FrameList::CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
-                                       UnorderedFrameList* free_frames) {
+                                        UnorderedFrameList* free_frames) {
   while (!empty()) {
     VCMFrameBuffer* oldest_frame = Front();
     bool remove_frame = false;
@@ -168,6 +172,7 @@ void Vp9SsMap::AdvanceFront(uint32_t timestamp) {
   ss_map_[timestamp] = gof;
 }
 
+// TODO(asapersson): Update according to updates in RTP payload profile.
 bool Vp9SsMap::UpdatePacket(VCMPacket* packet) {
   uint8_t gof_idx = packet->codecSpecificHeader.codecHeader.VP9.gof_idx;
   if (gof_idx == kNoGofIdx)
@@ -186,7 +191,7 @@ bool Vp9SsMap::UpdatePacket(VCMPacket* packet) {
 
   // TODO(asapersson): Set vp9.ref_picture_id[i] and add usage.
   vp9->num_ref_pics = it->second.num_ref_pics[gof_idx];
-  for (size_t i = 0; i < it->second.num_ref_pics[gof_idx]; ++i) {
+  for (uint8_t i = 0; i < it->second.num_ref_pics[gof_idx]; ++i) {
     vp9->pid_diff[i] = it->second.pid_diff[gof_idx][i];
   }
   return true;
@@ -214,7 +219,7 @@ VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
     : clock_(clock),
       running_(false),
       crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      frame_event_(event.Pass()),
+      frame_event_(std::move(event)),
       max_number_of_frames_(kStartNumberOfFrames),
       free_frames_(),
       decodable_frames_(),
@@ -276,17 +281,18 @@ void VCMJitterBuffer::UpdateHistograms() {
     return;
   }
 
-  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
-      num_discarded_packets_ * 100 / num_packets_);
-  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
-      num_duplicated_packets_ * 100 / num_packets_);
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DiscardedPacketsInPercent",
+                                  num_discarded_packets_ * 100 / num_packets_);
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DuplicatedPacketsInPercent",
+                                  num_duplicated_packets_ * 100 / num_packets_);
 
   int total_frames =
       receive_statistics_.key_frames + receive_statistics_.delta_frames;
   if (total_frames > 0) {
-    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.CompleteFramesReceivedPerSecond",
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(
+        "WebRTC.Video.CompleteFramesReceivedPerSecond",
         static_cast<int>((total_frames / elapsed_sec) + 0.5f));
-    RTC_HISTOGRAM_COUNTS_1000(
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(
         "WebRTC.Video.KeyFramesReceivedInPermille",
         static_cast<int>(
             (receive_statistics_.key_frames * 1000.0f / total_frames) + 0.5f));
@@ -316,7 +322,6 @@ void VCMJitterBuffer::Start() {
   first_packet_since_reset_ = true;
   rtt_ms_ = kDefaultRtt;
   last_decoded_state_.Reset();
-  vp9_ss_map_.Reset();
 }
 
 void VCMJitterBuffer::Stop() {
@@ -324,7 +329,6 @@ void VCMJitterBuffer::Stop() {
   UpdateHistograms();
   running_ = false;
   last_decoded_state_.Reset();
-  vp9_ss_map_.Reset();
 
   // Make sure all frames are free and reset.
   for (FrameList::iterator it = decodable_frames_.begin();
@@ -356,7 +360,6 @@ void VCMJitterBuffer::Flush() {
   decodable_frames_.Reset(&free_frames_);
   incomplete_frames_.Reset(&free_frames_);
   last_decoded_state_.Reset();  // TODO(mikhal): sync reset.
-  vp9_ss_map_.Reset();
   num_consecutive_old_packets_ = 0;
   // Also reset the jitter and delay estimates
   jitter_estimate_.Reset();
@@ -428,8 +431,8 @@ void VCMJitterBuffer::IncomingRateStatistics(unsigned int* framerate,
     if (incoming_bit_count_ == 0) {
       *bitrate = 0;
     } else {
-      *bitrate = 10 * ((100 * incoming_bit_count_) /
-                       static_cast<unsigned int>(diff));
+      *bitrate =
+          10 * ((100 * incoming_bit_count_) / static_cast<unsigned int>(diff));
     }
     incoming_bit_rate_ = *bitrate;
 
@@ -470,8 +473,8 @@ bool VCMJitterBuffer::CompleteSequenceWithNextFrame() {
 
 // Returns immediately or a |max_wait_time_ms| ms event hang waiting for a
 // complete frame, |max_wait_time_ms| decided by caller.
-bool VCMJitterBuffer::NextCompleteTimestamp(
-    uint32_t max_wait_time_ms, uint32_t* timestamp) {
+bool VCMJitterBuffer::NextCompleteTimestamp(uint32_t max_wait_time_ms,
+                                            uint32_t* timestamp) {
   crit_sect_->Enter();
   if (!running_) {
     crit_sect_->Leave();
@@ -481,13 +484,13 @@ bool VCMJitterBuffer::NextCompleteTimestamp(
 
   if (decodable_frames_.empty() ||
       decodable_frames_.Front()->GetState() != kStateComplete) {
-    const int64_t end_wait_time_ms = clock_->TimeInMilliseconds() +
-        max_wait_time_ms;
+    const int64_t end_wait_time_ms =
+        clock_->TimeInMilliseconds() + max_wait_time_ms;
     int64_t wait_time_ms = max_wait_time_ms;
     while (wait_time_ms > 0) {
       crit_sect_->Leave();
       const EventTypeWrapper ret =
-        frame_event_->Wait(static_cast<uint32_t>(wait_time_ms));
+          frame_event_->Wait(static_cast<uint32_t>(wait_time_ms));
       crit_sect_->Enter();
       if (ret == kEventSignaled) {
         // Are we shutting down the jitter buffer?
@@ -530,16 +533,25 @@ bool VCMJitterBuffer::NextMaybeIncompleteTimestamp(uint32_t* timestamp) {
 
   CleanUpOldOrEmptyFrames();
 
+  VCMFrameBuffer* oldest_frame;
   if (decodable_frames_.empty()) {
-    return false;
-  }
-  VCMFrameBuffer* oldest_frame = decodable_frames_.Front();
-  // If we have exactly one frame in the buffer, release it only if it is
-  // complete. We know decodable_frames_ is  not empty due to the previous
-  // check.
-  if (decodable_frames_.size() == 1 && incomplete_frames_.empty()
-      && oldest_frame->GetState() != kStateComplete) {
-    return false;
+    if (nack_mode_ != kNoNack || incomplete_frames_.size() <= 1) {
+      return false;
+    }
+    oldest_frame = incomplete_frames_.Front();
+    // Frame will only be removed from buffer if it is complete (or decodable).
+    if (oldest_frame->GetState() < kStateComplete) {
+      return false;
+    }
+  } else {
+    oldest_frame = decodable_frames_.Front();
+    // If we have exactly one frame in the buffer, release it only if it is
+    // complete. We know decodable_frames_ is  not empty due to the previous
+    // check.
+    if (decodable_frames_.size() == 1 && incomplete_frames_.empty() &&
+        oldest_frame->GetState() != kStateComplete) {
+      return false;
+    }
   }
 
   *timestamp = oldest_frame->TimeStamp();
@@ -576,8 +588,7 @@ VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) {
     } else {
       // Wait for this one to get complete.
       waiting_for_completion_.frame_size = frame->Length();
-      waiting_for_completion_.latest_packet_time =
-          frame->LatestPacketTimeMs();
+      waiting_for_completion_.latest_packet_time = frame->LatestPacketTimeMs();
       waiting_for_completion_.timestamp = frame->TimeStamp();
     }
   }
@@ -688,21 +699,6 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
 
   num_consecutive_old_packets_ = 0;
 
-  if (packet.codec == kVideoCodecVP9) {
-    if (packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
-      // TODO(asapersson): Add support for flexible mode.
-      return kGeneralError;
-    }
-    if (!packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
-      if (vp9_ss_map_.Insert(packet))
-        vp9_ss_map_.UpdateFrames(&incomplete_frames_);
-
-      vp9_ss_map_.UpdatePacket(const_cast<VCMPacket*>(&packet));
-    }
-    if (!last_decoded_state_.in_initial_state())
-      vp9_ss_map_.RemoveOld(last_decoded_state_.time_stamp());
-  }
-
   VCMFrameBuffer* frame;
   FrameList* frame_list;
   const VCMFrameBufferEnum error = GetFrame(packet, &frame, &frame_list);
@@ -745,8 +741,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
       frame->InsertPacket(packet, now_ms, decode_error_mode_, frame_data);
 
   if (previous_state != kStateComplete) {
-    TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", frame->TimeStamp(),
-                             "timestamp", frame->TimeStamp());
+    TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", frame->TimeStamp(), "timestamp",
+                             frame->TimeStamp());
   }
 
   if (buffer_state > 0) {
@@ -763,8 +759,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
         buffer_state = kFlushIndicator;
       }
 
-      latest_received_sequence_number_ = LatestSequenceNumber(
-          latest_received_sequence_number_, packet.seqNum);
+      latest_received_sequence_number_ =
+          LatestSequenceNumber(latest_received_sequence_number_, packet.seqNum);
     }
   }
 
@@ -796,6 +792,12 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
         FindAndInsertContinuousFrames(*frame);
       } else {
         incomplete_frames_.InsertFrame(frame);
+        // If NACKs are enabled, keyframes are triggered by |GetNackList|.
+        if (nack_mode_ == kNoNack &&
+            NonContinuousOrIncompleteDuration() >
+                90 * kMaxDiscontinuousFramesTime) {
+          return kFlushIndicator;
+        }
       }
       break;
     }
@@ -806,6 +808,12 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
         return kNoError;
       } else {
         incomplete_frames_.InsertFrame(frame);
+        // If NACKs are enabled, keyframes are triggered by |GetNackList|.
+        if (nack_mode_ == kNoNack &&
+            NonContinuousOrIncompleteDuration() >
+                90 * kMaxDiscontinuousFramesTime) {
+          return kFlushIndicator;
+        }
       }
       break;
     }
@@ -824,15 +832,15 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
     case kFlushIndicator:
       free_frames_.push_back(frame);
       return kFlushIndicator;
-    default: assert(false);
+    default:
+      assert(false);
   }
   return buffer_state;
 }
 
-bool VCMJitterBuffer::IsContinuousInState(const VCMFrameBuffer& frame,
+bool VCMJitterBuffer::IsContinuousInState(
+    const VCMFrameBuffer& frame,
     const VCMDecodingState& decoding_state) const {
-  if (decode_error_mode_ == kWithErrors)
-    return true;
   // Is this frame (complete or decodable) and continuous?
   // kStateDecodable will never be set when decode_error_mode_ is false
   // as SessionInfo determines this state based on the error mode (and frame
@@ -849,7 +857,7 @@ bool VCMJitterBuffer::IsContinuous(const VCMFrameBuffer& frame) const {
   VCMDecodingState decoding_state;
   decoding_state.CopyFrom(last_decoded_state_);
   for (FrameList::const_iterator it = decodable_frames_.begin();
-       it != decodable_frames_.end(); ++it)  {
+       it != decodable_frames_.end(); ++it) {
     VCMFrameBuffer* decodable_frame = it->second;
     if (IsNewerTimestamp(decodable_frame->TimeStamp(), frame.TimeStamp())) {
       break;
@@ -882,7 +890,7 @@ void VCMJitterBuffer::FindAndInsertContinuousFramesWithState(
   // 1. Continuous base or sync layer.
   // 2. The end of the list was reached.
   for (FrameList::iterator it = incomplete_frames_.begin();
-       it != incomplete_frames_.end();)  {
+       it != incomplete_frames_.end();) {
     VCMFrameBuffer* frame = it->second;
     if (IsNewerTimestamp(original_decoded_state.time_stamp(),
                          frame->TimeStamp())) {
@@ -992,16 +1000,18 @@ std::vector<uint16_t> VCMJitterBuffer::GetNackList(bool* request_key_frame) {
   if (last_decoded_state_.in_initial_state()) {
     VCMFrameBuffer* next_frame = NextFrame();
     const bool first_frame_is_key = next_frame &&
-        next_frame->FrameType() == kVideoFrameKey &&
-        next_frame->HaveFirstPacket();
+                                    next_frame->FrameType() == kVideoFrameKey &&
+                                    next_frame->HaveFirstPacket();
     if (!first_frame_is_key) {
-      bool have_non_empty_frame = decodable_frames_.end() != find_if(
-          decodable_frames_.begin(), decodable_frames_.end(),
-          HasNonEmptyState);
+      bool have_non_empty_frame =
+          decodable_frames_.end() != find_if(decodable_frames_.begin(),
+                                             decodable_frames_.end(),
+                                             HasNonEmptyState);
       if (!have_non_empty_frame) {
-        have_non_empty_frame = incomplete_frames_.end() != find_if(
-            incomplete_frames_.begin(), incomplete_frames_.end(),
-            HasNonEmptyState);
+        have_non_empty_frame =
+            incomplete_frames_.end() != find_if(incomplete_frames_.begin(),
+                                                incomplete_frames_.end(),
+                                                HasNonEmptyState);
       }
       bool found_key_frame = RecycleFramesUntilKeyFrame();
       if (!found_key_frame) {
@@ -1020,8 +1030,8 @@ std::vector<uint16_t> VCMJitterBuffer::GetNackList(bool* request_key_frame) {
       LOG_F(LS_WARNING) << "Too long non-decodable duration: "
                         << non_continuous_incomplete_duration << " > "
                         << 90 * max_incomplete_time_ms_;
-      FrameList::reverse_iterator rit = find_if(incomplete_frames_.rbegin(),
-          incomplete_frames_.rend(), IsKeyFrame);
+      FrameList::reverse_iterator rit = find_if(
+          incomplete_frames_.rbegin(), incomplete_frames_.rend(), IsKeyFrame);
       if (rit == incomplete_frames_.rend()) {
         // Request a key frame if we don't have one already.
         *request_key_frame = true;
@@ -1061,8 +1071,7 @@ bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) {
   // Make sure we don't add packets which are already too old to be decoded.
   if (!last_decoded_state_.in_initial_state()) {
     latest_received_sequence_number_ = LatestSequenceNumber(
-        latest_received_sequence_number_,
-        last_decoded_state_.sequence_num());
+        latest_received_sequence_number_, last_decoded_state_.sequence_num());
   }
   if (IsNewerSequenceNumber(sequence_number,
                             latest_received_sequence_number_)) {
@@ -1112,8 +1121,8 @@ bool VCMJitterBuffer::MissingTooOldPacket(
   if (missing_sequence_numbers_.empty()) {
     return false;
   }
-  const uint16_t age_of_oldest_missing_packet = latest_sequence_number -
-      *missing_sequence_numbers_.begin();
+  const uint16_t age_of_oldest_missing_packet =
+      latest_sequence_number - *missing_sequence_numbers_.begin();
   // Recycle frames if the NACK list contains too old sequence numbers as
   // the packets may have already been dropped by the sender.
   return age_of_oldest_missing_packet > max_packet_age_to_nack_;
@@ -1121,8 +1130,8 @@ bool VCMJitterBuffer::MissingTooOldPacket(
 
 bool VCMJitterBuffer::HandleTooOldPackets(uint16_t latest_sequence_number) {
   bool key_frame_found = false;
-  const uint16_t age_of_oldest_missing_packet = latest_sequence_number -
-      *missing_sequence_numbers_.begin();
+  const uint16_t age_of_oldest_missing_packet =
+      latest_sequence_number - *missing_sequence_numbers_.begin();
   LOG_F(LS_WARNING) << "NACK list contains too old sequence numbers: "
                     << age_of_oldest_missing_packet << " > "
                     << max_packet_age_to_nack_;
@@ -1136,9 +1145,9 @@ void VCMJitterBuffer::DropPacketsFromNackList(
     uint16_t last_decoded_sequence_number) {
   // Erase all sequence numbers from the NACK list which we won't need any
   // longer.
-  missing_sequence_numbers_.erase(missing_sequence_numbers_.begin(),
-                                  missing_sequence_numbers_.upper_bound(
-                                      last_decoded_sequence_number));
+  missing_sequence_numbers_.erase(
+      missing_sequence_numbers_.begin(),
+      missing_sequence_numbers_.upper_bound(last_decoded_sequence_number));
 }
 
 int64_t VCMJitterBuffer::LastDecodedTimestamp() const {
@@ -1222,11 +1231,11 @@ void VCMJitterBuffer::CountFrame(const VCMFrameBuffer& frame) {
   incoming_frame_count_++;
 
   if (frame.FrameType() == kVideoFrameKey) {
-    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video",
-                            frame.TimeStamp(), "KeyComplete");
+    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", frame.TimeStamp(),
+                            "KeyComplete");
   } else {
-    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video",
-                            frame.TimeStamp(), "DeltaComplete");
+    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", frame.TimeStamp(),
+                            "DeltaComplete");
   }
 
   // Update receive statistics. We count all layers, thus when you use layers
@@ -1244,13 +1253,13 @@ void VCMJitterBuffer::CountFrame(const VCMFrameBuffer& frame) {
 
 void VCMJitterBuffer::UpdateAveragePacketsPerFrame(int current_number_packets) {
   if (frame_counter_ > kFastConvergeThreshold) {
-    average_packets_per_frame_ = average_packets_per_frame_
-              * (1 - kNormalConvergeMultiplier)
-            + current_number_packets * kNormalConvergeMultiplier;
+    average_packets_per_frame_ =
+        average_packets_per_frame_ * (1 - kNormalConvergeMultiplier) +
+        current_number_packets * kNormalConvergeMultiplier;
   } else if (frame_counter_ > 0) {
-    average_packets_per_frame_ = average_packets_per_frame_
-              * (1 - kFastConvergeMultiplier)
-            + current_number_packets * kFastConvergeMultiplier;
+    average_packets_per_frame_ =
+        average_packets_per_frame_ * (1 - kFastConvergeMultiplier) +
+        current_number_packets * kFastConvergeMultiplier;
     frame_counter_++;
   } else {
     average_packets_per_frame_ = current_number_packets;
@@ -1272,7 +1281,7 @@ void VCMJitterBuffer::CleanUpOldOrEmptyFrames() {
 // Must be called from within |crit_sect_|.
 bool VCMJitterBuffer::IsPacketRetransmitted(const VCMPacket& packet) const {
   return missing_sequence_numbers_.find(packet.seqNum) !=
-      missing_sequence_numbers_.end();
+         missing_sequence_numbers_.end();
 }
 
 // Must be called under the critical section |crit_sect_|. Should never be
@@ -1304,18 +1313,16 @@ void VCMJitterBuffer::UpdateJitterEstimate(const VCMFrameBuffer& frame,
 // Must be called under the critical section |crit_sect_|. Should never be
 // called with retransmitted frames, they must be filtered out before this
 // function is called.
-void VCMJitterBuffer::UpdateJitterEstimate(
-    int64_t latest_packet_time_ms,
-    uint32_t timestamp,
-    unsigned int frame_size,
-    bool incomplete_frame) {
+void VCMJitterBuffer::UpdateJitterEstimate(int64_t latest_packet_time_ms,
+                                           uint32_t timestamp,
+                                           unsigned int frame_size,
+                                           bool incomplete_frame) {
   if (latest_packet_time_ms == -1) {
     return;
   }
   int64_t frame_delay;
-  bool not_reordered = inter_frame_delay_.CalculateDelay(timestamp,
-                                                      &frame_delay,
-                                                      latest_packet_time_ms);
+  bool not_reordered = inter_frame_delay_.CalculateDelay(
+      timestamp, &frame_delay, latest_packet_time_ms);
   // Filter out frames which have been reordered in time by the network
   if (not_reordered) {
     // Update the jitter estimate with the new samples
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/jitter_buffer.h
index f4a3638f7d..01e27752d2 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/webrtc/modules/video_coding/jitter_buffer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
 
 #include <list>
 #include <map>
@@ -18,22 +18,19 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/decoding_state.h"
-#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/decoding_state.h"
+#include "webrtc/modules/video_coding/inter_frame_delay.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-enum VCMNackMode {
-  kNack,
-  kNoNack
-};
+enum VCMNackMode { kNack, kNoNack };
 
 // forward declarations
 class Clock;
@@ -54,8 +51,7 @@ struct VCMJitterSample {
 
 class TimestampLessThan {
  public:
-  bool operator() (uint32_t timestamp1,
-                   uint32_t timestamp2) const {
+  bool operator()(uint32_t timestamp1, uint32_t timestamp2) const {
     return IsNewerTimestamp(timestamp2, timestamp1);
   }
 };
@@ -68,7 +64,7 @@ class FrameList
   VCMFrameBuffer* Front() const;
   VCMFrameBuffer* Back() const;
   int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
-      UnorderedFrameList* free_frames);
+                                 UnorderedFrameList* free_frames);
   void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
                                UnorderedFrameList* free_frames);
   void Reset(UnorderedFrameList* free_frames);
@@ -141,8 +137,7 @@ class VCMJitterBuffer {
   int num_discarded_packets() const;
 
   // Statistics, Calculate frame and bit rates.
-  void IncomingRateStatistics(unsigned int* framerate,
-                              unsigned int* bitrate);
+  void IncomingRateStatistics(unsigned int* framerate, unsigned int* bitrate);
 
   // Checks if the packet sequence will be complete if the next frame would be
   // grabbed for decoding. That is, if a frame has been lost between the
@@ -177,8 +172,7 @@ class VCMJitterBuffer {
   // Inserts a packet into a frame returned from GetFrame().
   // If the return value is <= 0, |frame| is invalidated and the pointer must
   // be dropped after this function returns.
-  VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
-                                  bool* retransmitted);
+  VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, bool* retransmitted);
 
   // Returns the estimated jitter in milliseconds.
   uint32_t EstimatedJitterMs();
@@ -192,7 +186,8 @@ class VCMJitterBuffer {
   // |low_rtt_nack_threshold_ms| is an RTT threshold in ms below which we expect
   // to rely on NACK only, and therefore are using larger buffers to have time
   // to wait for retransmissions.
-  void SetNackMode(VCMNackMode mode, int64_t low_rtt_nack_threshold_ms,
+  void SetNackMode(VCMNackMode mode,
+                   int64_t low_rtt_nack_threshold_ms,
                    int64_t high_rtt_nack_threshold_ms);
 
   void SetNackSettings(size_t max_nack_list_size,
@@ -209,7 +204,7 @@ class VCMJitterBuffer {
   // session. Changes will not influence frames already in the buffer.
   void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
   int64_t LastDecodedTimestamp() const;
-  VCMDecodeErrorMode decode_error_mode() const {return decode_error_mode_;}
+  VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
 
   // Used to compute time of complete continuous frames. Returns the timestamps
   // corresponding to the start and end of the continuous complete buffer.
@@ -220,8 +215,8 @@ class VCMJitterBuffer {
  private:
   class SequenceNumberLessThan {
    public:
-    bool operator() (const uint16_t& sequence_number1,
-                     const uint16_t& sequence_number2) const {
+    bool operator()(const uint16_t& sequence_number1,
+                    const uint16_t& sequence_number2) const {
       return IsNewerSequenceNumber(sequence_number2, sequence_number1);
     }
   };
@@ -338,8 +333,6 @@ class VCMJitterBuffer {
   FrameList incomplete_frames_ GUARDED_BY(crit_sect_);
   VCMDecodingState last_decoded_state_ GUARDED_BY(crit_sect_);
   bool first_packet_since_reset_;
-  // Contains scalability structure data for VP9.
-  Vp9SsMap vp9_ss_map_ GUARDED_BY(crit_sect_);
 
   // Statistics.
   VCMReceiveStatisticsCallback* stats_callback_ GUARDED_BY(crit_sect_);
@@ -393,4 +386,4 @@ class VCMJitterBuffer {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_common.h b/webrtc/modules/video_coding/jitter_buffer_common.h
index 97af78087a..65356f1d1b 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_common.h
+++ b/webrtc/modules/video_coding/jitter_buffer_common.h
@@ -19,11 +19,11 @@ namespace webrtc {
 static const float kFastConvergeMultiplier = 0.4f;
 static const float kNormalConvergeMultiplier = 0.2f;
 
-enum { kMaxNumberOfFrames     = 300 };
-enum { kStartNumberOfFrames   = 6 };
-enum { kMaxVideoDelayMs       = 10000 };
+enum { kMaxNumberOfFrames = 300 };
+enum { kStartNumberOfFrames = 6 };
+enum { kMaxVideoDelayMs = 10000 };
 enum { kPacketsPerFrameMultiplier = 5 };
-enum { kFastConvergeThreshold = 5};
+enum { kFastConvergeThreshold = 5 };
 
 enum VCMJitterBufferEnum {
   kMaxConsecutiveOldFrames = 60,
@@ -36,36 +36,36 @@ enum VCMJitterBufferEnum {
 };
 
 enum VCMFrameBufferEnum {
-  kOutOfBoundsPacket    = -7,
-  kNotInitialized       = -6,
-  kOldPacket            = -5,
-  kGeneralError         = -4,
-  kFlushIndicator       = -3,   // Indicator that a flush has occurred.
-  kTimeStampError       = -2,
-  kSizeError            = -1,
-  kNoError              = 0,
-  kIncomplete           = 1,    // Frame incomplete.
-  kCompleteSession      = 3,    // at least one layer in the frame complete.
-  kDecodableSession     = 4,    // Frame incomplete, but ready to be decoded
-  kDuplicatePacket      = 5     // We're receiving a duplicate packet.
+  kOutOfBoundsPacket = -7,
+  kNotInitialized = -6,
+  kOldPacket = -5,
+  kGeneralError = -4,
+  kFlushIndicator = -3,  // Indicator that a flush has occurred.
+  kTimeStampError = -2,
+  kSizeError = -1,
+  kNoError = 0,
+  kIncomplete = 1,        // Frame incomplete.
+  kCompleteSession = 3,   // at least one layer in the frame complete.
+  kDecodableSession = 4,  // Frame incomplete, but ready to be decoded
+  kDuplicatePacket = 5    // We're receiving a duplicate packet.
 };
 
 enum VCMFrameBufferStateEnum {
-  kStateEmpty,              // frame popped by the RTP receiver
-  kStateIncomplete,         // frame that have one or more packet(s) stored
-  kStateComplete,           // frame that have all packets
-  kStateDecodable           // Hybrid mode - frame can be decoded
+  kStateEmpty,       // frame popped by the RTP receiver
+  kStateIncomplete,  // frame that have one or more packet(s) stored
+  kStateComplete,    // frame that have all packets
+  kStateDecodable    // Hybrid mode - frame can be decoded
 };
 
-enum { kH264StartCodeLengthBytes = 4};
+enum { kH264StartCodeLengthBytes = 4 };
 
 // Used to indicate if a received packet contain a complete NALU (or equivalent)
 enum VCMNaluCompleteness {
-  kNaluUnset = 0,       // Packet has not been filled.
-  kNaluComplete = 1,    // Packet can be decoded as is.
-  kNaluStart,           // Packet contain beginning of NALU
-  kNaluIncomplete,      // Packet is not beginning or end of NALU
-  kNaluEnd,             // Packet is the end of a NALU
+  kNaluUnset = 0,     // Packet has not been filled.
+  kNaluComplete = 1,  // Packet can be decoded as is.
+  kNaluStart,         // Packet contain beginning of NALU
+  kNaluIncomplete,    // Packet is not beginning or end of NALU
+  kNaluEnd,           // Packet is the end of a NALU
 };
 }  // namespace webrtc
 
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/jitter_buffer_unittest.cc
index d6c6d4985b..8abc1b5471 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ b/webrtc/modules/video_coding/jitter_buffer_unittest.cc
@@ -13,12 +13,12 @@
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/test/stream_generator.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/media_opt_util.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/test/stream_generator.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/test/histogram.h"
@@ -26,13 +26,12 @@
 namespace webrtc {
 
 namespace {
-  const uint32_t kProcessIntervalSec = 60;
+const uint32_t kProcessIntervalSec = 60;
 }  // namespace
 
 class Vp9SsMapTest : public ::testing::Test {
  protected:
-  Vp9SsMapTest()
-      : packet_(data_, 1400, 1234, 1, true) {}
+  Vp9SsMapTest() : packet_(data_, 1400, 1234, 1, true) {}
 
   virtual void SetUp() {
     packet_.isFirstPacket = true;
@@ -234,8 +233,8 @@ class TestBasicJitterBuffer : public ::testing::Test {
   }
 
   void CheckOutFrame(VCMEncodedFrame* frame_out,
-                    unsigned int size,
-                    bool startCode) {
+                     unsigned int size,
+                     bool startCode) {
     ASSERT_TRUE(frame_out);
 
     const uint8_t* outData = frame_out->Buffer();
@@ -280,7 +279,6 @@ class TestBasicJitterBuffer : public ::testing::Test {
   rtc::scoped_ptr<VCMJitterBuffer> jitter_buffer_;
 };
 
-
 class TestRunningJitterBuffer : public ::testing::Test {
  protected:
   enum { kDataBufferSize = 10 };
@@ -294,8 +292,8 @@ class TestRunningJitterBuffer : public ::testing::Test {
         rtc::scoped_ptr<EventWrapper>(event_factory_.CreateEvent()));
     stream_generator_ = new StreamGenerator(0, clock_->TimeInMilliseconds());
     jitter_buffer_->Start();
-    jitter_buffer_->SetNackSettings(max_nack_list_size_,
-                                    oldest_packet_to_nack_, 0);
+    jitter_buffer_->SetNackSettings(max_nack_list_size_, oldest_packet_to_nack_,
+                                    0);
     memset(data_buffer_, 0, kDataBufferSize);
   }
 
@@ -396,9 +394,7 @@ class TestJitterBufferNack : public TestRunningJitterBuffer {
     jitter_buffer_->SetNackMode(kNack, -1, -1);
   }
 
-  virtual void TearDown() {
-    TestRunningJitterBuffer::TearDown();
-  }
+  virtual void TearDown() { TestRunningJitterBuffer::TearDown(); }
 };
 
 TEST_F(TestBasicJitterBuffer, StopRunning) {
@@ -431,8 +427,8 @@ TEST_F(TestBasicJitterBuffer, SinglePacketFrame) {
 
   // Insert the packet to the jitter buffer and get a frame.
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -450,8 +446,8 @@ TEST_F(TestBasicJitterBuffer, VerifyHistogramStats) {
 
   // Insert single packet frame to the jitter buffer and get a frame.
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -460,25 +456,25 @@ TEST_F(TestBasicJitterBuffer, VerifyHistogramStats) {
   // Verify that histograms are updated when the jitter buffer is stopped.
   clock_->AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000);
   jitter_buffer_->Stop();
-  EXPECT_EQ(0, test::LastHistogramSample(
-      "WebRTC.Video.DiscardedPacketsInPercent"));
-  EXPECT_EQ(0, test::LastHistogramSample(
-      "WebRTC.Video.DuplicatedPacketsInPercent"));
+  EXPECT_EQ(
+      0, test::LastHistogramSample("WebRTC.Video.DiscardedPacketsInPercent"));
+  EXPECT_EQ(
+      0, test::LastHistogramSample("WebRTC.Video.DuplicatedPacketsInPercent"));
   EXPECT_NE(-1, test::LastHistogramSample(
-      "WebRTC.Video.CompleteFramesReceivedPerSecond"));
+                    "WebRTC.Video.CompleteFramesReceivedPerSecond"));
   EXPECT_EQ(1000, test::LastHistogramSample(
-      "WebRTC.Video.KeyFramesReceivedInPermille"));
+                      "WebRTC.Video.KeyFramesReceivedInPermille"));
 
   // Verify that histograms are not updated if stop is called again.
   jitter_buffer_->Stop();
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.DiscardedPacketsInPercent"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.DuplicatedPacketsInPercent"));
   EXPECT_EQ(1, test::NumHistogramSamples(
-      "WebRTC.Video.DiscardedPacketsInPercent"));
-  EXPECT_EQ(1, test::NumHistogramSamples(
-      "WebRTC.Video.DuplicatedPacketsInPercent"));
-  EXPECT_EQ(1, test::NumHistogramSamples(
-      "WebRTC.Video.CompleteFramesReceivedPerSecond"));
-  EXPECT_EQ(1, test::NumHistogramSamples(
-      "WebRTC.Video.KeyFramesReceivedInPermille"));
+                   "WebRTC.Video.CompleteFramesReceivedPerSecond"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
 }
 
 TEST_F(TestBasicJitterBuffer, DualPacketFrame) {
@@ -487,8 +483,8 @@ TEST_F(TestBasicJitterBuffer, DualPacketFrame) {
   packet_->markerBit = false;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   // Should not be complete.
   EXPECT_TRUE(frame_out == NULL);
@@ -498,8 +494,8 @@ TEST_F(TestBasicJitterBuffer, DualPacketFrame) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
@@ -514,8 +510,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) {
   packet_->markerBit = false;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -530,8 +526,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     loop++;
   } while (loop < 98);
 
@@ -541,8 +537,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -558,8 +554,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
   packet_->markerBit = true;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_FALSE(frame_out == NULL);
   jitter_buffer_->ReleaseFrame(frame_out);
@@ -570,8 +566,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
   packet_->frameType = kVideoFrameDelta;
   packet_->timestamp += 33 * 90;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -586,8 +582,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
     packet_->seqNum = seq_num_;
 
     // Insert a packet into a frame.
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     loop++;
   } while (loop < 98);
 
@@ -597,8 +593,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -617,8 +613,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -632,8 +628,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     loop++;
   } while (loop < 98);
 
@@ -643,10 +639,10 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
-  frame_out = DecodeCompleteFrame();;
+  frame_out = DecodeCompleteFrame();
 
   CheckOutFrame(frame_out, 100 * size_, false);
 
@@ -660,8 +656,8 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
   packet_->markerBit = false;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -672,23 +668,23 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   // check that we fail to get frame since seqnum is not continuous
   frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
 
   seq_num_ -= 3;
-  timestamp_ -= 33*90;
+  timestamp_ -= 33 * 90;
   packet_->frameType = kVideoFrameKey;
   packet_->isFirstPacket = true;
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -700,8 +696,8 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
@@ -781,8 +777,8 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
   EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -791,8 +787,8 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
   EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
 
   // Insert a packet into a frame.
-  EXPECT_EQ(kDuplicatePacket, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kDuplicatePacket,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_EQ(2, jitter_buffer_->num_packets());
   EXPECT_EQ(1, jitter_buffer_->num_duplicated_packets());
 
@@ -801,8 +797,8 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
   packet_->markerBit = true;
   packet_->isFirstPacket = false;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   ASSERT_TRUE(frame_out != NULL);
@@ -885,7 +881,6 @@ TEST_F(TestBasicJitterBuffer, TestSkipForwardVp9) {
   packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
   packet_->codecSpecificHeader.codecHeader.VP9.beginning_of_frame = true;
   packet_->codecSpecificHeader.codecHeader.VP9.end_of_frame = true;
-  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = kNoTemporalIdx;
   packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
 
   packet_->seqNum = 65485;
@@ -893,7 +888,7 @@ TEST_F(TestBasicJitterBuffer, TestSkipForwardVp9) {
   packet_->frameType = kVideoFrameKey;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
   packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 200;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
   packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
   packet_->codecSpecificHeader.codecHeader.VP9.gof.SetGofInfoVP9(
       kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
@@ -905,7 +900,7 @@ TEST_F(TestBasicJitterBuffer, TestSkipForwardVp9) {
   packet_->frameType = kVideoFrameDelta;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 9;
   packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 201;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
   packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = false;
   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
 
@@ -939,22 +934,22 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_3TlLayers) {
   packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
   packet_->codecSpecificHeader.codecHeader.VP9.beginning_of_frame = true;
   packet_->codecSpecificHeader.codecHeader.VP9.end_of_frame = true;
-  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = kNoTemporalIdx;
-  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
   packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 200;
 
   packet_->seqNum = 65486;
   packet_->timestamp = 6000;
   packet_->frameType = kVideoFrameDelta;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 6;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 2;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
 
   packet_->seqNum = 65487;
   packet_->timestamp = 9000;
   packet_->frameType = kVideoFrameDelta;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 7;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 2;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
 
   // Insert first frame with SS data.
@@ -964,7 +959,8 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_3TlLayers) {
   packet_->width = 352;
   packet_->height = 288;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
   packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
   packet_->codecSpecificHeader.codecHeader.VP9.gof.SetGofInfoVP9(
       kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
@@ -1011,8 +1007,6 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
   packet_->codecSpecificHeader.codecHeader.VP9.flexible_mode = false;
   packet_->codecSpecificHeader.codecHeader.VP9.beginning_of_frame = true;
   packet_->codecSpecificHeader.codecHeader.VP9.end_of_frame = true;
-  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = kNoTemporalIdx;
-  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
   packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 200;
 
   packet_->isFirstPacket = true;
@@ -1022,7 +1016,8 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
   packet_->frameType = kVideoFrameDelta;
   packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 6;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
   EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
 
   packet_->isFirstPacket = false;
@@ -1031,7 +1026,8 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
   packet_->frameType = kVideoFrameDelta;
   packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 1;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 6;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
 
   packet_->isFirstPacket = false;
@@ -1041,7 +1037,8 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
   packet_->frameType = kVideoFrameKey;
   packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 1;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
   EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
 
   // Insert first frame with SS data.
@@ -1053,7 +1050,8 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
   packet_->height = 288;
   packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
   packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
-  packet_->codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
   packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
   packet_->codecSpecificHeader.codecHeader.VP9.gof.SetGofInfoVP9(
       kTemporalStructureMode2);  // kTemporalStructureMode3: 0-1-0-1..
@@ -1084,8 +1082,8 @@ TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
   packet_->insertStartCode = true;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -1097,8 +1095,8 @@ TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_ * 2 + 4 * 2, true);
@@ -1118,8 +1116,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t timestamp = 0;
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
@@ -1127,8 +1125,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->isFirstPacket = false;
   for (int i = 1; i < 9; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
@@ -1137,8 +1135,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->markerBit = true;
   packet_->seqNum++;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 10 * size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -1152,8 +1150,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->seqNum += 100;
   packet_->timestamp += 33 * 90 * 8;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -1161,23 +1159,23 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->seqNum -= 99;
   packet_->timestamp -= 33 * 90 * 7;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
   packet_->isFirstPacket = false;
   for (int i = 1; i < 8; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kDecodableSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
 
   packet_->seqNum++;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -1189,8 +1187,7 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
 
   packet_->markerBit = true;
   packet_->seqNum++;
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 }
 
 // Make sure first packet is present before a frame can be decoded.
@@ -1204,8 +1201,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -1217,9 +1214,9 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->isFirstPacket = false;
   packet_->markerBit = false;
   packet_->seqNum += 100;
-  packet_->timestamp += 33*90*8;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  packet_->timestamp += 33 * 90 * 8;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t timestamp;
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
@@ -1228,10 +1225,10 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->frameType = kVideoFrameKey;
   packet_->isFirstPacket = true;
   packet_->seqNum -= 99;
-  packet_->timestamp -= 33*90*7;
+  packet_->timestamp -= 33 * 90 * 7;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -1240,8 +1237,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->isFirstPacket = false;
   for (int i = 1; i < 5; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
@@ -1249,8 +1246,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   // Complete key frame.
   packet_->markerBit = true;
   packet_->seqNum++;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 6 * size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -1268,8 +1265,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -1281,9 +1278,9 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   packet_->isFirstPacket = false;
   packet_->markerBit = false;
   packet_->seqNum += 100;
-  packet_->timestamp += 33*90*8;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  packet_->timestamp += 33 * 90 * 8;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t timestamp;
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
@@ -1291,17 +1288,17 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   // Insert second frame with the first packet missing. Make sure we're waiting
   // for the key frame to be complete.
   packet_->seqNum -= 98;
-  packet_->timestamp -= 33*90*7;
+  packet_->timestamp -= 33 * 90 * 7;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
   for (int i = 0; i < 5; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
@@ -1309,8 +1306,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   // Add first packet. Frame should now be decodable, but incomplete.
   packet_->isFirstPacket = true;
   packet_->seqNum -= 6;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -1329,8 +1326,8 @@ TEST_F(TestBasicJitterBuffer, DiscontinuousStreamWhenDecodingWithErrors) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t next_timestamp;
   EXPECT_TRUE(jitter_buffer_->NextCompleteTimestamp(0, &next_timestamp));
   EXPECT_EQ(packet_->timestamp, next_timestamp);
@@ -1346,8 +1343,8 @@ TEST_F(TestBasicJitterBuffer, DiscontinuousStreamWhenDecodingWithErrors) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // Insert a packet (so the previous one will be released).
   timestamp_ += 33 * 90;
   seq_num_ += 2;
@@ -1356,8 +1353,8 @@ TEST_F(TestBasicJitterBuffer, DiscontinuousStreamWhenDecodingWithErrors) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &next_timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&next_timestamp));
   EXPECT_EQ(packet_->timestamp - 33 * 90, next_timestamp);
@@ -1382,12 +1379,12 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
   packet_->completeNALU = kNaluStart;
 
   bool retransmitted = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   for (int i = 0; i < 11; ++i) {
     webrtc::FrameType frametype = kVideoFrameDelta;
     seq_num_++;
-    timestamp_ += 33*90;
+    timestamp_ += 33 * 90;
     packet_->frameType = frametype;
     packet_->isFirstPacket = true;
     packet_->markerBit = false;
@@ -1395,8 +1392,8 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
     packet_->timestamp = timestamp_;
     packet_->completeNALU = kNaluStart;
 
-    EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kDecodableSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -1430,9 +1427,9 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
       CheckOutFrame(frame_out, size_, false);
 
       if (i == 0) {
-          EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
+        EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
       } else {
-         EXPECT_EQ(frametype, frame_out->FrameType());
+        EXPECT_EQ(frametype, frame_out->FrameType());
       }
       EXPECT_FALSE(frame_out->Complete());
       EXPECT_FALSE(frame_out->MissingFrame());
@@ -1446,18 +1443,15 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
   timestamp_ -= 33 * 90;
   packet_->timestamp = timestamp_ - 1000;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   packet_->timestamp = timestamp_ - 500;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   packet_->timestamp = timestamp_ - 100;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   EXPECT_EQ(3, jitter_buffer_->num_discarded_packets());
 
@@ -1476,8 +1470,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -1490,8 +1484,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     frame_out = DecodeCompleteFrame();
 
@@ -1505,8 +1499,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -1525,8 +1519,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) {
   packet_->seqNum = seq_num_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
   // Should not be complete.
@@ -1540,8 +1534,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     frame_out = DecodeCompleteFrame();
 
@@ -1556,8 +1550,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 100 * size_, false);
@@ -1579,8 +1573,8 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrame) {
   packet_->seqNum = seq_num_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(3000u, frame_out->TimeStamp());
@@ -1596,8 +1590,7 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrame) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 }
 
 TEST_F(TestBasicJitterBuffer, TestInsertOldFrameWithSeqNumWrap) {
@@ -1615,8 +1608,8 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrameWithSeqNumWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(timestamp_, frame_out->TimeStamp());
@@ -1635,10 +1628,8 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrameWithSeqNumWrap) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-
   // This timestamp is old.
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 }
 
 TEST_F(TestBasicJitterBuffer, TimestampWrap) {
@@ -1655,8 +1646,8 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
@@ -1666,23 +1657,23 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
   jitter_buffer_->ReleaseFrame(frame_out);
 
   seq_num_++;
-  timestamp_ += 33*90;
+  timestamp_ += 33 * 90;
   packet_->frameType = kVideoFrameDelta;
   packet_->isFirstPacket = true;
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
@@ -1692,8 +1683,8 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
@@ -1715,8 +1706,8 @@ TEST_F(TestBasicJitterBuffer, 2FrameWithTimestampWrap) {
 
   bool retransmitted = false;
   // Insert first frame (session will be complete).
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   // Insert next frame.
   seq_num_++;
@@ -1727,8 +1718,8 @@ TEST_F(TestBasicJitterBuffer, 2FrameWithTimestampWrap) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(0xffffff00, frame_out->TimeStamp());
@@ -1758,8 +1749,8 @@ TEST_F(TestBasicJitterBuffer, Insert2FramesReOrderedWithTimestampWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   // Insert second frame
   seq_num_--;
@@ -1770,8 +1761,8 @@ TEST_F(TestBasicJitterBuffer, Insert2FramesReOrderedWithTimestampWrap) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(0xffffff00, frame_out->TimeStamp());
@@ -1798,12 +1789,12 @@ TEST_F(TestBasicJitterBuffer, DeltaFrameWithMoreThanMaxNumberOfPackets) {
     packet_->seqNum = seq_num_;
 
     if (firstPacket) {
-      EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+      EXPECT_EQ(kIncomplete,
+                jitter_buffer_->InsertPacket(*packet_, &retransmitted));
       firstPacket = false;
     } else {
-      EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                          &retransmitted));
+      EXPECT_EQ(kIncomplete,
+                jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     }
 
     loop++;
@@ -1817,10 +1808,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrameWithMoreThanMaxNumberOfPackets) {
   packet_->seqNum = seq_num_;
 
   // Insert the packet -> frame recycled.
-  EXPECT_EQ(kSizeError, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kSizeError, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_TRUE(NULL == DecodeCompleteFrame());
-
 }
 
 TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
@@ -1832,13 +1821,18 @@ TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
   //  --------------------------------------------------------------
   // |<-----------delta frames------------->|<------key frames----->|
 
+  // Make sure the jitter doesn't request a keyframe after too much non-
+  // decodable frames.
+  jitter_buffer_->SetNackMode(kNack, -1, -1);
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
+
   int loop = 0;
   seq_num_ = 65485;
   uint32_t first_key_frame_timestamp = 0;
   bool retransmitted = false;
   // Insert MAX_NUMBER_OF_FRAMES frames.
   do {
-    timestamp_ += 33*90;
+    timestamp_ += 33 * 90;
     seq_num_++;
     packet_->isFirstPacket = true;
     packet_->markerBit = true;
@@ -1851,8 +1845,8 @@ TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
     }
 
     // Insert frame.
-    EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                             &retransmitted));
+    EXPECT_EQ(kCompleteSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     loop++;
   } while (loop < kMaxNumberOfFrames);
@@ -1860,7 +1854,7 @@ TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
   // Max number of frames inserted.
 
   // Insert one more frame.
-  timestamp_ += 33*90;
+  timestamp_ += 33 * 90;
   seq_num_++;
   packet_->isFirstPacket = true;
   packet_->markerBit = true;
@@ -1894,8 +1888,7 @@ TEST_F(TestBasicJitterBuffer, EmptyLastFrame) {
     packet_->timestamp = timestamp_;
     packet_->frameType = kEmptyFrame;
 
-    EXPECT_EQ(kNoError, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+    EXPECT_EQ(kNoError, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     VCMEncodedFrame* testFrame = DecodeIncompleteFrame();
     // Timestamp should never be the last TS inserted.
     if (testFrame != NULL) {
@@ -1919,8 +1912,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->markerBit = false;
   bool retransmitted = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_ += 2;  // Skip one packet.
   packet_->seqNum = seq_num_;
@@ -1929,8 +1922,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluIncomplete;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_++;
   packet_->seqNum = seq_num_;
@@ -1939,15 +1932,15 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluEnd;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_++;
   packet_->seqNum = seq_num_;
   packet_->completeNALU = kNaluComplete;
   packet_->markerBit = true;  // Last packet.
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // The JB will only output (incomplete) frames if a packet belonging to a
   // subsequent frame was already inserted. Insert one packet of a subsequent
   // frame. place high timestamp so the JB would always have a next frame
@@ -1960,8 +1953,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluStart;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeIncompleteFrame();
 
@@ -1973,7 +1966,7 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
 
   // Test reordered start frame + 1 lost.
   seq_num_ += 2;  // Re-order 1 frame.
-  timestamp_ += 33*90;
+  timestamp_ += 33 * 90;
   insertedLength = 0;
 
   packet_->seqNum = seq_num_;
@@ -1982,9 +1975,9 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluEnd;
   packet_->markerBit = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
-  insertedLength += packet_->sizeBytes; // This packet should be decoded.
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  insertedLength += packet_->sizeBytes;  // This packet should be decoded.
   seq_num_--;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
@@ -1993,8 +1986,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluStart;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   insertedLength += packet_->sizeBytes;  // This packet should be decoded.
 
   seq_num_ += 3;  // One packet drop.
@@ -2004,8 +1997,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluComplete;
   packet_->markerBit = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   insertedLength += packet_->sizeBytes;  // This packet should be decoded.
   seq_num_++;
   packet_->seqNum = seq_num_;
@@ -2014,8 +2007,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluStart;
   packet_->markerBit = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // This packet should be decoded since it's the beginning of a NAL.
   insertedLength += packet_->sizeBytes;
 
@@ -2026,8 +2019,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluEnd;
   packet_->markerBit = true;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // This packet should not be decoded because it is an incomplete NAL if it
   // is the last.
   frame_out = DecodeIncompleteFrame();
@@ -2045,8 +2038,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   emptypacket.isFirstPacket = true;
   emptypacket.completeNALU = kNaluComplete;
   emptypacket.markerBit = true;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(emptypacket,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(emptypacket, &retransmitted));
   // This packet should not be decoded because it is an incomplete NAL if it
   // is the last.
 
@@ -2067,8 +2060,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluComplete;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_++;
   emptypacket.seqNum = seq_num_;
@@ -2077,8 +2070,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   emptypacket.isFirstPacket = true;
   emptypacket.completeNALU = kNaluComplete;
   emptypacket.markerBit = true;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(emptypacket,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(emptypacket, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   // Only last NALU is complete
@@ -2097,8 +2090,8 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
   packet_->markerBit = true;
   bool retransmitted = false;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out != NULL);
   jitter_buffer_->ReleaseFrame(frame_out);
@@ -2109,9 +2102,8 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
   packet_->isFirstPacket = false;
   packet_->markerBit = false;
 
-
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeIncompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
@@ -2120,8 +2112,8 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
   packet_->timestamp += 33 * 90;
   packet_->isFirstPacket = true;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeIncompleteFrame();
   CheckOutFrame(frame_out, packet_->sizeBytes, false);
@@ -2129,6 +2121,10 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
 }
 
 TEST_F(TestRunningJitterBuffer, Full) {
+  // Make sure the jitter doesn't request a keyframe after too much non-
+  // decodable frames.
+  jitter_buffer_->SetNackMode(kNack, -1, -1);
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
   // Insert a key frame and decode it.
   EXPECT_GE(InsertFrame(kVideoFrameKey), kNoError);
   EXPECT_TRUE(DecodeCompleteFrame());
@@ -2277,8 +2273,8 @@ TEST_F(TestJitterBufferNack, NackTooOldPackets) {
   // old packet.
   DropFrame(1);
   // Insert a frame which should trigger a recycle until the next key frame.
-  EXPECT_EQ(kFlushIndicator, InsertFrames(oldest_packet_to_nack_ + 1,
-                                          kVideoFrameDelta));
+  EXPECT_EQ(kFlushIndicator,
+            InsertFrames(oldest_packet_to_nack_ + 1, kVideoFrameDelta));
   EXPECT_FALSE(DecodeCompleteFrame());
 
   bool request_key_frame = false;
@@ -2369,7 +2365,7 @@ TEST_F(TestJitterBufferNack, NackListBuiltBeforeFirstDecode) {
   stream_generator_->Init(0, clock_->TimeInMilliseconds());
   InsertFrame(kVideoFrameKey);
   stream_generator_->GenerateFrame(kVideoFrameDelta, 2, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   stream_generator_->NextPacket(NULL);  // Drop packet.
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
   EXPECT_TRUE(DecodeCompleteFrame());
@@ -2397,8 +2393,8 @@ TEST_F(TestJitterBufferNack, VerifyRetransmittedFlag) {
   EXPECT_EQ(1u, nack_list.size());
   stream_generator_->PopPacket(&packet, 0);
   EXPECT_EQ(packet.seqNum, nack_list[0]);
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(packet,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(packet, &retransmitted));
   EXPECT_TRUE(retransmitted);
   EXPECT_TRUE(DecodeCompleteFrame());
 }
@@ -2406,7 +2402,7 @@ TEST_F(TestJitterBufferNack, VerifyRetransmittedFlag) {
 TEST_F(TestJitterBufferNack, UseNackToRecoverFirstKeyFrame) {
   stream_generator_->Init(0, clock_->TimeInMilliseconds());
   stream_generator_->GenerateFrame(kVideoFrameKey, 3, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
   // Drop second packet.
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(1));
@@ -2454,7 +2450,7 @@ TEST_F(TestJitterBufferNack, NormalOperation) {
   // | 1 | 2 | .. | 8 | 9 | x | 11 | 12 | .. | 19 | x | 21 | .. | 100 |
   //  ----------------------------------------------------------------
   stream_generator_->GenerateFrame(kVideoFrameKey, 100, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
   EXPECT_EQ(kDecodableSession, InsertPacketAndPop(0));
   // Verify that the frame is incomplete.
@@ -2490,7 +2486,7 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap) {
   EXPECT_FALSE(request_key_frame);
   EXPECT_TRUE(DecodeCompleteFrame());
   stream_generator_->GenerateFrame(kVideoFrameDelta, 100, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
   while (stream_generator_->PacketsRemaining() > 1) {
     if (stream_generator_->NextSequenceNumber() % 10 != 0) {
@@ -2527,7 +2523,7 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap2) {
                                    clock_->TimeInMilliseconds());
   clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
   for (int i = 0; i < 5; ++i) {
-    if (stream_generator_->NextSequenceNumber()  != 65535) {
+    if (stream_generator_->NextSequenceNumber() != 65535) {
       EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
       EXPECT_FALSE(request_key_frame);
     } else {
diff --git a/webrtc/modules/video_coding/jitter_estimator.cc b/webrtc/modules/video_coding/jitter_estimator.cc
new file mode 100644
index 0000000000..8270c60e01
--- /dev/null
+++ b/webrtc/modules/video_coding/jitter_estimator.cc
@@ -0,0 +1,443 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/jitter_estimator.h"
+
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <string>
+
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/rtt_filter.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+enum { kStartupDelaySamples = 30 };
+enum { kFsAccuStartupSamples = 5 };
+enum { kMaxFramerateEstimate = 200 };
+
+VCMJitterEstimator::VCMJitterEstimator(const Clock* clock,
+                                       int32_t vcmId,
+                                       int32_t receiverId)
+    : _vcmId(vcmId),
+      _receiverId(receiverId),
+      _phi(0.97),
+      _psi(0.9999),
+      _alphaCountMax(400),
+      _thetaLow(0.000001),
+      _nackLimit(3),
+      _numStdDevDelayOutlier(15),
+      _numStdDevFrameSizeOutlier(3),
+      _noiseStdDevs(2.33),       // ~Less than 1% chance
+                                 // (look up in normal distribution table)...
+      _noiseStdDevOffset(30.0),  // ...of getting 30 ms freezes
+      _rttFilter(),
+      fps_counter_(30),  // TODO(sprang): Use an estimator with limit based on
+                         // time, rather than number of samples.
+      low_rate_experiment_(kInit),
+      clock_(clock) {
+  Reset();
+}
+
+VCMJitterEstimator::~VCMJitterEstimator() {}
+
+VCMJitterEstimator& VCMJitterEstimator::operator=(
+    const VCMJitterEstimator& rhs) {
+  if (this != &rhs) {
+    memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov));
+    memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov));
+
+    _vcmId = rhs._vcmId;
+    _receiverId = rhs._receiverId;
+    _avgFrameSize = rhs._avgFrameSize;
+    _varFrameSize = rhs._varFrameSize;
+    _maxFrameSize = rhs._maxFrameSize;
+    _fsSum = rhs._fsSum;
+    _fsCount = rhs._fsCount;
+    _lastUpdateT = rhs._lastUpdateT;
+    _prevEstimate = rhs._prevEstimate;
+    _prevFrameSize = rhs._prevFrameSize;
+    _avgNoise = rhs._avgNoise;
+    _alphaCount = rhs._alphaCount;
+    _filterJitterEstimate = rhs._filterJitterEstimate;
+    _startupCount = rhs._startupCount;
+    _latestNackTimestamp = rhs._latestNackTimestamp;
+    _nackCount = rhs._nackCount;
+    _rttFilter = rhs._rttFilter;
+  }
+  return *this;
+}
+
+// Resets the JitterEstimate
+void VCMJitterEstimator::Reset() {
+  _theta[0] = 1 / (512e3 / 8);
+  _theta[1] = 0;
+  _varNoise = 4.0;
+
+  _thetaCov[0][0] = 1e-4;
+  _thetaCov[1][1] = 1e2;
+  _thetaCov[0][1] = _thetaCov[1][0] = 0;
+  _Qcov[0][0] = 2.5e-10;
+  _Qcov[1][1] = 1e-10;
+  _Qcov[0][1] = _Qcov[1][0] = 0;
+  _avgFrameSize = 500;
+  _maxFrameSize = 500;
+  _varFrameSize = 100;
+  _lastUpdateT = -1;
+  _prevEstimate = -1.0;
+  _prevFrameSize = 0;
+  _avgNoise = 0.0;
+  _alphaCount = 1;
+  _filterJitterEstimate = 0.0;
+  _latestNackTimestamp = 0;
+  _nackCount = 0;
+  _fsSum = 0;
+  _fsCount = 0;
+  _startupCount = 0;
+  _rttFilter.Reset();
+  fps_counter_.Reset();
+}
+
+void VCMJitterEstimator::ResetNackCount() {
+  _nackCount = 0;
+}
+
+// Updates the estimates with the new measurements
+void VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS,
+                                        uint32_t frameSizeBytes,
+                                        bool incompleteFrame /* = false */) {
+  if (frameSizeBytes == 0) {
+    return;
+  }
+  int deltaFS = frameSizeBytes - _prevFrameSize;
+  if (_fsCount < kFsAccuStartupSamples) {
+    _fsSum += frameSizeBytes;
+    _fsCount++;
+  } else if (_fsCount == kFsAccuStartupSamples) {
+    // Give the frame size filter
+    _avgFrameSize = static_cast<double>(_fsSum) / static_cast<double>(_fsCount);
+    _fsCount++;
+  }
+  if (!incompleteFrame || frameSizeBytes > _avgFrameSize) {
+    double avgFrameSize = _phi * _avgFrameSize + (1 - _phi) * frameSizeBytes;
+    if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize)) {
+      // Only update the average frame size if this sample wasn't a
+      // key frame
+      _avgFrameSize = avgFrameSize;
+    }
+    // Update the variance anyway since we want to capture cases where we only
+    // get
+    // key frames.
+    _varFrameSize = VCM_MAX(_phi * _varFrameSize +
+                                (1 - _phi) * (frameSizeBytes - avgFrameSize) *
+                                    (frameSizeBytes - avgFrameSize),
+                            1.0);
+  }
+
+  // Update max frameSize estimate
+  _maxFrameSize =
+      VCM_MAX(_psi * _maxFrameSize, static_cast<double>(frameSizeBytes));
+
+  if (_prevFrameSize == 0) {
+    _prevFrameSize = frameSizeBytes;
+    return;
+  }
+  _prevFrameSize = frameSizeBytes;
+
+  // Only update the Kalman filter if the sample is not considered
+  // an extreme outlier. Even if it is an extreme outlier from a
+  // delay point of view, if the frame size also is large the
+  // deviation is probably due to an incorrect line slope.
+  double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
+
+  if (fabs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
+      frameSizeBytes >
+          _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize)) {
+    // Update the variance of the deviation from the
+    // line given by the Kalman filter
+    EstimateRandomJitter(deviation, incompleteFrame);
+    // Prevent updating with frames which have been congested by a large
+    // frame, and therefore arrives almost at the same time as that frame.
+    // This can occur when we receive a large frame (key frame) which
+    // has been delayed. The next frame is of normal size (delta frame),
+    // and thus deltaFS will be << 0. This removes all frame samples
+    // which arrives after a key frame.
+    if ((!incompleteFrame || deviation >= 0.0) &&
+        static_cast<double>(deltaFS) > -0.25 * _maxFrameSize) {
+      // Update the Kalman filter with the new data
+      KalmanEstimateChannel(frameDelayMS, deltaFS);
+    }
+  } else {
+    int nStdDev =
+        (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier;
+    EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame);
+  }
+  // Post process the total estimated jitter
+  if (_startupCount >= kStartupDelaySamples) {
+    PostProcessEstimate();
+  } else {
+    _startupCount++;
+  }
+}
+
+// Updates the nack/packet ratio
+void VCMJitterEstimator::FrameNacked() {
+  // Wait until _nackLimit retransmissions has been received,
+  // then always add ~1 RTT delay.
+  // TODO(holmer): Should we ever remove the additional delay if the
+  // the packet losses seem to have stopped? We could for instance scale
+  // the number of RTTs to add with the amount of retransmissions in a given
+  // time interval, or similar.
+  if (_nackCount < _nackLimit) {
+    _nackCount++;
+  }
+}
+
+// Updates Kalman estimate of the channel
+// The caller is expected to sanity check the inputs.
+void VCMJitterEstimator::KalmanEstimateChannel(int64_t frameDelayMS,
+                                               int32_t deltaFSBytes) {
+  double Mh[2];
+  double hMh_sigma;
+  double kalmanGain[2];
+  double measureRes;
+  double t00, t01;
+
+  // Kalman filtering
+
+  // Prediction
+  // M = M + Q
+  _thetaCov[0][0] += _Qcov[0][0];
+  _thetaCov[0][1] += _Qcov[0][1];
+  _thetaCov[1][0] += _Qcov[1][0];
+  _thetaCov[1][1] += _Qcov[1][1];
+
+  // Kalman gain
+  // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h')
+  // h = [dFS 1]
+  // Mh = M*h'
+  // hMh_sigma = h*M*h' + R
+  Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1];
+  Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1];
+  // sigma weights measurements with a small deltaFS as noisy and
+  // measurements with large deltaFS as good
+  if (_maxFrameSize < 1.0) {
+    return;
+  }
+  double sigma = (300.0 * exp(-fabs(static_cast<double>(deltaFSBytes)) /
+                              (1e0 * _maxFrameSize)) +
+                  1) *
+                 sqrt(_varNoise);
+  if (sigma < 1.0) {
+    sigma = 1.0;
+  }
+  hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma;
+  if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) ||
+      (hMh_sigma > -1e-9 && hMh_sigma <= 0)) {
+    assert(false);
+    return;
+  }
+  kalmanGain[0] = Mh[0] / hMh_sigma;
+  kalmanGain[1] = Mh[1] / hMh_sigma;
+
+  // Correction
+  // theta = theta + K*(dT - h*theta)
+  measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]);
+  _theta[0] += kalmanGain[0] * measureRes;
+  _theta[1] += kalmanGain[1] * measureRes;
+
+  if (_theta[0] < _thetaLow) {
+    _theta[0] = _thetaLow;
+  }
+
+  // M = (I - K*h)*M
+  t00 = _thetaCov[0][0];
+  t01 = _thetaCov[0][1];
+  _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 -
+                    kalmanGain[0] * _thetaCov[1][0];
+  _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 -
+                    kalmanGain[0] * _thetaCov[1][1];
+  _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) -
+                    kalmanGain[1] * deltaFSBytes * t00;
+  _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) -
+                    kalmanGain[1] * deltaFSBytes * t01;
+
+  // Covariance matrix, must be positive semi-definite
+  assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 &&
+         _thetaCov[0][0] * _thetaCov[1][1] -
+                 _thetaCov[0][1] * _thetaCov[1][0] >=
+             0 &&
+         _thetaCov[0][0] >= 0);
+}
+
+// Calculate difference in delay between a sample and the
+// expected delay estimated by the Kalman filter
+double VCMJitterEstimator::DeviationFromExpectedDelay(
+    int64_t frameDelayMS,
+    int32_t deltaFSBytes) const {
+  return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]);
+}
+
+// Estimates the random jitter by calculating the variance of the
+// sample distance from the line given by theta.
+void VCMJitterEstimator::EstimateRandomJitter(double d_dT,
+                                              bool incompleteFrame) {
+  uint64_t now = clock_->TimeInMicroseconds();
+  if (_lastUpdateT != -1) {
+    fps_counter_.AddSample(now - _lastUpdateT);
+  }
+  _lastUpdateT = now;
+
+  if (_alphaCount == 0) {
+    assert(false);
+    return;
+  }
+  double alpha =
+      static_cast<double>(_alphaCount - 1) / static_cast<double>(_alphaCount);
+  _alphaCount++;
+  if (_alphaCount > _alphaCountMax)
+    _alphaCount = _alphaCountMax;
+
+  if (LowRateExperimentEnabled()) {
+    // In order to avoid a low frame rate stream to react slower to changes,
+    // scale the alpha weight relative a 30 fps stream.
+    double fps = GetFrameRate();
+    if (fps > 0.0) {
+      double rate_scale = 30.0 / fps;
+      // At startup, there can be a lot of noise in the fps estimate.
+      // Interpolate rate_scale linearly, from 1.0 at sample #1, to 30.0 / fps
+      // at sample #kStartupDelaySamples.
+      if (_alphaCount < kStartupDelaySamples) {
+        rate_scale =
+            (_alphaCount * rate_scale + (kStartupDelaySamples - _alphaCount)) /
+            kStartupDelaySamples;
+      }
+      alpha = pow(alpha, rate_scale);
+    }
+  }
+
+  double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
+  double varNoise =
+      alpha * _varNoise + (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise);
+  if (!incompleteFrame || varNoise > _varNoise) {
+    _avgNoise = avgNoise;
+    _varNoise = varNoise;
+  }
+  if (_varNoise < 1.0) {
+    // The variance should never be zero, since we might get
+    // stuck and consider all samples as outliers.
+    _varNoise = 1.0;
+  }
+}
+
+double VCMJitterEstimator::NoiseThreshold() const {
+  double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset;
+  if (noiseThreshold < 1.0) {
+    noiseThreshold = 1.0;
+  }
+  return noiseThreshold;
+}
+
+// Calculates the current jitter estimate from the filtered estimates
+double VCMJitterEstimator::CalculateEstimate() {
+  double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold();
+
+  // A very low estimate (or negative) is neglected
+  if (ret < 1.0) {
+    if (_prevEstimate <= 0.01) {
+      ret = 1.0;
+    } else {
+      ret = _prevEstimate;
+    }
+  }
+  if (ret > 10000.0) {  // Sanity
+    ret = 10000.0;
+  }
+  _prevEstimate = ret;
+  return ret;
+}
+
+void VCMJitterEstimator::PostProcessEstimate() {
+  _filterJitterEstimate = CalculateEstimate();
+}
+
+void VCMJitterEstimator::UpdateRtt(int64_t rttMs) {
+  _rttFilter.Update(rttMs);
+}
+
+void VCMJitterEstimator::UpdateMaxFrameSize(uint32_t frameSizeBytes) {
+  if (_maxFrameSize < frameSizeBytes) {
+    _maxFrameSize = frameSizeBytes;
+  }
+}
+
+// Returns the current filtered estimate if available,
+// otherwise tries to calculate an estimate.
+int VCMJitterEstimator::GetJitterEstimate(double rttMultiplier) {
+  double jitterMS = CalculateEstimate() + OPERATING_SYSTEM_JITTER;
+  if (_filterJitterEstimate > jitterMS)
+    jitterMS = _filterJitterEstimate;
+  if (_nackCount >= _nackLimit)
+    jitterMS += _rttFilter.RttMs() * rttMultiplier;
+
+  if (LowRateExperimentEnabled()) {
+    static const double kJitterScaleLowThreshold = 5.0;
+    static const double kJitterScaleHighThreshold = 10.0;
+    double fps = GetFrameRate();
+    // Ignore jitter for very low fps streams.
+    if (fps < kJitterScaleLowThreshold) {
+      if (fps == 0.0) {
+        return jitterMS;
+      }
+      return 0;
+    }
+
+    // Semi-low frame rate; scale by factor linearly interpolated from 0.0 at
+    // kJitterScaleLowThreshold to 1.0 at kJitterScaleHighThreshold.
+    if (fps < kJitterScaleHighThreshold) {
+      jitterMS =
+          (1.0 / (kJitterScaleHighThreshold - kJitterScaleLowThreshold)) *
+          (fps - kJitterScaleLowThreshold) * jitterMS;
+    }
+  }
+
+  return static_cast<uint32_t>(jitterMS + 0.5);
+}
+
+bool VCMJitterEstimator::LowRateExperimentEnabled() {
+  if (low_rate_experiment_ == kInit) {
+    std::string group =
+        webrtc::field_trial::FindFullName("WebRTC-ReducedJitterDelay");
+    if (group == "Disabled") {
+      low_rate_experiment_ = kDisabled;
+    } else {
+      low_rate_experiment_ = kEnabled;
+    }
+  }
+  return low_rate_experiment_ == kEnabled ? true : false;
+}
+
+double VCMJitterEstimator::GetFrameRate() const {
+  if (fps_counter_.count() == 0)
+    return 0;
+
+  double fps = 1000000.0 / fps_counter_.ComputeMean();
+  // Sanity check.
+  assert(fps >= 0.0);
+  if (fps > kMaxFramerateEstimate) {
+    fps = kMaxFramerateEstimate;
+  }
+  return fps;
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/jitter_estimator.h b/webrtc/modules/video_coding/jitter_estimator.h
new file mode 100644
index 0000000000..a7b4b3e3df
--- /dev/null
+++ b/webrtc/modules/video_coding/jitter_estimator.h
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+
+#include "webrtc/base/rollingaccumulator.h"
+#include "webrtc/modules/video_coding/rtt_filter.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Clock;
+
+class VCMJitterEstimator {
+ public:
+  VCMJitterEstimator(const Clock* clock,
+                     int32_t vcmId = 0,
+                     int32_t receiverId = 0);
+  virtual ~VCMJitterEstimator();
+  VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
+
+  // Resets the estimate to the initial state
+  void Reset();
+  void ResetNackCount();
+
+  // Updates the jitter estimate with the new data.
+  //
+  // Input:
+  //          - frameDelay      : Delay-delta calculated by UTILDelayEstimate in
+  //          milliseconds
+  //          - frameSize       : Frame size of the current frame.
+  //          - incompleteFrame : Flags if the frame is used to update the
+  //          estimate before it
+  //                              was complete. Default is false.
+  void UpdateEstimate(int64_t frameDelayMS,
+                      uint32_t frameSizeBytes,
+                      bool incompleteFrame = false);
+
+  // Returns the current jitter estimate in milliseconds and adds
+  // also adds an RTT dependent term in cases of retransmission.
+  //  Input:
+  //          - rttMultiplier  : RTT param multiplier (when applicable).
+  //
+  // Return value                   : Jitter estimate in milliseconds
+  int GetJitterEstimate(double rttMultiplier);
+
+  // Updates the nack counter.
+  void FrameNacked();
+
+  // Updates the RTT filter.
+  //
+  // Input:
+  //          - rttMs               : RTT in ms
+  void UpdateRtt(int64_t rttMs);
+
+  void UpdateMaxFrameSize(uint32_t frameSizeBytes);
+
+  // A constant describing the delay from the jitter buffer
+  // to the delay on the receiving side which is not accounted
+  // for by the jitter buffer nor the decoding delay estimate.
+  static const uint32_t OPERATING_SYSTEM_JITTER = 10;
+
+ protected:
+  // These are protected for better testing possibilities
+  double _theta[2];  // Estimated line parameters (slope, offset)
+  double _varNoise;  // Variance of the time-deviation from the line
+
+  virtual bool LowRateExperimentEnabled();
+
+ private:
+  // Updates the Kalman filter for the line describing
+  // the frame size dependent jitter.
+  //
+  // Input:
+  //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in
+  //          milliseconds
+  //          - deltaFSBytes    : Frame size delta, i.e.
+  //                            : frame size at time T minus frame size at time
+  //                            T-1
+  void KalmanEstimateChannel(int64_t frameDelayMS, int32_t deltaFSBytes);
+
+  // Updates the random jitter estimate, i.e. the variance
+  // of the time deviations from the line given by the Kalman filter.
+  //
+  // Input:
+  //          - d_dT              : The deviation from the kalman estimate
+  //          - incompleteFrame   : True if the frame used to update the
+  //          estimate
+  //                                with was incomplete
+  void EstimateRandomJitter(double d_dT, bool incompleteFrame);
+
+  double NoiseThreshold() const;
+
+  // Calculates the current jitter estimate.
+  //
+  // Return value                 : The current jitter estimate in milliseconds
+  double CalculateEstimate();
+
+  // Post process the calculated estimate
+  void PostProcessEstimate();
+
+  // Calculates the difference in delay between a sample and the
+  // expected delay estimated by the Kalman filter.
+  //
+  // Input:
+  //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in
+  //          milliseconds
+  //          - deltaFS         : Frame size delta, i.e. frame size at time
+  //                              T minus frame size at time T-1
+  //
+  // Return value                 : The difference in milliseconds
+  double DeviationFromExpectedDelay(int64_t frameDelayMS,
+                                    int32_t deltaFSBytes) const;
+
+  double GetFrameRate() const;
+
+  // Constants, filter parameters
+  int32_t _vcmId;
+  int32_t _receiverId;
+  const double _phi;
+  const double _psi;
+  const uint32_t _alphaCountMax;
+  const double _thetaLow;
+  const uint32_t _nackLimit;
+  const int32_t _numStdDevDelayOutlier;
+  const int32_t _numStdDevFrameSizeOutlier;
+  const double _noiseStdDevs;
+  const double _noiseStdDevOffset;
+
+  double _thetaCov[2][2];  // Estimate covariance
+  double _Qcov[2][2];      // Process noise covariance
+  double _avgFrameSize;    // Average frame size
+  double _varFrameSize;    // Frame size variance
+  double _maxFrameSize;    // Largest frame size received (descending
+                           // with a factor _psi)
+  uint32_t _fsSum;
+  uint32_t _fsCount;
+
+  int64_t _lastUpdateT;
+  double _prevEstimate;     // The previously returned jitter estimate
+  uint32_t _prevFrameSize;  // Frame size of the previous frame
+  double _avgNoise;         // Average of the random jitter
+  uint32_t _alphaCount;
+  double _filterJitterEstimate;  // The filtered sum of jitter estimates
+
+  uint32_t _startupCount;
+
+  int64_t
+      _latestNackTimestamp;  // Timestamp in ms when the latest nack was seen
+  uint32_t _nackCount;       // Keeps track of the number of nacks received,
+                             // but never goes above _nackLimit
+  VCMRttFilter _rttFilter;
+
+  rtc::RollingAccumulator<uint64_t> fps_counter_;
+  enum ExperimentFlag { kInit, kEnabled, kDisabled };
+  ExperimentFlag low_rate_experiment_;
+  const Clock* clock_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
diff --git a/webrtc/modules/video_coding/main/source/jitter_estimator_tests.cc b/webrtc/modules/video_coding/jitter_estimator_tests.cc
index c69c4bcdad..3d46ce2bcd 100644
--- a/webrtc/modules/video_coding/main/source/jitter_estimator_tests.cc
+++ b/webrtc/modules/video_coding/jitter_estimator_tests.cc
@@ -7,7 +7,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/system_wrappers/include/clock.h"
diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h
deleted file mode 100644
index 67f7b635cb..0000000000
--- a/webrtc/modules/video_coding/main/interface/video_coding.h
+++ /dev/null
@@ -1,544 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
-#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
-
-#if defined(WEBRTC_WIN)
-// This is a workaround on Windows due to the fact that some Windows
-// headers define CreateEvent as a macro to either CreateEventW or CreateEventA.
-// This can cause problems since we use that name as well and could
-// declare them as one thing here whereas in another place a windows header
-// may have been included and then implementing CreateEvent() causes compilation
-// errors.  So for consistency, we include the main windows header here.
-#include <windows.h>
-#endif
-
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/video_frame.h"
-
-namespace webrtc
-{
-
-class Clock;
-class EncodedImageCallback;
-class VideoEncoder;
-class VideoDecoder;
-struct CodecSpecificInfo;
-
-class EventFactory {
- public:
-  virtual ~EventFactory() {}
-
-  virtual EventWrapper* CreateEvent() = 0;
-};
-
-class EventFactoryImpl : public EventFactory {
- public:
-  virtual ~EventFactoryImpl() {}
-
-  virtual EventWrapper* CreateEvent() {
-    return EventWrapper::Create();
-  }
-};
-
-// Used to indicate which decode with errors mode should be used.
-enum VCMDecodeErrorMode {
-  kNoErrors,                // Never decode with errors. Video will freeze
-                            // if nack is disabled.
-  kSelectiveErrors,         // Frames that are determined decodable in
-                            // VCMSessionInfo may be decoded with missing
-                            // packets. As not all incomplete frames will be
-                            // decodable, video will freeze if nack is disabled.
-  kWithErrors               // Release frames as needed. Errors may be
-                            // introduced as some encoded frames may not be
-                            // complete.
-};
-
-class VideoCodingModule : public Module
-{
-public:
-    enum SenderNackMode {
-        kNackNone,
-        kNackAll,
-        kNackSelective
-    };
-
-    enum ReceiverRobustness {
-        kNone,
-        kHardNack,
-        kSoftNack,
-        kReferenceSelection
-    };
-
-    static VideoCodingModule* Create(
-        Clock* clock,
-        VideoEncoderRateObserver* encoder_rate_observer,
-        VCMQMSettingsCallback* qm_settings_callback);
-
-    static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory);
-
-    static void Destroy(VideoCodingModule* module);
-
-    // Get number of supported codecs
-    //
-    // Return value     : Number of supported codecs
-    static uint8_t NumberOfCodecs();
-
-    // Get supported codec settings with using id
-    //
-    // Input:
-    //      - listId         : Id or index of the codec to look up
-    //      - codec          : Memory where the codec settings will be stored
-    //
-    // Return value     : VCM_OK,              on success
-    //                    VCM_PARAMETER_ERROR  if codec not supported or id too high
-    static int32_t Codec(const uint8_t listId, VideoCodec* codec);
-
-    // Get supported codec settings using codec type
-    //
-    // Input:
-    //      - codecType      : The codec type to get settings for
-    //      - codec          : Memory where the codec settings will be stored
-    //
-    // Return value     : VCM_OK,              on success
-    //                    VCM_PARAMETER_ERROR  if codec not supported
-    static int32_t Codec(VideoCodecType codecType, VideoCodec* codec);
-
-    /*
-    *   Sender
-    */
-
-    // Registers a codec to be used for encoding. Calling this
-    // API multiple times overwrites any previously registered codecs.
-    //
-    // NOTE: Must be called on the thread that constructed the VCM instance.
-    //
-    // Input:
-    //      - sendCodec      : Settings for the codec to be registered.
-    //      - numberOfCores  : The number of cores the codec is allowed
-    //                         to use.
-    //      - maxPayloadSize : The maximum size each payload is allowed
-    //                                to have. Usually MTU - overhead.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterSendCodec(const VideoCodec* sendCodec,
-                                            uint32_t numberOfCores,
-                                            uint32_t maxPayloadSize) = 0;
-
-    // Get the current send codec in use.
-    //
-    // If a codec has not been set yet, the |id| property of the return value
-    // will be 0 and |name| empty.
-    //
-    // NOTE: This method intentionally does not hold locks and minimizes data
-    // copying.  It must be called on the thread where the VCM was constructed.
-    virtual const VideoCodec& GetSendCodec() const = 0;
-
-    // DEPRECATED: Use GetSendCodec() instead.
-    //
-    // API to get the current send codec in use.
-    //
-    // Input:
-    //      - currentSendCodec : Address where the sendCodec will be written.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    //
-    // NOTE: The returned codec information is not guaranteed to be current when
-    // the call returns.  This method acquires a lock that is aligned with
-    // video encoding, so it should be assumed to be allowed to block for
-    // several milliseconds.
-    virtual int32_t SendCodec(VideoCodec* currentSendCodec) const = 0;
-
-    // DEPRECATED: Use GetSendCodec() instead.
-    //
-    // API to get the current send codec type
-    //
-    // Return value      : Codec type, on success.
-    //                     kVideoCodecUnknown, on error or if no send codec is set
-    // NOTE: Same notes apply as for SendCodec() above.
-    virtual VideoCodecType SendCodec() const = 0;
-
-    // Register an external encoder object. This can not be used together with
-    // external decoder callbacks.
-    //
-    // Input:
-    //      - externalEncoder : Encoder object to be used for encoding frames inserted
-    //                          with the AddVideoFrame API.
-    //      - payloadType     : The payload type bound which this encoder is bound to.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
-                                            uint8_t payloadType,
-                                            bool internalSource = false) = 0;
-
-    // API to get currently configured encoder target bitrate in bits/s.
-    //
-    // Return value      : 0,   on success.
-    //                     < 0, on error.
-    virtual int Bitrate(unsigned int* bitrate) const = 0;
-
-    // API to get currently configured encoder target frame rate.
-    //
-    // Return value      : 0,   on success.
-    //                     < 0, on error.
-    virtual int FrameRate(unsigned int* framerate) const = 0;
-
-    // Sets the parameters describing the send channel. These parameters are inputs to the
-    // Media Optimization inside the VCM and also specifies the target bit rate for the
-    // encoder. Bit rate used by NACK should already be compensated for by the user.
-    //
-    // Input:
-    //      - target_bitrate        : The target bitrate for VCM in bits/s.
-    //      - lossRate              : Fractions of lost packets the past second.
-    //                                (loss rate in percent = 100 * packetLoss / 255)
-    //      - rtt                   : Current round-trip time in ms.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t SetChannelParameters(uint32_t target_bitrate,
-                                         uint8_t lossRate,
-                                         int64_t rtt) = 0;
-
-    // Sets the parameters describing the receive channel. These parameters are inputs to the
-    // Media Optimization inside the VCM.
-    //
-    // Input:
-    //      - rtt                   : Current round-trip time in ms.
-    //                                with the most amount available bandwidth in a conference
-    //                                scenario
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t SetReceiveChannelParameters(int64_t rtt) = 0;
-
-    // Register a transport callback which will be called to deliver the encoded data and
-    // side information.
-    //
-    // Input:
-    //      - transport  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterTransportCallback(VCMPacketizationCallback* transport) = 0;
-
-    // Register video output information callback which will be called to deliver information
-    // about the video stream produced by the encoder, for instance the average frame rate and
-    // bit rate.
-    //
-    // Input:
-    //      - outputInformation  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterSendStatisticsCallback(
-                                     VCMSendStatisticsCallback* sendStats) = 0;
-
-    // Register a video protection callback which will be called to deliver
-    // the requested FEC rate and NACK status (on/off).
-    //
-    // Input:
-    //      - protection  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterProtectionCallback(VCMProtectionCallback* protection) = 0;
-
-    // Enable or disable a video protection method.
-    //
-    // Input:
-    //      - videoProtection  : The method to enable or disable.
-    //      - enable           : True if the method should be enabled, false if
-    //                           it should be disabled.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t SetVideoProtection(VCMVideoProtection videoProtection,
-                                       bool enable) = 0;
-
-    // Add one raw video frame to the encoder. This function does all the necessary
-    // processing, then decides what frame type to encode, or if the frame should be
-    // dropped. If the frame should be encoded it passes the frame to the encoder
-    // before it returns.
-    //
-    // Input:
-    //      - videoFrame        : Video frame to encode.
-    //      - codecSpecificInfo : Extra codec information, e.g., pre-parsed in-band signaling.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t AddVideoFrame(
-        const VideoFrame& videoFrame,
-        const VideoContentMetrics* contentMetrics = NULL,
-        const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
-
-    // Next frame encoded should be an intra frame (keyframe).
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t IntraFrameRequest(int stream_index) = 0;
-
-    // Frame Dropper enable. Can be used to disable the frame dropping when the encoder
-    // over-uses its bit rate. This API is designed to be used when the encoded frames
-    // are supposed to be stored to an AVI file, or when the I420 codec is used and the
-    // target bit rate shouldn't affect the frame rate.
-    //
-    // Input:
-    //      - enable            : True to enable the setting, false to disable it.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t EnableFrameDropper(bool enable) = 0;
-
-
-    /*
-    *   Receiver
-    */
-
-    // Register possible receive codecs, can be called multiple times for different codecs.
-    // The module will automatically switch between registered codecs depending on the
-    // payload type of incoming frames. The actual decoder will be created when needed.
-    //
-    // Input:
-    //      - receiveCodec      : Settings for the codec to be registered.
-    //      - numberOfCores     : Number of CPU cores that the decoder is allowed to use.
-    //      - requireKeyFrame   : Set this to true if you don't want any delta frames
-    //                            to be decoded until the first key frame has been decoded.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
-                                         int32_t numberOfCores,
-                                         bool requireKeyFrame = false) = 0;
-
-    // Register an externally defined decoder/renderer object. Can be a decoder only or a
-    // decoder coupled with a renderer. Note that RegisterReceiveCodec must be called to
-    // be used for decoding incoming streams.
-    //
-    // Input:
-    //      - externalDecoder        : The external decoder/renderer object.
-    //      - payloadType            : The payload type which this decoder should be
-    //                                 registered to.
-    //      - internalRenderTiming   : True if the internal renderer (if any) of the decoder
-    //                                 object can make sure to render at a given time in ms.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                            uint8_t payloadType,
-                                            bool internalRenderTiming) = 0;
-
-    // Register a receive callback. Will be called whenever there is a new frame ready
-    // for rendering.
-    //
-    // Input:
-    //      - receiveCallback        : The callback object to be used by the module when a
-    //                                 frame is ready for rendering.
-    //                                 De-register with a NULL pointer.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterReceiveCallback(VCMReceiveCallback* receiveCallback) = 0;
-
-    // Register a receive statistics callback which will be called to deliver information
-    // about the video stream received by the receiving side of the VCM, for instance the
-    // average frame rate and bit rate.
-    //
-    // Input:
-    //      - receiveStats  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterReceiveStatisticsCallback(
-                               VCMReceiveStatisticsCallback* receiveStats) = 0;
-
-    // Register a decoder timing callback which will be called to deliver
-    // information about the timing of the decoder in the receiving side of the
-    // VCM, for instance the current and maximum frame decode latency.
-    //
-    // Input:
-    //      - decoderTiming  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterDecoderTimingCallback(
-        VCMDecoderTimingCallback* decoderTiming) = 0;
-
-    // Register a frame type request callback. This callback will be called when the
-    // module needs to request specific frame types from the send side.
-    //
-    // Input:
-    //      - frameTypeCallback      : The callback object to be used by the module when
-    //                                 requesting a specific type of frame from the send side.
-    //                                 De-register with a NULL pointer.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t RegisterFrameTypeCallback(
-                                  VCMFrameTypeCallback* frameTypeCallback) = 0;
-
-    // Registers a callback which is called whenever the receive side of the VCM
-    // encounters holes in the packet sequence and needs packets to be retransmitted.
-    //
-    // Input:
-    //              - callback      : The callback to be registered in the VCM.
-    //
-    // Return value     : VCM_OK,     on success.
-    //                    <0,         on error.
-    virtual int32_t RegisterPacketRequestCallback(
-                                        VCMPacketRequestCallback* callback) = 0;
-
-    // Waits for the next frame in the jitter buffer to become complete
-    // (waits no longer than maxWaitTimeMs), then passes it to the decoder for decoding.
-    // Should be called as often as possible to get the most out of the decoder.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t Decode(uint16_t maxWaitTimeMs = 200) = 0;
-
-    // Registers a callback which conveys the size of the render buffer.
-    virtual int RegisterRenderBufferSizeCallback(
-        VCMRenderBufferSizeCallback* callback) = 0;
-
-    // Reset the decoder state to the initial state.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t ResetDecoder() = 0;
-
-    // API to get the codec which is currently used for decoding by the module.
-    //
-    // Input:
-    //      - currentReceiveCodec      : Settings for the codec to be registered.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
-
-    // API to get the codec type currently used for decoding by the module.
-    //
-    // Return value      : codecy type,            on success.
-    //                     kVideoCodecUnknown, on error or if no receive codec is registered
-    virtual VideoCodecType ReceiveCodec() const = 0;
-
-    // Insert a parsed packet into the receiver side of the module. Will be placed in the
-    // jitter buffer waiting for the frame to become complete. Returns as soon as the packet
-    // has been placed in the jitter buffer.
-    //
-    // Input:
-    //      - incomingPayload      : Payload of the packet.
-    //      - payloadLength        : Length of the payload.
-    //      - rtpInfo              : The parsed header.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t IncomingPacket(const uint8_t* incomingPayload,
-                                   size_t payloadLength,
-                                   const WebRtcRTPHeader& rtpInfo) = 0;
-
-    // Minimum playout delay (Used for lip-sync). This is the minimum delay required
-    // to sync with audio. Not included in  VideoCodingModule::Delay()
-    // Defaults to 0 ms.
-    //
-    // Input:
-    //      - minPlayoutDelayMs   : Additional delay in ms.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) = 0;
-
-    // Set the time required by the renderer to render a frame.
-    //
-    // Input:
-    //      - timeMS        : The time in ms required by the renderer to render a frame.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,    on error.
-    virtual int32_t SetRenderDelay(uint32_t timeMS) = 0;
-
-    // The total delay desired by the VCM. Can be less than the minimum
-    // delay set with SetMinimumPlayoutDelay.
-    //
-    // Return value      : Total delay in ms, on success.
-    //                     < 0,               on error.
-    virtual int32_t Delay() const = 0;
-
-    // Returns the number of packets discarded by the jitter buffer due to being
-    // too late. This can include duplicated packets which arrived after the
-    // frame was sent to the decoder. Therefore packets which were prematurely
-    // NACKed will be counted.
-    virtual uint32_t DiscardedPackets() const = 0;
-
-
-    // Robustness APIs
-
-    // Set the receiver robustness mode. The mode decides how the receiver
-    // responds to losses in the stream. The type of counter-measure (soft or
-    // hard NACK, dual decoder, RPS, etc.) is selected through the
-    // robustnessMode parameter. The errorMode parameter decides if it is
-    // allowed to display frames corrupted by losses. Note that not all
-    // combinations of the two parameters are feasible. An error will be
-    // returned for invalid combinations.
-    // Input:
-    //      - robustnessMode : selected robustness mode.
-    //      - errorMode      : selected error mode.
-    //
-    // Return value      : VCM_OK, on success;
-    //                     < 0, on error.
-    virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
-                                          VCMDecodeErrorMode errorMode) = 0;
-
-    // Set the decode error mode. The mode decides which errors (if any) are
-    // allowed in decodable frames. Note that setting decode_error_mode to
-    // anything other than kWithErrors without enabling nack will cause
-    // long-term freezes (resulting from frequent key frame requests) if
-    // packet loss occurs.
-    virtual void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) = 0;
-
-    // Sets the maximum number of sequence numbers that we are allowed to NACK
-    // and the oldest sequence number that we will consider to NACK. If a
-    // sequence number older than |max_packet_age_to_nack| is missing
-    // a key frame will be requested. A key frame will also be requested if the
-    // time of incomplete or non-continuous frames in the jitter buffer is above
-    // |max_incomplete_time_ms|.
-    virtual void SetNackSettings(size_t max_nack_list_size,
-                                 int max_packet_age_to_nack,
-                                 int max_incomplete_time_ms) = 0;
-
-    // Setting a desired delay to the VCM receiver. Video rendering will be
-    // delayed by at least desired_delay_ms.
-    virtual int SetMinReceiverDelay(int desired_delay_ms) = 0;
-
-    // Lets the sender suspend video when the rate drops below
-    // |threshold_bps|, and turns back on when the rate goes back up above
-    // |threshold_bps| + |window_bps|.
-    virtual void SuspendBelowMinBitrate() = 0;
-
-    // Returns true if SuspendBelowMinBitrate is engaged and the video has been
-    // suspended due to bandwidth limitations; otherwise false.
-    virtual bool VideoSuspended() const = 0;
-
-    virtual void RegisterPreDecodeImageCallback(
-        EncodedImageCallback* observer) = 0;
-    virtual void RegisterPostEncodeImageCallback(
-        EncodedImageCallback* post_encode_callback) = 0;
-    // Releases pending decode calls, permitting faster thread shutdown.
-    virtual void TriggerDecoderShutdown() = 0;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
diff --git a/webrtc/modules/video_coding/main/source/OWNERS b/webrtc/modules/video_coding/main/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/webrtc/modules/video_coding/main/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/webrtc/modules/video_coding/main/source/codec_timer.cc b/webrtc/modules/video_coding/main/source/codec_timer.cc
deleted file mode 100644
index a462258813..0000000000
--- a/webrtc/modules/video_coding/main/source/codec_timer.cc
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/codec_timer.h"
-
-#include <assert.h>
-
-namespace webrtc
-{
-
-// The first kIgnoredSampleCount samples will be ignored.
-static const int32_t kIgnoredSampleCount = 5;
-
-VCMCodecTimer::VCMCodecTimer()
-:
-_filteredMax(0),
-_ignoredSampleCount(0),
-_shortMax(0),
-_history()
-{
-    Reset();
-}
-
-int32_t VCMCodecTimer::StopTimer(int64_t startTimeMs, int64_t nowMs)
-{
-    const int32_t timeDiff = static_cast<int32_t>(nowMs - startTimeMs);
-    MaxFilter(timeDiff, nowMs);
-    return timeDiff;
-}
-
-void VCMCodecTimer::Reset()
-{
-    _filteredMax = 0;
-    _ignoredSampleCount = 0;
-    _shortMax = 0;
-    for (int i=0; i < MAX_HISTORY_SIZE; i++)
-    {
-        _history[i].shortMax = 0;
-        _history[i].timeMs = -1;
-    }
-}
-
-// Update the max-value filter
-void VCMCodecTimer::MaxFilter(int32_t decodeTime, int64_t nowMs)
-{
-    if (_ignoredSampleCount >= kIgnoredSampleCount)
-    {
-        UpdateMaxHistory(decodeTime, nowMs);
-        ProcessHistory(nowMs);
-    }
-    else
-    {
-        _ignoredSampleCount++;
-    }
-}
-
-void
-VCMCodecTimer::UpdateMaxHistory(int32_t decodeTime, int64_t now)
-{
-    if (_history[0].timeMs >= 0 &&
-        now - _history[0].timeMs < SHORT_FILTER_MS)
-    {
-        if (decodeTime > _shortMax)
-        {
-            _shortMax = decodeTime;
-        }
-    }
-    else
-    {
-        // Only add a new value to the history once a second
-        if(_history[0].timeMs == -1)
-        {
-            // First, no shift
-            _shortMax = decodeTime;
-        }
-        else
-        {
-            // Shift
-            for(int i = (MAX_HISTORY_SIZE - 2); i >= 0 ; i--)
-            {
-                _history[i+1].shortMax = _history[i].shortMax;
-                _history[i+1].timeMs = _history[i].timeMs;
-            }
-        }
-        if (_shortMax == 0)
-        {
-            _shortMax = decodeTime;
-        }
-
-        _history[0].shortMax = _shortMax;
-        _history[0].timeMs = now;
-        _shortMax = 0;
-    }
-}
-
-void
-VCMCodecTimer::ProcessHistory(int64_t nowMs)
-{
-    _filteredMax = _shortMax;
-    if (_history[0].timeMs == -1)
-    {
-        return;
-    }
-    for (int i=0; i < MAX_HISTORY_SIZE; i++)
-    {
-        if (_history[i].timeMs == -1)
-        {
-            break;
-        }
-        if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS)
-        {
-            // This sample (and all samples after this) is too old
-            break;
-        }
-        if (_history[i].shortMax > _filteredMax)
-        {
-            // This sample is the largest one this far into the history
-            _filteredMax = _history[i].shortMax;
-        }
-    }
-}
-
-// Get the maximum observed time within a time window
-int32_t VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const
-{
-    return _filteredMax;
-}
-
-}
diff --git a/webrtc/modules/video_coding/main/source/codec_timer.h b/webrtc/modules/video_coding/main/source/codec_timer.h
deleted file mode 100644
index 9268e8d817..0000000000
--- a/webrtc/modules/video_coding/main/source/codec_timer.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
-#define MAX_HISTORY_SIZE 10
-#define SHORT_FILTER_MS 1000
-
-class VCMShortMaxSample
-{
-public:
-    VCMShortMaxSample() : shortMax(0), timeMs(-1) {};
-
-    int32_t     shortMax;
-    int64_t     timeMs;
-};
-
-class VCMCodecTimer
-{
-public:
-    VCMCodecTimer();
-
-    // Updates and returns the max filtered decode time.
-    int32_t StopTimer(int64_t startTimeMs, int64_t nowMs);
-
-    // Empty the list of timers.
-    void Reset();
-
-    // Get the required decode time in ms.
-    int32_t RequiredDecodeTimeMs(FrameType frameType) const;
-
-private:
-    void UpdateMaxHistory(int32_t decodeTime, int64_t now);
-    void MaxFilter(int32_t newTime, int64_t nowMs);
-    void ProcessHistory(int64_t nowMs);
-
-    int32_t                     _filteredMax;
-    // The number of samples ignored so far.
-    int32_t                     _ignoredSampleCount;
-    int32_t                     _shortMax;
-    VCMShortMaxSample           _history[MAX_HISTORY_SIZE];
-
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.h b/webrtc/modules/video_coding/main/source/encoded_frame.h
deleted file mode 100644
index 608578c35d..0000000000
--- a/webrtc/modules/video_coding/main/source/encoded_frame.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
-#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
-
-#include <vector>
-
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/video_image.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-
-namespace webrtc
-{
-
-class VCMEncodedFrame : protected EncodedImage
-{
-public:
-    VCMEncodedFrame();
-    VCMEncodedFrame(const webrtc::EncodedImage& rhs);
-    VCMEncodedFrame(const VCMEncodedFrame& rhs);
-
-    ~VCMEncodedFrame();
-    /**
-    *   Delete VideoFrame and resets members to zero
-    */
-    void Free();
-    /**
-    *   Set render time in milliseconds
-    */
-    void SetRenderTime(const int64_t renderTimeMs) {_renderTimeMs = renderTimeMs;}
-
-    /**
-    *   Set the encoded frame size
-    */
-    void SetEncodedSize(uint32_t width, uint32_t height)
-                       { _encodedWidth  = width; _encodedHeight = height; }
-    /**
-    *   Get the encoded image
-    */
-    const webrtc::EncodedImage& EncodedImage() const
-                       { return static_cast<const webrtc::EncodedImage&>(*this); }
-    /**
-    *   Get pointer to frame buffer
-    */
-    const uint8_t* Buffer() const {return _buffer;}
-    /**
-    *   Get frame length
-    */
-    size_t Length() const {return _length;}
-    /**
-    *   Get frame timestamp (90kHz)
-    */
-    uint32_t TimeStamp() const {return _timeStamp;}
-    /**
-    *   Get render time in milliseconds
-    */
-    int64_t RenderTimeMs() const {return _renderTimeMs;}
-    /**
-    *   Get frame type
-    */
-    webrtc::FrameType FrameType() const { return _frameType; }
-    /**
-    *   Get frame rotation
-    */
-    VideoRotation rotation() const { return _rotation; }
-    /**
-    *   True if this frame is complete, false otherwise
-    */
-    bool Complete() const { return _completeFrame; }
-    /**
-    *   True if there's a frame missing before this frame
-    */
-    bool MissingFrame() const { return _missingFrame; }
-    /**
-    *   Payload type of the encoded payload
-    */
-    uint8_t PayloadType() const { return _payloadType; }
-    /**
-    *   Get codec specific info.
-    *   The returned pointer is only valid as long as the VCMEncodedFrame
-    *   is valid. Also, VCMEncodedFrame owns the pointer and will delete
-    *   the object.
-    */
-    const CodecSpecificInfo* CodecSpecific() const {return &_codecSpecificInfo;}
-
-    const RTPFragmentationHeader* FragmentationHeader() const;
-
-protected:
-    /**
-    * Verifies that current allocated buffer size is larger than or equal to the input size.
-    * If the current buffer size is smaller, a new allocation is made and the old buffer data
-    * is copied to the new buffer.
-    * Buffer size is updated to minimumSize.
-    */
-    void VerifyAndAllocate(size_t minimumSize);
-
-    void Reset();
-
-    void CopyCodecSpecific(const RTPVideoHeader* header);
-
-    int64_t                 _renderTimeMs;
-    uint8_t                 _payloadType;
-    bool                          _missingFrame;
-    CodecSpecificInfo             _codecSpecificInfo;
-    webrtc::VideoCodecType        _codec;
-    RTPFragmentationHeader        _fragmentation;
-    VideoRotation                 _rotation;
-
-    // Video rotation is only set along with the last packet for each frame
-    // (same as marker bit). This |_rotation_set| is only for debugging purpose
-    // to ensure we don't set it twice for a frame.
-    bool                          _rotation_set;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
diff --git a/webrtc/modules/video_coding/main/source/fec_tables_xor.h b/webrtc/modules/video_coding/main/source/fec_tables_xor.h
deleted file mode 100644
index 28c67b4565..0000000000
--- a/webrtc/modules/video_coding/main/source/fec_tables_xor.h
+++ /dev/null
@@ -1,6481 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
-
-// This is a private header for media_opt_util.cc.
-// It should not be included by other files.
-
-namespace webrtc {
-
-// Table for Protection factor (code rate) of delta frames, for the XOR FEC.
-// Input is the packet loss and an effective rate (bits/frame).
-// Output is array kCodeRateXORTable[k], where k = rate_i*129 + loss_j;
-// loss_j = 0,1,..128, and rate_i varies over some range.
-static const int kSizeCodeRateXORTable = 6450;
-static const unsigned char kCodeRateXORTable[kSizeCodeRateXORTable] = {
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-
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
diff --git a/webrtc/modules/video_coding/main/source/frame_buffer.cc b/webrtc/modules/video_coding/main/source/frame_buffer.cc
deleted file mode 100644
index 5b6680ec61..0000000000
--- a/webrtc/modules/video_coding/main/source/frame_buffer.cc
+++ /dev/null
@@ -1,297 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-
-#include <assert.h>
-#include <string.h>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/system_wrappers/include/logging.h"
-
-namespace webrtc {
-
-VCMFrameBuffer::VCMFrameBuffer()
-  :
-    _state(kStateEmpty),
-    _nackCount(0),
-    _latestPacketTimeMs(-1) {
-}
-
-VCMFrameBuffer::~VCMFrameBuffer() {
-}
-
-VCMFrameBuffer::VCMFrameBuffer(const VCMFrameBuffer& rhs)
-:
-VCMEncodedFrame(rhs),
-_state(rhs._state),
-_sessionInfo(),
-_nackCount(rhs._nackCount),
-_latestPacketTimeMs(rhs._latestPacketTimeMs) {
-    _sessionInfo = rhs._sessionInfo;
-    _sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
-}
-
-webrtc::FrameType
-VCMFrameBuffer::FrameType() const {
-    return _sessionInfo.FrameType();
-}
-
-int32_t
-VCMFrameBuffer::GetLowSeqNum() const {
-    return _sessionInfo.LowSequenceNumber();
-}
-
-int32_t
-VCMFrameBuffer::GetHighSeqNum() const {
-    return _sessionInfo.HighSequenceNumber();
-}
-
-int VCMFrameBuffer::PictureId() const {
-  return _sessionInfo.PictureId();
-}
-
-int VCMFrameBuffer::TemporalId() const {
-  return _sessionInfo.TemporalId();
-}
-
-bool VCMFrameBuffer::LayerSync() const {
-  return _sessionInfo.LayerSync();
-}
-
-int VCMFrameBuffer::Tl0PicId() const {
-  return _sessionInfo.Tl0PicId();
-}
-
-bool VCMFrameBuffer::NonReference() const {
-  return _sessionInfo.NonReference();
-}
-
-void VCMFrameBuffer::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
-  _sessionInfo.SetGofInfo(gof_info, idx);
-  // TODO(asapersson): Consider adding hdr->VP9.ref_picture_id for testing.
-  _codecSpecificInfo.codecSpecific.VP9.temporal_idx =
-      gof_info.temporal_idx[idx];
-  _codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
-      gof_info.temporal_up_switch[idx];
-}
-
-bool
-VCMFrameBuffer::IsSessionComplete() const {
-    return _sessionInfo.complete();
-}
-
-// Insert packet
-VCMFrameBufferEnum
-VCMFrameBuffer::InsertPacket(const VCMPacket& packet,
-                             int64_t timeInMs,
-                             VCMDecodeErrorMode decode_error_mode,
-                             const FrameData& frame_data) {
-    assert(!(NULL == packet.dataPtr && packet.sizeBytes > 0));
-    if (packet.dataPtr != NULL) {
-        _payloadType = packet.payloadType;
-    }
-
-    if (kStateEmpty == _state) {
-        // First packet (empty and/or media) inserted into this frame.
-        // store some info and set some initial values.
-        _timeStamp = packet.timestamp;
-        // We only take the ntp timestamp of the first packet of a frame.
-        ntp_time_ms_ = packet.ntp_time_ms_;
-        _codec = packet.codec;
-        if (packet.frameType != kEmptyFrame) {
-            // first media packet
-            SetState(kStateIncomplete);
-        }
-    }
-
-    uint32_t requiredSizeBytes = Length() + packet.sizeBytes +
-                   (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
-    if (requiredSizeBytes >= _size) {
-        const uint8_t* prevBuffer = _buffer;
-        const uint32_t increments = requiredSizeBytes /
-                                          kBufferIncStepSizeBytes +
-                                        (requiredSizeBytes %
-                                         kBufferIncStepSizeBytes > 0);
-        const uint32_t newSize = _size +
-                                       increments * kBufferIncStepSizeBytes;
-        if (newSize > kMaxJBFrameSizeBytes) {
-            LOG(LS_ERROR) << "Failed to insert packet due to frame being too "
-                             "big.";
-            return kSizeError;
-        }
-        VerifyAndAllocate(newSize);
-        _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
-    }
-
-    if (packet.width > 0 && packet.height > 0) {
-      _encodedWidth = packet.width;
-      _encodedHeight = packet.height;
-    }
-
-    // Don't copy payload specific data for empty packets (e.g padding packets).
-    if (packet.sizeBytes > 0)
-      CopyCodecSpecific(&packet.codecSpecificHeader);
-
-    int retVal = _sessionInfo.InsertPacket(packet, _buffer,
-                                           decode_error_mode,
-                                           frame_data);
-    if (retVal == -1) {
-        return kSizeError;
-    } else if (retVal == -2) {
-        return kDuplicatePacket;
-    } else if (retVal == -3) {
-        return kOutOfBoundsPacket;
-    }
-    // update length
-    _length = Length() + static_cast<uint32_t>(retVal);
-
-    _latestPacketTimeMs = timeInMs;
-
-    // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
-    // ts_126114v120700p.pdf Section 7.4.5.
-    // The MTSI client shall add the payload bytes as defined in this clause
-    // onto the last RTP packet in each group of packets which make up a key
-    // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
-    // (HEVC)).
-    if (packet.markerBit) {
-      RTC_DCHECK(!_rotation_set);
-      _rotation = packet.codecSpecificHeader.rotation;
-      _rotation_set = true;
-    }
-
-    if (_sessionInfo.complete()) {
-      SetState(kStateComplete);
-      return kCompleteSession;
-    } else if (_sessionInfo.decodable()) {
-      SetState(kStateDecodable);
-      return kDecodableSession;
-    }
-    return kIncomplete;
-}
-
-int64_t
-VCMFrameBuffer::LatestPacketTimeMs() const {
-    return _latestPacketTimeMs;
-}
-
-void
-VCMFrameBuffer::IncrementNackCount() {
-    _nackCount++;
-}
-
-int16_t
-VCMFrameBuffer::GetNackCount() const {
-    return _nackCount;
-}
-
-bool
-VCMFrameBuffer::HaveFirstPacket() const {
-    return _sessionInfo.HaveFirstPacket();
-}
-
-bool
-VCMFrameBuffer::HaveLastPacket() const {
-    return _sessionInfo.HaveLastPacket();
-}
-
-int
-VCMFrameBuffer::NumPackets() const {
-    return _sessionInfo.NumPackets();
-}
-
-void
-VCMFrameBuffer::Reset() {
-    _length = 0;
-    _timeStamp = 0;
-    _sessionInfo.Reset();
-    _payloadType = 0;
-    _nackCount = 0;
-    _latestPacketTimeMs = -1;
-    _state = kStateEmpty;
-    VCMEncodedFrame::Reset();
-}
-
-// Set state of frame
-void
-VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state) {
-    if (_state == state) {
-        return;
-    }
-    switch (state) {
-    case kStateIncomplete:
-        // we can go to this state from state kStateEmpty
-        assert(_state == kStateEmpty);
-
-        // Do nothing, we received a packet
-        break;
-
-    case kStateComplete:
-        assert(_state == kStateEmpty ||
-               _state == kStateIncomplete ||
-               _state == kStateDecodable);
-
-        break;
-
-    case kStateEmpty:
-        // Should only be set to empty through Reset().
-        assert(false);
-        break;
-
-    case kStateDecodable:
-        assert(_state == kStateEmpty ||
-               _state == kStateIncomplete);
-        break;
-    }
-    _state = state;
-}
-
-// Get current state of frame
-VCMFrameBufferStateEnum
-VCMFrameBuffer::GetState() const {
-    return _state;
-}
-
-// Get current state of frame
-VCMFrameBufferStateEnum
-VCMFrameBuffer::GetState(uint32_t& timeStamp) const {
-    timeStamp = TimeStamp();
-    return GetState();
-}
-
-bool
-VCMFrameBuffer::IsRetransmitted() const {
-    return _sessionInfo.session_nack();
-}
-
-void
-VCMFrameBuffer::PrepareForDecode(bool continuous) {
-#ifdef INDEPENDENT_PARTITIONS
-    if (_codec == kVideoCodecVP8) {
-        _length =
-            _sessionInfo.BuildVP8FragmentationHeader(_buffer, _length,
-                                                     &_fragmentation);
-    } else {
-        size_t bytes_removed = _sessionInfo.MakeDecodable();
-        _length -= bytes_removed;
-    }
-#else
-    size_t bytes_removed = _sessionInfo.MakeDecodable();
-    _length -= bytes_removed;
-#endif
-    // Transfer frame information to EncodedFrame and create any codec
-    // specific information.
-    _frameType = _sessionInfo.FrameType();
-    _completeFrame = _sessionInfo.complete();
-    _missingFrame = !continuous;
-}
-
-}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/generic_decoder.cc b/webrtc/modules/video_coding/main/source/generic_decoder.cc
deleted file mode 100644
index 8b2d3974de..0000000000
--- a/webrtc/modules/video_coding/main/source/generic_decoder.cc
+++ /dev/null
@@ -1,198 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/generic_decoder.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/logging.h"
-
-namespace webrtc {
-
-VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing,
-                                                 Clock* clock)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_clock(clock),
-_receiveCallback(NULL),
-_timing(timing),
-_timestampMap(kDecoderFrameMemoryLength),
-_lastReceivedPictureID(0)
-{
-}
-
-VCMDecodedFrameCallback::~VCMDecodedFrameCallback()
-{
-    delete _critSect;
-}
-
-void VCMDecodedFrameCallback::SetUserReceiveCallback(
-    VCMReceiveCallback* receiveCallback)
-{
-    CriticalSectionScoped cs(_critSect);
-    _receiveCallback = receiveCallback;
-}
-
-VCMReceiveCallback* VCMDecodedFrameCallback::UserReceiveCallback()
-{
-    CriticalSectionScoped cs(_critSect);
-    return _receiveCallback;
-}
-
-int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage) {
-    // TODO(holmer): We should improve this so that we can handle multiple
-    // callbacks from one call to Decode().
-    VCMFrameInformation* frameInfo;
-    VCMReceiveCallback* callback;
-    {
-        CriticalSectionScoped cs(_critSect);
-        frameInfo = _timestampMap.Pop(decodedImage.timestamp());
-        callback = _receiveCallback;
-    }
-
-    if (frameInfo == NULL) {
-      LOG(LS_WARNING) << "Too many frames backed up in the decoder, dropping "
-                         "this one.";
-      return WEBRTC_VIDEO_CODEC_OK;
-    }
-
-    _timing.StopDecodeTimer(
-        decodedImage.timestamp(),
-        frameInfo->decodeStartTimeMs,
-        _clock->TimeInMilliseconds(),
-        frameInfo->renderTimeMs);
-
-    if (callback != NULL)
-    {
-        decodedImage.set_render_time_ms(frameInfo->renderTimeMs);
-        decodedImage.set_rotation(frameInfo->rotation);
-        callback->FrameToRender(decodedImage);
-    }
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t
-VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(
-    const uint64_t pictureId)
-{
-    CriticalSectionScoped cs(_critSect);
-    if (_receiveCallback != NULL)
-    {
-        return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
-    }
-    return -1;
-}
-
-int32_t
-VCMDecodedFrameCallback::ReceivedDecodedFrame(const uint64_t pictureId)
-{
-    _lastReceivedPictureID = pictureId;
-    return 0;
-}
-
-uint64_t VCMDecodedFrameCallback::LastReceivedPictureID() const
-{
-    return _lastReceivedPictureID;
-}
-
-void VCMDecodedFrameCallback::Map(uint32_t timestamp,
-                                  VCMFrameInformation* frameInfo) {
-  CriticalSectionScoped cs(_critSect);
-  _timestampMap.Add(timestamp, frameInfo);
-}
-
-int32_t VCMDecodedFrameCallback::Pop(uint32_t timestamp)
-{
-    CriticalSectionScoped cs(_critSect);
-    if (_timestampMap.Pop(timestamp) == NULL)
-    {
-        return VCM_GENERAL_ERROR;
-    }
-    return VCM_OK;
-}
-
-VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, bool isExternal)
-:
-_callback(NULL),
-_frameInfos(),
-_nextFrameInfoIdx(0),
-_decoder(decoder),
-_codecType(kVideoCodecUnknown),
-_isExternal(isExternal),
-_keyFrameDecoded(false)
-{
-}
-
-VCMGenericDecoder::~VCMGenericDecoder()
-{
-}
-
-int32_t VCMGenericDecoder::InitDecode(const VideoCodec* settings,
-                                      int32_t numberOfCores)
-{
-    _codecType = settings->codecType;
-
-    return _decoder.InitDecode(settings, numberOfCores);
-}
-
-int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame,
-                                        int64_t nowMs)
-{
-    _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
-    _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
-    _frameInfos[_nextFrameInfoIdx].rotation = frame.rotation();
-    _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
-
-    _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
-    int32_t ret = _decoder.Decode(frame.EncodedImage(),
-                                        frame.MissingFrame(),
-                                        frame.FragmentationHeader(),
-                                        frame.CodecSpecific(),
-                                        frame.RenderTimeMs());
-
-    if (ret < WEBRTC_VIDEO_CODEC_OK)
-    {
-        LOG(LS_WARNING) << "Failed to decode frame with timestamp "
-                        << frame.TimeStamp() << ", error code: " << ret;
-        _callback->Pop(frame.TimeStamp());
-        return ret;
-    }
-    else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
-             ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
-    {
-        // No output
-        _callback->Pop(frame.TimeStamp());
-    }
-    return ret;
-}
-
-int32_t
-VCMGenericDecoder::Release()
-{
-    return _decoder.Release();
-}
-
-int32_t VCMGenericDecoder::Reset()
-{
-    return _decoder.Reset();
-}
-
-int32_t VCMGenericDecoder::RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback)
-{
-    _callback = callback;
-    return _decoder.RegisterDecodeCompleteCallback(callback);
-}
-
-bool VCMGenericDecoder::External() const
-{
-    return _isExternal;
-}
-
-}  // namespace
diff --git a/webrtc/modules/video_coding/main/source/generic_decoder.h b/webrtc/modules/video_coding/main/source/generic_decoder.h
deleted file mode 100644
index 09929e64f4..0000000000
--- a/webrtc/modules/video_coding/main/source/generic_decoder.h
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/timestamp_map.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-
-namespace webrtc
-{
-
-class VCMReceiveCallback;
-
-enum { kDecoderFrameMemoryLength = 10 };
-
-struct VCMFrameInformation
-{
-    int64_t     renderTimeMs;
-    int64_t     decodeStartTimeMs;
-    void*             userData;
-    VideoRotation rotation;
-};
-
-class VCMDecodedFrameCallback : public DecodedImageCallback
-{
-public:
-    VCMDecodedFrameCallback(VCMTiming& timing, Clock* clock);
-    virtual ~VCMDecodedFrameCallback();
-    void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
-    VCMReceiveCallback* UserReceiveCallback();
-
-    virtual int32_t Decoded(VideoFrame& decodedImage);
-    virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);
-    virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId);
-
-    uint64_t LastReceivedPictureID() const;
-
-    void Map(uint32_t timestamp, VCMFrameInformation* frameInfo);
-    int32_t Pop(uint32_t timestamp);
-
-private:
-    // Protect |_receiveCallback| and |_timestampMap|.
-    CriticalSectionWrapper* _critSect;
-    Clock* _clock;
-    VCMReceiveCallback* _receiveCallback;  // Guarded by |_critSect|.
-    VCMTiming& _timing;
-    VCMTimestampMap _timestampMap;  // Guarded by |_critSect|.
-    uint64_t _lastReceivedPictureID;
-};
-
-
-class VCMGenericDecoder
-{
-    friend class VCMCodecDataBase;
-public:
-    VCMGenericDecoder(VideoDecoder& decoder, bool isExternal = false);
-    ~VCMGenericDecoder();
-
-    /**
-    *	Initialize the decoder with the information from the VideoCodec
-    */
-    int32_t InitDecode(const VideoCodec* settings,
-                             int32_t numberOfCores);
-
-    /**
-    *	Decode to a raw I420 frame,
-    *
-    *	inputVideoBuffer	reference to encoded video frame
-    */
-    int32_t Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
-
-    /**
-    *	Free the decoder memory
-    */
-    int32_t Release();
-
-    /**
-    *	Reset the decoder state, prepare for a new call
-    */
-    int32_t Reset();
-
-    /**
-    * Set decode callback. Deregistering while decoding is illegal.
-    */
-    int32_t RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
-
-    bool External() const;
-
-private:
-    VCMDecodedFrameCallback*    _callback;
-    VCMFrameInformation         _frameInfos[kDecoderFrameMemoryLength];
-    uint32_t                    _nextFrameInfoIdx;
-    VideoDecoder&               _decoder;
-    VideoCodecType              _codecType;
-    bool                        _isExternal;
-    bool                        _keyFrameDecoded;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
diff --git a/webrtc/modules/video_coding/main/source/generic_encoder.h b/webrtc/modules/video_coding/main/source/generic_encoder.h
deleted file mode 100644
index 3a7132860f..0000000000
--- a/webrtc/modules/video_coding/main/source/generic_encoder.h
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
-
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-
-#include <stdio.h>
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-namespace media_optimization {
-class MediaOptimization;
-}  // namespace media_optimization
-
-struct EncoderParameters {
-  uint32_t target_bitrate;
-  uint8_t loss_rate;
-  int64_t rtt;
-  uint32_t input_frame_rate;
-};
-
-/*************************************/
-/* VCMEncodeFrameCallback class     */
-/***********************************/
-class VCMEncodedFrameCallback : public EncodedImageCallback
-{
-public:
-    VCMEncodedFrameCallback(EncodedImageCallback* post_encode_callback);
-    virtual ~VCMEncodedFrameCallback();
-
-    /*
-    * Callback implementation - codec encode complete
-    */
-    int32_t Encoded(
-        const EncodedImage& encodedImage,
-        const CodecSpecificInfo* codecSpecificInfo = NULL,
-        const RTPFragmentationHeader* fragmentationHeader = NULL);
-    /*
-    * Callback implementation - generic encoder encode complete
-    */
-    int32_t SetTransportCallback(VCMPacketizationCallback* transport);
-    /**
-    * Set media Optimization
-    */
-    void SetMediaOpt (media_optimization::MediaOptimization* mediaOpt);
-
-    void SetPayloadType(uint8_t payloadType) { _payloadType = payloadType; };
-    void SetInternalSource(bool internalSource) { _internalSource = internalSource; };
-
-    void SetRotation(VideoRotation rotation) { _rotation = rotation; }
-
-private:
-    VCMPacketizationCallback* _sendCallback;
-    media_optimization::MediaOptimization* _mediaOpt;
-    uint8_t _payloadType;
-    bool _internalSource;
-    VideoRotation _rotation;
-
-    EncodedImageCallback* post_encode_callback_;
-
-#ifdef DEBUG_ENCODER_BIT_STREAM
-    FILE* _bitStreamAfterEncoder;
-#endif
-};// end of VCMEncodeFrameCallback class
-
-
-/******************************/
-/* VCMGenericEncoder class    */
-/******************************/
-class VCMGenericEncoder
-{
-    friend class VCMCodecDataBase;
-public:
-    VCMGenericEncoder(VideoEncoder* encoder,
-                      VideoEncoderRateObserver* rate_observer,
-                      VCMEncodedFrameCallback* encoded_frame_callback,
-                      bool internalSource);
-    ~VCMGenericEncoder();
-    /**
-    * Free encoder memory
-    */
-    int32_t Release();
-    /**
-    * Initialize the encoder with the information from the VideoCodec
-    */
-    int32_t InitEncode(const VideoCodec* settings,
-                       int32_t numberOfCores,
-                       size_t maxPayloadSize);
-    /**
-    * Encode raw image
-    * inputFrame        : Frame containing raw image
-    * codecSpecificInfo : Specific codec data
-    * cameraFrameRate   : Request or information from the remote side
-    * frameType         : The requested frame type to encode
-    */
-    int32_t Encode(const VideoFrame& inputFrame,
-                   const CodecSpecificInfo* codecSpecificInfo,
-                   const std::vector<FrameType>& frameTypes);
-
-    void SetEncoderParameters(const EncoderParameters& params);
-    EncoderParameters GetEncoderParameters() const;
-
-    int32_t SetPeriodicKeyFrames(bool enable);
-
-    int32_t RequestFrame(const std::vector<FrameType>& frame_types);
-
-    bool InternalSource() const;
-
-    void OnDroppedFrame();
-
-    bool SupportsNativeHandle() const;
-
-    int GetTargetFramerate();
-
-private:
-    VideoEncoder* const encoder_;
-    VideoEncoderRateObserver* const rate_observer_;
-    VCMEncodedFrameCallback* const vcm_encoded_frame_callback_;
-    const bool internal_source_;
-    mutable rtc::CriticalSection params_lock_;
-    EncoderParameters encoder_params_ GUARDED_BY(params_lock_);
-    VideoRotation rotation_;
-    bool is_screenshare_;
-}; // end of VCMGenericEncoder class
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
diff --git a/webrtc/modules/video_coding/main/source/inter_frame_delay.cc b/webrtc/modules/video_coding/main/source/inter_frame_delay.cc
deleted file mode 100644
index 4786917e16..0000000000
--- a/webrtc/modules/video_coding/main/source/inter_frame_delay.cc
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
-
-namespace webrtc {
-
-VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock)
-{
-    Reset(currentWallClock);
-}
-
-// Resets the delay estimate
-void
-VCMInterFrameDelay::Reset(int64_t currentWallClock)
-{
-    _zeroWallClock = currentWallClock;
-    _wrapArounds = 0;
-    _prevWallClock = 0;
-    _prevTimestamp = 0;
-    _dTS = 0;
-}
-
-// Calculates the delay of a frame with the given timestamp.
-// This method is called when the frame is complete.
-bool
-VCMInterFrameDelay::CalculateDelay(uint32_t timestamp,
-                                int64_t *delay,
-                                int64_t currentWallClock)
-{
-    if (_prevWallClock == 0)
-    {
-        // First set of data, initialization, wait for next frame
-        _prevWallClock = currentWallClock;
-        _prevTimestamp = timestamp;
-        *delay = 0;
-        return true;
-    }
-
-    int32_t prevWrapArounds = _wrapArounds;
-    CheckForWrapArounds(timestamp);
-
-    // This will be -1 for backward wrap arounds and +1 for forward wrap arounds
-    int32_t wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
-
-    // Account for reordering in jitter variance estimate in the future?
-    // Note that this also captures incomplete frames which are grabbed
-    // for decoding after a later frame has been complete, i.e. real
-    // packet losses.
-    if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) || wrapAroundsSincePrev < 0)
-    {
-        *delay = 0;
-        return false;
-    }
-
-    // Compute the compensated timestamp difference and convert it to ms and
-    // round it to closest integer.
-    _dTS = static_cast<int64_t>((timestamp + wrapAroundsSincePrev *
-                (static_cast<int64_t>(1)<<32) - _prevTimestamp) / 90.0 + 0.5);
-
-    // frameDelay is the difference of dT and dTS -- i.e. the difference of
-    // the wall clock time difference and the timestamp difference between
-    // two following frames.
-    *delay = static_cast<int64_t>(currentWallClock - _prevWallClock - _dTS);
-
-    _prevTimestamp = timestamp;
-    _prevWallClock = currentWallClock;
-
-    return true;
-}
-
-// Returns the current difference between incoming timestamps
-uint32_t VCMInterFrameDelay::CurrentTimeStampDiffMs() const
-{
-    if (_dTS < 0)
-    {
-        return 0;
-    }
-    return static_cast<uint32_t>(_dTS);
-}
-
-// Investigates if the timestamp clock has overflowed since the last timestamp and
-// keeps track of the number of wrap arounds since reset.
-void
-VCMInterFrameDelay::CheckForWrapArounds(uint32_t timestamp)
-{
-    if (timestamp < _prevTimestamp)
-    {
-        // This difference will probably be less than -2^31 if we have had a wrap around
-        // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to a Word32,
-        // it should be positive.
-        if (static_cast<int32_t>(timestamp - _prevTimestamp) > 0)
-        {
-            // Forward wrap around
-            _wrapArounds++;
-        }
-    }
-    // This difference will probably be less than -2^31 if we have had a backward wrap around.
-    // Since it is cast to a Word32, it should be positive.
-    else if (static_cast<int32_t>(_prevTimestamp - timestamp) > 0)
-    {
-        // Backward wrap around
-        _wrapArounds--;
-    }
-}
-
-}
diff --git a/webrtc/modules/video_coding/main/source/inter_frame_delay.h b/webrtc/modules/video_coding/main/source/inter_frame_delay.h
deleted file mode 100644
index 58b326ae96..0000000000
--- a/webrtc/modules/video_coding/main/source/inter_frame_delay.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
-#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class VCMInterFrameDelay
-{
-public:
-    VCMInterFrameDelay(int64_t currentWallClock);
-
-    // Resets the estimate. Zeros are given as parameters.
-    void Reset(int64_t currentWallClock);
-
-    // Calculates the delay of a frame with the given timestamp.
-    // This method is called when the frame is complete.
-    //
-    // Input:
-    //          - timestamp         : RTP timestamp of a received frame
-    //          - *delay            : Pointer to memory where the result should be stored
-    //          - currentWallClock  : The current time in milliseconds.
-    //                                Should be -1 for normal operation, only used for testing.
-    // Return value                 : true if OK, false when reordered timestamps
-    bool CalculateDelay(uint32_t timestamp,
-                        int64_t *delay,
-                        int64_t currentWallClock);
-
-    // Returns the current difference between incoming timestamps
-    //
-    // Return value                 : Wrap-around compensated difference between incoming
-    //                                timestamps.
-    uint32_t CurrentTimeStampDiffMs() const;
-
-private:
-    // Controls if the RTP timestamp counter has had a wrap around
-    // between the current and the previously received frame.
-    //
-    // Input:
-    //          - timestmap         : RTP timestamp of the current frame.
-    void CheckForWrapArounds(uint32_t timestamp);
-
-    int64_t         _zeroWallClock; // Local timestamp of the first video packet received
-    int32_t         _wrapArounds;   // Number of wrapArounds detected
-    // The previous timestamp passed to the delay estimate
-    uint32_t        _prevTimestamp;
-    // The previous wall clock timestamp used by the delay estimate
-    int64_t         _prevWallClock;
-    // Wrap-around compensated difference between incoming timestamps
-    int64_t         _dTS;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
diff --git a/webrtc/modules/video_coding/main/source/internal_defines.h b/webrtc/modules/video_coding/main/source/internal_defines.h
deleted file mode 100644
index adc940f20d..0000000000
--- a/webrtc/modules/video_coding/main/source/internal_defines.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
-
-inline uint32_t MaskWord64ToUWord32(int64_t w64)
-{
-    return static_cast<uint32_t>(MASK_32_BITS(w64));
-}
-
-#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
-#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
-
-#define VCM_DEFAULT_CODEC_WIDTH 352
-#define VCM_DEFAULT_CODEC_HEIGHT 288
-#define VCM_DEFAULT_FRAME_RATE 30
-#define VCM_MIN_BITRATE 30
-#define VCM_FLUSH_INDICATOR 4
-
-// Helper macros for creating the static codec list
-#define VCM_NO_CODEC_IDX -1
-#ifdef VIDEOCODEC_VP8
-  #define VCM_VP8_IDX (VCM_NO_CODEC_IDX + 1)
-#else
-  #define VCM_VP8_IDX VCM_NO_CODEC_IDX
-#endif
-#ifdef VIDEOCODEC_VP9
-  #define VCM_VP9_IDX (VCM_VP8_IDX + 1)
-#else
-  #define VCM_VP9_IDX VCM_VP8_IDX
-#endif
-#ifdef VIDEOCODEC_H264
-  #define VCM_H264_IDX (VCM_VP9_IDX + 1)
-#else
-  #define VCM_H264_IDX VCM_VP9_IDX
-#endif
-#ifdef VIDEOCODEC_I420
-  #define VCM_I420_IDX (VCM_H264_IDX + 1)
-#else
-  #define VCM_I420_IDX VCM_H264_IDX
-#endif
-#define VCM_NUM_VIDEO_CODECS_AVAILABLE (VCM_I420_IDX + 1)
-
-#define VCM_NO_RECEIVER_ID 0
-
-inline int32_t VCMId(const int32_t vcmId, const int32_t receiverId = 0)
-{
-    return static_cast<int32_t>((vcmId << 16) + receiverId);
-}
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
diff --git a/webrtc/modules/video_coding/main/source/jitter_estimator.cc b/webrtc/modules/video_coding/main/source/jitter_estimator.cc
deleted file mode 100644
index 5894c88d72..0000000000
--- a/webrtc/modules/video_coding/main/source/jitter_estimator.cc
+++ /dev/null
@@ -1,482 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
-#include "webrtc/modules/video_coding/main/source/rtt_filter.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
-
-#include <assert.h>
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-
-namespace webrtc {
-
-enum { kStartupDelaySamples = 30 };
-enum { kFsAccuStartupSamples = 5 };
-enum { kMaxFramerateEstimate = 200 };
-
-VCMJitterEstimator::VCMJitterEstimator(const Clock* clock,
-                                       int32_t vcmId,
-                                       int32_t receiverId)
-    : _vcmId(vcmId),
-      _receiverId(receiverId),
-      _phi(0.97),
-      _psi(0.9999),
-      _alphaCountMax(400),
-      _thetaLow(0.000001),
-      _nackLimit(3),
-      _numStdDevDelayOutlier(15),
-      _numStdDevFrameSizeOutlier(3),
-      _noiseStdDevs(2.33),       // ~Less than 1% chance
-                                 // (look up in normal distribution table)...
-      _noiseStdDevOffset(30.0),  // ...of getting 30 ms freezes
-      _rttFilter(),
-      fps_counter_(30),  // TODO(sprang): Use an estimator with limit based on
-                         // time, rather than number of samples.
-      low_rate_experiment_(kInit),
-      clock_(clock) {
-  Reset();
-}
-
-VCMJitterEstimator::~VCMJitterEstimator() {
-}
-
-VCMJitterEstimator&
-VCMJitterEstimator::operator=(const VCMJitterEstimator& rhs)
-{
-    if (this != &rhs)
-    {
-        memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov));
-        memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov));
-
-        _vcmId = rhs._vcmId;
-        _receiverId = rhs._receiverId;
-        _avgFrameSize = rhs._avgFrameSize;
-        _varFrameSize = rhs._varFrameSize;
-        _maxFrameSize = rhs._maxFrameSize;
-        _fsSum = rhs._fsSum;
-        _fsCount = rhs._fsCount;
-        _lastUpdateT = rhs._lastUpdateT;
-        _prevEstimate = rhs._prevEstimate;
-        _prevFrameSize = rhs._prevFrameSize;
-        _avgNoise = rhs._avgNoise;
-        _alphaCount = rhs._alphaCount;
-        _filterJitterEstimate = rhs._filterJitterEstimate;
-        _startupCount = rhs._startupCount;
-        _latestNackTimestamp = rhs._latestNackTimestamp;
-        _nackCount = rhs._nackCount;
-        _rttFilter = rhs._rttFilter;
-    }
-    return *this;
-}
-
-// Resets the JitterEstimate
-void
-VCMJitterEstimator::Reset()
-{
-    _theta[0] = 1/(512e3/8);
-    _theta[1] = 0;
-    _varNoise = 4.0;
-
-    _thetaCov[0][0] = 1e-4;
-    _thetaCov[1][1] = 1e2;
-    _thetaCov[0][1] = _thetaCov[1][0] = 0;
-    _Qcov[0][0] = 2.5e-10;
-    _Qcov[1][1] = 1e-10;
-    _Qcov[0][1] = _Qcov[1][0] = 0;
-    _avgFrameSize = 500;
-    _maxFrameSize = 500;
-    _varFrameSize = 100;
-    _lastUpdateT = -1;
-    _prevEstimate = -1.0;
-    _prevFrameSize = 0;
-    _avgNoise = 0.0;
-    _alphaCount = 1;
-    _filterJitterEstimate = 0.0;
-    _latestNackTimestamp = 0;
-    _nackCount = 0;
-    _fsSum = 0;
-    _fsCount = 0;
-    _startupCount = 0;
-    _rttFilter.Reset();
-    fps_counter_.Reset();
-}
-
-void
-VCMJitterEstimator::ResetNackCount()
-{
-    _nackCount = 0;
-}
-
-// Updates the estimates with the new measurements
-void
-VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS, uint32_t frameSizeBytes,
-                                            bool incompleteFrame /* = false */)
-{
-    if (frameSizeBytes == 0)
-    {
-        return;
-    }
-    int deltaFS = frameSizeBytes - _prevFrameSize;
-    if (_fsCount < kFsAccuStartupSamples)
-    {
-        _fsSum += frameSizeBytes;
-        _fsCount++;
-    }
-    else if (_fsCount == kFsAccuStartupSamples)
-    {
-        // Give the frame size filter
-        _avgFrameSize = static_cast<double>(_fsSum) /
-                        static_cast<double>(_fsCount);
-        _fsCount++;
-    }
-    if (!incompleteFrame || frameSizeBytes > _avgFrameSize)
-    {
-        double avgFrameSize = _phi * _avgFrameSize +
-                              (1 - _phi) * frameSizeBytes;
-        if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize))
-        {
-            // Only update the average frame size if this sample wasn't a
-            // key frame
-            _avgFrameSize = avgFrameSize;
-        }
-        // Update the variance anyway since we want to capture cases where we only get
-        // key frames.
-        _varFrameSize = VCM_MAX(_phi * _varFrameSize + (1 - _phi) *
-                                (frameSizeBytes - avgFrameSize) *
-                                (frameSizeBytes - avgFrameSize), 1.0);
-    }
-
-    // Update max frameSize estimate
-    _maxFrameSize = VCM_MAX(_psi * _maxFrameSize, static_cast<double>(frameSizeBytes));
-
-    if (_prevFrameSize == 0)
-    {
-        _prevFrameSize = frameSizeBytes;
-        return;
-    }
-    _prevFrameSize = frameSizeBytes;
-
-    // Only update the Kalman filter if the sample is not considered
-    // an extreme outlier. Even if it is an extreme outlier from a
-    // delay point of view, if the frame size also is large the
-    // deviation is probably due to an incorrect line slope.
-    double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
-
-    if (fabs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
-        frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize))
-    {
-        // Update the variance of the deviation from the
-        // line given by the Kalman filter
-        EstimateRandomJitter(deviation, incompleteFrame);
-        // Prevent updating with frames which have been congested by a large
-        // frame, and therefore arrives almost at the same time as that frame.
-        // This can occur when we receive a large frame (key frame) which
-        // has been delayed. The next frame is of normal size (delta frame),
-        // and thus deltaFS will be << 0. This removes all frame samples
-        // which arrives after a key frame.
-        if ((!incompleteFrame || deviation >= 0.0) &&
-            static_cast<double>(deltaFS) > - 0.25 * _maxFrameSize)
-        {
-            // Update the Kalman filter with the new data
-            KalmanEstimateChannel(frameDelayMS, deltaFS);
-        }
-    }
-    else
-    {
-        int nStdDev = (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier;
-        EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame);
-    }
-    // Post process the total estimated jitter
-    if (_startupCount >= kStartupDelaySamples)
-    {
-        PostProcessEstimate();
-    }
-    else
-    {
-        _startupCount++;
-    }
-}
-
-// Updates the nack/packet ratio
-void
-VCMJitterEstimator::FrameNacked()
-{
-    // Wait until _nackLimit retransmissions has been received,
-    // then always add ~1 RTT delay.
-    // TODO(holmer): Should we ever remove the additional delay if the
-    // the packet losses seem to have stopped? We could for instance scale
-    // the number of RTTs to add with the amount of retransmissions in a given
-    // time interval, or similar.
-    if (_nackCount < _nackLimit)
-    {
-        _nackCount++;
-    }
-}
-
-// Updates Kalman estimate of the channel
-// The caller is expected to sanity check the inputs.
-void
-VCMJitterEstimator::KalmanEstimateChannel(int64_t frameDelayMS,
-                                          int32_t deltaFSBytes)
-{
-    double Mh[2];
-    double hMh_sigma;
-    double kalmanGain[2];
-    double measureRes;
-    double t00, t01;
-
-    // Kalman filtering
-
-    // Prediction
-    // M = M + Q
-    _thetaCov[0][0] += _Qcov[0][0];
-    _thetaCov[0][1] += _Qcov[0][1];
-    _thetaCov[1][0] += _Qcov[1][0];
-    _thetaCov[1][1] += _Qcov[1][1];
-
-    // Kalman gain
-    // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h')
-    // h = [dFS 1]
-    // Mh = M*h'
-    // hMh_sigma = h*M*h' + R
-    Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1];
-    Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1];
-    // sigma weights measurements with a small deltaFS as noisy and
-    // measurements with large deltaFS as good
-    if (_maxFrameSize < 1.0)
-    {
-        return;
-    }
-    double sigma = (300.0 * exp(-fabs(static_cast<double>(deltaFSBytes)) /
-                   (1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise);
-    if (sigma < 1.0)
-    {
-        sigma = 1.0;
-    }
-    hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma;
-    if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) || (hMh_sigma > -1e-9 && hMh_sigma <= 0))
-    {
-        assert(false);
-        return;
-    }
-    kalmanGain[0] = Mh[0] / hMh_sigma;
-    kalmanGain[1] = Mh[1] / hMh_sigma;
-
-    // Correction
-    // theta = theta + K*(dT - h*theta)
-    measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]);
-    _theta[0] += kalmanGain[0] * measureRes;
-    _theta[1] += kalmanGain[1] * measureRes;
-
-    if (_theta[0] < _thetaLow)
-    {
-        _theta[0] = _thetaLow;
-    }
-
-    // M = (I - K*h)*M
-    t00 = _thetaCov[0][0];
-    t01 = _thetaCov[0][1];
-    _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 -
-                      kalmanGain[0] * _thetaCov[1][0];
-    _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 -
-                      kalmanGain[0] * _thetaCov[1][1];
-    _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) -
-                      kalmanGain[1] * deltaFSBytes * t00;
-    _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) -
-                      kalmanGain[1] * deltaFSBytes * t01;
-
-    // Covariance matrix, must be positive semi-definite
-    assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 &&
-           _thetaCov[0][0] * _thetaCov[1][1] - _thetaCov[0][1] * _thetaCov[1][0] >= 0 &&
-           _thetaCov[0][0] >= 0);
-}
-
-// Calculate difference in delay between a sample and the
-// expected delay estimated by the Kalman filter
-double
-VCMJitterEstimator::DeviationFromExpectedDelay(int64_t frameDelayMS,
-                                               int32_t deltaFSBytes) const
-{
-    return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]);
-}
-
-// Estimates the random jitter by calculating the variance of the
-// sample distance from the line given by theta.
-void VCMJitterEstimator::EstimateRandomJitter(double d_dT,
-                                              bool incompleteFrame) {
-  uint64_t now = clock_->TimeInMicroseconds();
-  if (_lastUpdateT != -1) {
-    fps_counter_.AddSample(now - _lastUpdateT);
-  }
-  _lastUpdateT = now;
-
-  if (_alphaCount == 0) {
-    assert(false);
-    return;
-  }
-  double alpha =
-      static_cast<double>(_alphaCount - 1) / static_cast<double>(_alphaCount);
-  _alphaCount++;
-  if (_alphaCount > _alphaCountMax)
-    _alphaCount = _alphaCountMax;
-
-  if (LowRateExperimentEnabled()) {
-    // In order to avoid a low frame rate stream to react slower to changes,
-    // scale the alpha weight relative a 30 fps stream.
-    double fps = GetFrameRate();
-    if (fps > 0.0) {
-      double rate_scale = 30.0 / fps;
-      // At startup, there can be a lot of noise in the fps estimate.
-      // Interpolate rate_scale linearly, from 1.0 at sample #1, to 30.0 / fps
-      // at sample #kStartupDelaySamples.
-      if (_alphaCount < kStartupDelaySamples) {
-        rate_scale =
-            (_alphaCount * rate_scale + (kStartupDelaySamples - _alphaCount)) /
-            kStartupDelaySamples;
-      }
-      alpha = pow(alpha, rate_scale);
-    }
-  }
-
-  double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
-  double varNoise =
-      alpha * _varNoise + (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise);
-  if (!incompleteFrame || varNoise > _varNoise) {
-    _avgNoise = avgNoise;
-    _varNoise = varNoise;
-  }
-  if (_varNoise < 1.0) {
-    // The variance should never be zero, since we might get
-    // stuck and consider all samples as outliers.
-    _varNoise = 1.0;
-  }
-}
-
-double
-VCMJitterEstimator::NoiseThreshold() const
-{
-    double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset;
-    if (noiseThreshold < 1.0)
-    {
-        noiseThreshold = 1.0;
-    }
-    return noiseThreshold;
-}
-
-// Calculates the current jitter estimate from the filtered estimates
-double
-VCMJitterEstimator::CalculateEstimate()
-{
-    double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold();
-
-    // A very low estimate (or negative) is neglected
-    if (ret < 1.0) {
-        if (_prevEstimate <= 0.01)
-        {
-            ret = 1.0;
-        }
-        else
-        {
-            ret = _prevEstimate;
-        }
-    }
-    if (ret > 10000.0) // Sanity
-    {
-        ret = 10000.0;
-    }
-    _prevEstimate = ret;
-    return ret;
-}
-
-void
-VCMJitterEstimator::PostProcessEstimate()
-{
-    _filterJitterEstimate = CalculateEstimate();
-}
-
-void
-VCMJitterEstimator::UpdateRtt(int64_t rttMs)
-{
-    _rttFilter.Update(rttMs);
-}
-
-void
-VCMJitterEstimator::UpdateMaxFrameSize(uint32_t frameSizeBytes)
-{
-    if (_maxFrameSize < frameSizeBytes)
-    {
-        _maxFrameSize = frameSizeBytes;
-    }
-}
-
-// Returns the current filtered estimate if available,
-// otherwise tries to calculate an estimate.
-int VCMJitterEstimator::GetJitterEstimate(double rttMultiplier) {
-  double jitterMS = CalculateEstimate() + OPERATING_SYSTEM_JITTER;
-  if (_filterJitterEstimate > jitterMS)
-    jitterMS = _filterJitterEstimate;
-  if (_nackCount >= _nackLimit)
-    jitterMS += _rttFilter.RttMs() * rttMultiplier;
-
-  if (LowRateExperimentEnabled()) {
-    static const double kJitterScaleLowThreshold = 5.0;
-    static const double kJitterScaleHighThreshold = 10.0;
-    double fps = GetFrameRate();
-    // Ignore jitter for very low fps streams.
-    if (fps < kJitterScaleLowThreshold) {
-      if (fps == 0.0) {
-        return jitterMS;
-      }
-      return 0;
-    }
-
-    // Semi-low frame rate; scale by factor linearly interpolated from 0.0 at
-    // kJitterScaleLowThreshold to 1.0 at kJitterScaleHighThreshold.
-    if (fps < kJitterScaleHighThreshold) {
-      jitterMS =
-          (1.0 / (kJitterScaleHighThreshold - kJitterScaleLowThreshold)) *
-          (fps - kJitterScaleLowThreshold) * jitterMS;
-    }
-  }
-
-  return static_cast<uint32_t>(jitterMS + 0.5);
-}
-
-bool VCMJitterEstimator::LowRateExperimentEnabled() {
-  if (low_rate_experiment_ == kInit) {
-    std::string group =
-        webrtc::field_trial::FindFullName("WebRTC-ReducedJitterDelay");
-    if (group == "Disabled") {
-      low_rate_experiment_ = kDisabled;
-    } else {
-      low_rate_experiment_ = kEnabled;
-    }
-  }
-  return low_rate_experiment_ == kEnabled ? true : false;
-}
-
-double VCMJitterEstimator::GetFrameRate() const {
-  if (fps_counter_.count() == 0)
-    return 0;
-
-  double fps = 1000000.0 / fps_counter_.ComputeMean();
-  // Sanity check.
-  assert(fps >= 0.0);
-  if (fps > kMaxFramerateEstimate) {
-    fps = kMaxFramerateEstimate;
-  }
-  return fps;
-}
-
-}
diff --git a/webrtc/modules/video_coding/main/source/jitter_estimator.h b/webrtc/modules/video_coding/main/source/jitter_estimator.h
deleted file mode 100644
index 46ed67ba1d..0000000000
--- a/webrtc/modules/video_coding/main/source/jitter_estimator.h
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
-
-#include "webrtc/base/rollingaccumulator.h"
-#include "webrtc/modules/video_coding/main/source/rtt_filter.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class Clock;
-
-class VCMJitterEstimator
-{
-public:
-    VCMJitterEstimator(const Clock* clock,
-                       int32_t vcmId = 0,
-                       int32_t receiverId = 0);
-    virtual ~VCMJitterEstimator();
-    VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
-
-    // Resets the estimate to the initial state
-    void Reset();
-    void ResetNackCount();
-
-    // Updates the jitter estimate with the new data.
-    //
-    // Input:
-    //          - frameDelay      : Delay-delta calculated by UTILDelayEstimate in milliseconds
-    //          - frameSize       : Frame size of the current frame.
-    //          - incompleteFrame : Flags if the frame is used to update the estimate before it
-    //                              was complete. Default is false.
-    void UpdateEstimate(int64_t frameDelayMS,
-                        uint32_t frameSizeBytes,
-                        bool incompleteFrame = false);
-
-    // Returns the current jitter estimate in milliseconds and adds
-    // also adds an RTT dependent term in cases of retransmission.
-    //  Input:
-    //          - rttMultiplier  : RTT param multiplier (when applicable).
-    //
-    // Return value                   : Jitter estimate in milliseconds
-    int GetJitterEstimate(double rttMultiplier);
-
-    // Updates the nack counter.
-    void FrameNacked();
-
-    // Updates the RTT filter.
-    //
-    // Input:
-    //          - rttMs               : RTT in ms
-    void UpdateRtt(int64_t rttMs);
-
-    void UpdateMaxFrameSize(uint32_t frameSizeBytes);
-
-    // A constant describing the delay from the jitter buffer
-    // to the delay on the receiving side which is not accounted
-    // for by the jitter buffer nor the decoding delay estimate.
-    static const uint32_t OPERATING_SYSTEM_JITTER = 10;
-
-protected:
-    // These are protected for better testing possibilities
-    double              _theta[2]; // Estimated line parameters (slope, offset)
-    double              _varNoise; // Variance of the time-deviation from the line
-
-    virtual bool LowRateExperimentEnabled();
-
-private:
-    // Updates the Kalman filter for the line describing
-    // the frame size dependent jitter.
-    //
-    // Input:
-    //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in milliseconds
-    //          - deltaFSBytes    : Frame size delta, i.e.
-    //                            : frame size at time T minus frame size at time T-1
-    void KalmanEstimateChannel(int64_t frameDelayMS, int32_t deltaFSBytes);
-
-    // Updates the random jitter estimate, i.e. the variance
-    // of the time deviations from the line given by the Kalman filter.
-    //
-    // Input:
-    //          - d_dT              : The deviation from the kalman estimate
-    //          - incompleteFrame   : True if the frame used to update the estimate
-    //                                with was incomplete
-    void EstimateRandomJitter(double d_dT, bool incompleteFrame);
-
-    double NoiseThreshold() const;
-
-    // Calculates the current jitter estimate.
-    //
-    // Return value                 : The current jitter estimate in milliseconds
-    double CalculateEstimate();
-
-    // Post process the calculated estimate
-    void PostProcessEstimate();
-
-    // Calculates the difference in delay between a sample and the
-    // expected delay estimated by the Kalman filter.
-    //
-    // Input:
-    //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in milliseconds
-    //          - deltaFS         : Frame size delta, i.e. frame size at time
-    //                              T minus frame size at time T-1
-    //
-    // Return value                 : The difference in milliseconds
-    double DeviationFromExpectedDelay(int64_t frameDelayMS,
-                                      int32_t deltaFSBytes) const;
-
-    double GetFrameRate() const;
-
-    // Constants, filter parameters
-    int32_t         _vcmId;
-    int32_t         _receiverId;
-    const double          _phi;
-    const double          _psi;
-    const uint32_t  _alphaCountMax;
-    const double          _thetaLow;
-    const uint32_t  _nackLimit;
-    const int32_t   _numStdDevDelayOutlier;
-    const int32_t   _numStdDevFrameSizeOutlier;
-    const double          _noiseStdDevs;
-    const double          _noiseStdDevOffset;
-
-    double                _thetaCov[2][2]; // Estimate covariance
-    double                _Qcov[2][2];     // Process noise covariance
-    double                _avgFrameSize;   // Average frame size
-    double                _varFrameSize;   // Frame size variance
-    double                _maxFrameSize;   // Largest frame size received (descending
-                                           // with a factor _psi)
-    uint32_t        _fsSum;
-    uint32_t        _fsCount;
-
-    int64_t         _lastUpdateT;
-    double                _prevEstimate;         // The previously returned jitter estimate
-    uint32_t        _prevFrameSize;        // Frame size of the previous frame
-    double                _avgNoise;             // Average of the random jitter
-    uint32_t        _alphaCount;
-    double                _filterJitterEstimate; // The filtered sum of jitter estimates
-
-    uint32_t        _startupCount;
-
-    int64_t         _latestNackTimestamp;  // Timestamp in ms when the latest nack was seen
-    uint32_t        _nackCount;            // Keeps track of the number of nacks received,
-                                                 // but never goes above _nackLimit
-    VCMRttFilter          _rttFilter;
-
-    rtc::RollingAccumulator<uint64_t> fps_counter_;
-    enum ExperimentFlag { kInit, kEnabled, kDisabled };
-    ExperimentFlag low_rate_experiment_;
-    const Clock* clock_;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
diff --git a/webrtc/modules/video_coding/main/source/media_opt_util.cc b/webrtc/modules/video_coding/main/source/media_opt_util.cc
deleted file mode 100644
index 51decbed97..0000000000
--- a/webrtc/modules/video_coding/main/source/media_opt_util.cc
+++ /dev/null
@@ -1,774 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-
-#include <algorithm>
-#include <float.h>
-#include <limits.h>
-#include <math.h>
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/fec_tables_xor.h"
-#include "webrtc/modules/video_coding/main/source/nack_fec_tables.h"
-
-namespace webrtc {
-// Max value of loss rates in off-line model
-static const int kPacketLossMax = 129;
-
-namespace media_optimization {
-
-VCMProtectionMethod::VCMProtectionMethod()
-    : _effectivePacketLoss(0),
-      _protectionFactorK(0),
-      _protectionFactorD(0),
-      _scaleProtKey(2.0f),
-      _maxPayloadSize(1460),
-      _qmRobustness(new VCMQmRobustness()),
-      _useUepProtectionK(false),
-      _useUepProtectionD(true),
-      _corrFecCost(1.0),
-      _type(kNone) {
-}
-
-VCMProtectionMethod::~VCMProtectionMethod()
-{
-    delete _qmRobustness;
-}
-void
-VCMProtectionMethod::UpdateContentMetrics(const
-                                          VideoContentMetrics* contentMetrics)
-{
-    _qmRobustness->UpdateContent(contentMetrics);
-}
-
-VCMNackFecMethod::VCMNackFecMethod(int64_t lowRttNackThresholdMs,
-                                   int64_t highRttNackThresholdMs)
-    : VCMFecMethod(),
-      _lowRttNackMs(lowRttNackThresholdMs),
-      _highRttNackMs(highRttNackThresholdMs),
-      _maxFramesFec(1) {
-  assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
-  assert(highRttNackThresholdMs == -1 ||
-         lowRttNackThresholdMs <= highRttNackThresholdMs);
-  assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
-  _type = kNackFec;
-}
-
-VCMNackFecMethod::~VCMNackFecMethod()
-{
-    //
-}
-bool
-VCMNackFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
-{
-    // Hybrid Nack FEC has three operational modes:
-    // 1. Low RTT (below kLowRttNackMs) - Nack only: Set FEC rate
-    //    (_protectionFactorD) to zero. -1 means no FEC.
-    // 2. High RTT (above _highRttNackMs) - FEC Only: Keep FEC factors.
-    //    -1 means always allow NACK.
-    // 3. Medium RTT values - Hybrid mode: We will only nack the
-    //    residual following the decoding of the FEC (refer to JB logic). FEC
-    //    delta protection factor will be adjusted based on the RTT.
-
-    // Otherwise: we count on FEC; if the RTT is below a threshold, then we
-    // nack the residual, based on a decision made in the JB.
-
-    // Compute the protection factors
-    VCMFecMethod::ProtectionFactor(parameters);
-    if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs)
-    {
-        _protectionFactorD = 0;
-        VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
-    }
-
-    // When in Hybrid mode (RTT range), adjust FEC rates based on the
-    // RTT (NACK effectiveness) - adjustment factor is in the range [0,1].
-    else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs)
-    {
-        // TODO(mikhal): Disabling adjustment temporarily.
-        // uint16_t rttIndex = (uint16_t) parameters->rtt;
-        float adjustRtt = 1.0f;// (float)VCMNackFecTable[rttIndex] / 100.0f;
-
-        // Adjust FEC with NACK on (for delta frame only)
-        // table depends on RTT relative to rttMax (NACK Threshold)
-        _protectionFactorD = static_cast<uint8_t>
-                            (adjustRtt *
-                             static_cast<float>(_protectionFactorD));
-        // update FEC rates after applying adjustment
-        VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
-    }
-
-    return true;
-}
-
-int VCMNackFecMethod::ComputeMaxFramesFec(
-    const VCMProtectionParameters* parameters) {
-  if (parameters->numLayers > 2) {
-    // For more than 2 temporal layers we will only have FEC on the base layer,
-    // and the base layers will be pretty far apart. Therefore we force one
-    // frame FEC.
-    return 1;
-  }
-  // We set the max number of frames to base the FEC on so that on average
-  // we will have complete frames in one RTT. Note that this is an upper
-  // bound, and that the actual number of frames used for FEC is decided by the
-  // RTP module based on the actual number of packets and the protection factor.
-  float base_layer_framerate = parameters->frameRate /
-      static_cast<float>(1 << (parameters->numLayers - 1));
-  int max_frames_fec = std::max(static_cast<int>(
-      2.0f * base_layer_framerate * parameters->rtt /
-      1000.0f + 0.5f), 1);
-  // |kUpperLimitFramesFec| is the upper limit on how many frames we
-  // allow any FEC to be based on.
-  if (max_frames_fec > kUpperLimitFramesFec) {
-    max_frames_fec = kUpperLimitFramesFec;
-  }
-  return max_frames_fec;
-}
-
-int VCMNackFecMethod::MaxFramesFec() const {
-  return _maxFramesFec;
-}
-
-bool VCMNackFecMethod::BitRateTooLowForFec(
-    const VCMProtectionParameters* parameters) {
-  // Bitrate below which we turn off FEC, regardless of reported packet loss.
-  // The condition should depend on resolution and content. For now, use
-  // threshold on bytes per frame, with some effect for the frame size.
-  // The condition for turning off FEC is also based on other factors,
-  // such as |_numLayers|, |_maxFramesFec|, and |_rtt|.
-  int estimate_bytes_per_frame = 1000 * BitsPerFrame(parameters) / 8;
-  int max_bytes_per_frame = kMaxBytesPerFrameForFec;
-  int num_pixels = parameters->codecWidth * parameters->codecHeight;
-  if (num_pixels <= 352 * 288) {
-    max_bytes_per_frame = kMaxBytesPerFrameForFecLow;
-  } else if (num_pixels > 640 * 480) {
-    max_bytes_per_frame = kMaxBytesPerFrameForFecHigh;
-  }
-  // TODO (marpan): add condition based on maximum frames used for FEC,
-  // and expand condition based on frame size.
-  // Max round trip time threshold in ms.
-  const int64_t kMaxRttTurnOffFec = 200;
-  if (estimate_bytes_per_frame < max_bytes_per_frame &&
-      parameters->numLayers < 3 &&
-      parameters->rtt < kMaxRttTurnOffFec) {
-    return true;
-  }
-  return false;
-}
-
-bool
-VCMNackFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
-{
-    // Set the effective packet loss for encoder (based on FEC code).
-    // Compute the effective packet loss and residual packet loss due to FEC.
-    VCMFecMethod::EffectivePacketLoss(parameters);
-    return true;
-}
-
-bool
-VCMNackFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
-    ProtectionFactor(parameters);
-    EffectivePacketLoss(parameters);
-    _maxFramesFec = ComputeMaxFramesFec(parameters);
-    if (BitRateTooLowForFec(parameters)) {
-      _protectionFactorK = 0;
-      _protectionFactorD = 0;
-    }
-
-    // Protection/fec rates obtained above are defined relative to total number
-    // of packets (total rate: source + fec) FEC in RTP module assumes
-    // protection factor is defined relative to source number of packets so we
-    // should convert the factor to reduce mismatch between mediaOpt's rate and
-    // the actual one
-    _protectionFactorK = VCMFecMethod::ConvertFECRate(_protectionFactorK);
-    _protectionFactorD = VCMFecMethod::ConvertFECRate(_protectionFactorD);
-
-    return true;
-}
-
-VCMNackMethod::VCMNackMethod():
-VCMProtectionMethod()
-{
-    _type = kNack;
-}
-
-VCMNackMethod::~VCMNackMethod()
-{
-    //
-}
-
-bool
-VCMNackMethod::EffectivePacketLoss(const VCMProtectionParameters* parameter)
-{
-    // Effective Packet Loss, NA in current version.
-    _effectivePacketLoss = 0;
-    return true;
-}
-
-bool
-VCMNackMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
-    // Compute the effective packet loss
-    EffectivePacketLoss(parameters);
-
-    // nackCost  = (bitRate - nackCost) * (lossPr)
-    return true;
-}
-
-VCMFecMethod::VCMFecMethod():
-VCMProtectionMethod()
-{
-    _type = kFec;
-}
-VCMFecMethod::~VCMFecMethod()
-{
-    //
-}
-
-uint8_t
-VCMFecMethod::BoostCodeRateKey(uint8_t packetFrameDelta,
-                               uint8_t packetFrameKey) const
-{
-    uint8_t boostRateKey = 2;
-    // Default: ratio scales the FEC protection up for I frames
-    uint8_t ratio = 1;
-
-    if (packetFrameDelta > 0)
-    {
-        ratio = (int8_t) (packetFrameKey / packetFrameDelta);
-    }
-    ratio = VCM_MAX(boostRateKey, ratio);
-
-    return ratio;
-}
-
-uint8_t
-VCMFecMethod::ConvertFECRate(uint8_t codeRateRTP) const
-{
-    return static_cast<uint8_t> (VCM_MIN(255,(0.5 + 255.0 * codeRateRTP /
-                                      (float)(255 - codeRateRTP))));
-}
-
-// Update FEC with protectionFactorD
-void
-VCMFecMethod::UpdateProtectionFactorD(uint8_t protectionFactorD)
-{
-    _protectionFactorD = protectionFactorD;
-}
-
-// Update FEC with protectionFactorK
-void
-VCMFecMethod::UpdateProtectionFactorK(uint8_t protectionFactorK)
-{
-    _protectionFactorK = protectionFactorK;
-}
-
-bool
-VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
-{
-    // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
-
-    // No protection if (filtered) packetLoss is 0
-    uint8_t packetLoss = (uint8_t) (255 * parameters->lossPr);
-    if (packetLoss == 0)
-    {
-        _protectionFactorK = 0;
-        _protectionFactorD = 0;
-         return true;
-    }
-
-    // Parameters for FEC setting:
-    // first partition size, thresholds, table pars, spatial resoln fac.
-
-    // First partition protection: ~ 20%
-    uint8_t firstPartitionProt = (uint8_t) (255 * 0.20);
-
-    // Minimum protection level needed to generate one FEC packet for one
-    // source packet/frame (in RTP sender)
-    uint8_t minProtLevelFec = 85;
-
-    // Threshold on packetLoss and bitRrate/frameRate (=average #packets),
-    // above which we allocate protection to cover at least first partition.
-    uint8_t lossThr = 0;
-    uint8_t packetNumThr = 1;
-
-    // Parameters for range of rate index of table.
-    const uint8_t ratePar1 = 5;
-    const uint8_t ratePar2 = 49;
-
-    // Spatial resolution size, relative to a reference size.
-    float spatialSizeToRef = static_cast<float>
-                           (parameters->codecWidth * parameters->codecHeight) /
-                           (static_cast<float>(704 * 576));
-    // resolnFac: This parameter will generally increase/decrease the FEC rate
-    // (for fixed bitRate and packetLoss) based on system size.
-    // Use a smaller exponent (< 1) to control/soften system size effect.
-    const float resolnFac = 1.0 / powf(spatialSizeToRef, 0.3f);
-
-    const int bitRatePerFrame = BitsPerFrame(parameters);
-
-
-    // Average number of packets per frame (source and fec):
-    const uint8_t avgTotPackets = 1 + (uint8_t)
-                                        ((float) bitRatePerFrame * 1000.0
-                                       / (float) (8.0 * _maxPayloadSize) + 0.5);
-
-    // FEC rate parameters: for P and I frame
-    uint8_t codeRateDelta = 0;
-    uint8_t codeRateKey = 0;
-
-    // Get index for table: the FEC protection depends on an effective rate.
-    // The range on the rate index corresponds to rates (bps)
-    // from ~200k to ~8000k, for 30fps
-    const uint16_t effRateFecTable = static_cast<uint16_t>
-                                           (resolnFac * bitRatePerFrame);
-    uint8_t rateIndexTable =
-        (uint8_t) VCM_MAX(VCM_MIN((effRateFecTable - ratePar1) /
-                                         ratePar1, ratePar2), 0);
-
-    // Restrict packet loss range to 50:
-    // current tables defined only up to 50%
-    if (packetLoss >= kPacketLossMax)
-    {
-        packetLoss = kPacketLossMax - 1;
-    }
-    uint16_t indexTable = rateIndexTable * kPacketLossMax + packetLoss;
-
-    // Check on table index
-    assert(indexTable < kSizeCodeRateXORTable);
-
-    // Protection factor for P frame
-    codeRateDelta = kCodeRateXORTable[indexTable];
-
-    if (packetLoss > lossThr && avgTotPackets > packetNumThr)
-    {
-        // Set a minimum based on first partition size.
-        if (codeRateDelta < firstPartitionProt)
-        {
-            codeRateDelta = firstPartitionProt;
-        }
-    }
-
-    // Check limit on amount of protection for P frame; 50% is max.
-    if (codeRateDelta >= kPacketLossMax)
-    {
-        codeRateDelta = kPacketLossMax - 1;
-    }
-
-    float adjustFec = 1.0f;
-    // Avoid additional adjustments when layers are active.
-    // TODO(mikhal/marco): Update adjusmtent based on layer info.
-    if (parameters->numLayers == 1)
-    {
-        adjustFec = _qmRobustness->AdjustFecFactor(codeRateDelta,
-                                                   parameters->bitRate,
-                                                   parameters->frameRate,
-                                                   parameters->rtt,
-                                                   packetLoss);
-    }
-
-    codeRateDelta = static_cast<uint8_t>(codeRateDelta * adjustFec);
-
-    // For Key frame:
-    // Effectively at a higher rate, so we scale/boost the rate
-    // The boost factor may depend on several factors: ratio of packet
-    // number of I to P frames, how much protection placed on P frames, etc.
-    const uint8_t packetFrameDelta = (uint8_t)
-                                           (0.5 + parameters->packetsPerFrame);
-    const uint8_t packetFrameKey = (uint8_t)
-                                         (0.5 + parameters->packetsPerFrameKey);
-    const uint8_t boostKey = BoostCodeRateKey(packetFrameDelta,
-                                                    packetFrameKey);
-
-    rateIndexTable = (uint8_t) VCM_MAX(VCM_MIN(
-                      1 + (boostKey * effRateFecTable - ratePar1) /
-                      ratePar1,ratePar2),0);
-    uint16_t indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
-
-    indexTableKey = VCM_MIN(indexTableKey, kSizeCodeRateXORTable);
-
-    // Check on table index
-    assert(indexTableKey < kSizeCodeRateXORTable);
-
-    // Protection factor for I frame
-    codeRateKey = kCodeRateXORTable[indexTableKey];
-
-    // Boosting for Key frame.
-    int boostKeyProt = _scaleProtKey * codeRateDelta;
-    if (boostKeyProt >= kPacketLossMax)
-    {
-        boostKeyProt = kPacketLossMax - 1;
-    }
-
-    // Make sure I frame protection is at least larger than P frame protection,
-    // and at least as high as filtered packet loss.
-    codeRateKey = static_cast<uint8_t> (VCM_MAX(packetLoss,
-            VCM_MAX(boostKeyProt, codeRateKey)));
-
-    // Check limit on amount of protection for I frame: 50% is max.
-    if (codeRateKey >= kPacketLossMax)
-    {
-        codeRateKey = kPacketLossMax - 1;
-    }
-
-    _protectionFactorK = codeRateKey;
-    _protectionFactorD = codeRateDelta;
-
-    // Generally there is a rate mis-match between the FEC cost estimated
-    // in mediaOpt and the actual FEC cost sent out in RTP module.
-    // This is more significant at low rates (small # of source packets), where
-    // the granularity of the FEC decreases. In this case, non-zero protection
-    // in mediaOpt may generate 0 FEC packets in RTP sender (since actual #FEC
-    // is based on rounding off protectionFactor on actual source packet number).
-    // The correction factor (_corrFecCost) attempts to corrects this, at least
-    // for cases of low rates (small #packets) and low protection levels.
-
-    float numPacketsFl = 1.0f + ((float) bitRatePerFrame * 1000.0
-                                / (float) (8.0 * _maxPayloadSize) + 0.5);
-
-    const float estNumFecGen = 0.5f + static_cast<float> (_protectionFactorD *
-                                                         numPacketsFl / 255.0f);
-
-
-    // We reduce cost factor (which will reduce overhead for FEC and
-    // hybrid method) and not the protectionFactor.
-    _corrFecCost = 1.0f;
-    if (estNumFecGen < 1.1f && _protectionFactorD < minProtLevelFec)
-    {
-        _corrFecCost = 0.5f;
-    }
-    if (estNumFecGen < 0.9f && _protectionFactorD < minProtLevelFec)
-    {
-        _corrFecCost = 0.0f;
-    }
-
-     // TODO (marpan): Set the UEP protection on/off for Key and Delta frames
-    _useUepProtectionK = _qmRobustness->SetUepProtection(codeRateKey,
-                                                         parameters->bitRate,
-                                                         packetLoss,
-                                                         0);
-
-    _useUepProtectionD = _qmRobustness->SetUepProtection(codeRateDelta,
-                                                         parameters->bitRate,
-                                                         packetLoss,
-                                                         1);
-
-    // DONE WITH FEC PROTECTION SETTINGS
-    return true;
-}
-
-int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
-  // When temporal layers are available FEC will only be applied on the base
-  // layer.
-  const float bitRateRatio =
-    kVp8LayerRateAlloction[parameters->numLayers - 1][0];
-  float frameRateRatio = powf(1 / 2.0, parameters->numLayers - 1);
-  float bitRate = parameters->bitRate * bitRateRatio;
-  float frameRate = parameters->frameRate * frameRateRatio;
-
-  // TODO(mikhal): Update factor following testing.
-  float adjustmentFactor = 1;
-
-  // Average bits per frame (units of kbits)
-  return static_cast<int>(adjustmentFactor * bitRate / frameRate);
-}
-
-bool
-VCMFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
-{
-    // Effective packet loss to encoder is based on RPL (residual packet loss)
-    // this is a soft setting based on degree of FEC protection
-    // RPL = received/input packet loss - average_FEC_recovery
-    // note: received/input packet loss may be filtered based on FilteredLoss
-
-    // Effective Packet Loss, NA in current version.
-    _effectivePacketLoss = 0;
-
-    return true;
-}
-
-bool
-VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
-    // Compute the protection factor
-    ProtectionFactor(parameters);
-
-    // Compute the effective packet loss
-    EffectivePacketLoss(parameters);
-
-    // Protection/fec rates obtained above is defined relative to total number
-    // of packets (total rate: source+fec) FEC in RTP module assumes protection
-    // factor is defined relative to source number of packets so we should
-    // convert the factor to reduce mismatch between mediaOpt suggested rate and
-    // the actual rate
-    _protectionFactorK = ConvertFECRate(_protectionFactorK);
-    _protectionFactorD = ConvertFECRate(_protectionFactorD);
-
-    return true;
-}
-VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs):
-_currentParameters(),
-_rtt(0),
-_lossPr(0.0f),
-_bitRate(0.0f),
-_frameRate(0.0f),
-_keyFrameSize(0.0f),
-_fecRateKey(0),
-_fecRateDelta(0),
-_lastPrUpdateT(0),
-_lossPr255(0.9999f),
-_lossPrHistory(),
-_shortMaxLossPr255(0),
-_packetsPerFrame(0.9999f),
-_packetsPerFrameKey(0.9999f),
-_codecWidth(0),
-_codecHeight(0),
-_numLayers(1)
-{
-    Reset(nowMs);
-}
-
-VCMLossProtectionLogic::~VCMLossProtectionLogic()
-{
-    Release();
-}
-
-void VCMLossProtectionLogic::SetMethod(
-    enum VCMProtectionMethodEnum newMethodType) {
-  if (_selectedMethod && _selectedMethod->Type() == newMethodType)
-    return;
-
-  switch(newMethodType) {
-    case kNack:
-      _selectedMethod.reset(new VCMNackMethod());
-      break;
-    case kFec:
-      _selectedMethod.reset(new VCMFecMethod());
-      break;
-    case kNackFec:
-      _selectedMethod.reset(new VCMNackFecMethod(kLowRttNackMs, -1));
-      break;
-    case kNone:
-      _selectedMethod.reset();
-      break;
-  }
-  UpdateMethod();
-}
-
-void
-VCMLossProtectionLogic::UpdateRtt(int64_t rtt)
-{
-    _rtt = rtt;
-}
-
-void
-VCMLossProtectionLogic::UpdateMaxLossHistory(uint8_t lossPr255,
-                                             int64_t now)
-{
-    if (_lossPrHistory[0].timeMs >= 0 &&
-        now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs)
-    {
-        if (lossPr255 > _shortMaxLossPr255)
-        {
-            _shortMaxLossPr255 = lossPr255;
-        }
-    }
-    else
-    {
-        // Only add a new value to the history once a second
-        if (_lossPrHistory[0].timeMs == -1)
-        {
-            // First, no shift
-            _shortMaxLossPr255 = lossPr255;
-        }
-        else
-        {
-            // Shift
-            for (int32_t i = (kLossPrHistorySize - 2); i >= 0; i--)
-            {
-                _lossPrHistory[i + 1].lossPr255 = _lossPrHistory[i].lossPr255;
-                _lossPrHistory[i + 1].timeMs = _lossPrHistory[i].timeMs;
-            }
-        }
-        if (_shortMaxLossPr255 == 0)
-        {
-            _shortMaxLossPr255 = lossPr255;
-        }
-
-        _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
-        _lossPrHistory[0].timeMs = now;
-        _shortMaxLossPr255 = 0;
-    }
-}
-
-uint8_t
-VCMLossProtectionLogic::MaxFilteredLossPr(int64_t nowMs) const
-{
-    uint8_t maxFound = _shortMaxLossPr255;
-    if (_lossPrHistory[0].timeMs == -1)
-    {
-        return maxFound;
-    }
-    for (int32_t i = 0; i < kLossPrHistorySize; i++)
-    {
-        if (_lossPrHistory[i].timeMs == -1)
-        {
-            break;
-        }
-        if (nowMs - _lossPrHistory[i].timeMs >
-            kLossPrHistorySize * kLossPrShortFilterWinMs)
-        {
-            // This sample (and all samples after this) is too old
-            break;
-        }
-        if (_lossPrHistory[i].lossPr255 > maxFound)
-        {
-            // This sample is the largest one this far into the history
-            maxFound = _lossPrHistory[i].lossPr255;
-        }
-    }
-    return maxFound;
-}
-
-uint8_t VCMLossProtectionLogic::FilteredLoss(
-    int64_t nowMs,
-    FilterPacketLossMode filter_mode,
-    uint8_t lossPr255) {
-
-  // Update the max window filter.
-  UpdateMaxLossHistory(lossPr255, nowMs);
-
-  // Update the recursive average filter.
-  _lossPr255.Apply(static_cast<float> (nowMs - _lastPrUpdateT),
-                   static_cast<float> (lossPr255));
-  _lastPrUpdateT = nowMs;
-
-  // Filtered loss: default is received loss (no filtering).
-  uint8_t filtered_loss = lossPr255;
-
-  switch (filter_mode) {
-    case kNoFilter:
-      break;
-    case kAvgFilter:
-      filtered_loss = static_cast<uint8_t>(_lossPr255.filtered() + 0.5);
-      break;
-    case kMaxFilter:
-      filtered_loss = MaxFilteredLossPr(nowMs);
-      break;
-  }
-
-  return filtered_loss;
-}
-
-void
-VCMLossProtectionLogic::UpdateFilteredLossPr(uint8_t packetLossEnc)
-{
-    _lossPr = (float) packetLossEnc / (float) 255.0;
-}
-
-void
-VCMLossProtectionLogic::UpdateBitRate(float bitRate)
-{
-    _bitRate = bitRate;
-}
-
-void
-VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets, int64_t nowMs)
-{
-    _packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
-                           nPackets);
-    _lastPacketPerFrameUpdateT = nowMs;
-}
-
-void
-VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs)
-{
-    _packetsPerFrameKey.Apply(static_cast<float>(nowMs -
-                              _lastPacketPerFrameUpdateTKey), nPackets);
-    _lastPacketPerFrameUpdateTKey = nowMs;
-}
-
-void
-VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize)
-{
-    _keyFrameSize = keyFrameSize;
-}
-
-void
-VCMLossProtectionLogic::UpdateFrameSize(uint16_t width,
-                                        uint16_t height)
-{
-    _codecWidth = width;
-    _codecHeight = height;
-}
-
-void VCMLossProtectionLogic::UpdateNumLayers(int numLayers) {
-  _numLayers = (numLayers == 0) ? 1 : numLayers;
-}
-
-bool
-VCMLossProtectionLogic::UpdateMethod()
-{
-    if (!_selectedMethod)
-      return false;
-    _currentParameters.rtt = _rtt;
-    _currentParameters.lossPr = _lossPr;
-    _currentParameters.bitRate = _bitRate;
-    _currentParameters.frameRate = _frameRate; // rename actual frame rate?
-    _currentParameters.keyFrameSize = _keyFrameSize;
-    _currentParameters.fecRateDelta = _fecRateDelta;
-    _currentParameters.fecRateKey = _fecRateKey;
-    _currentParameters.packetsPerFrame = _packetsPerFrame.filtered();
-    _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.filtered();
-    _currentParameters.codecWidth = _codecWidth;
-    _currentParameters.codecHeight = _codecHeight;
-    _currentParameters.numLayers = _numLayers;
-    return _selectedMethod->UpdateParameters(&_currentParameters);
-}
-
-VCMProtectionMethod*
-VCMLossProtectionLogic::SelectedMethod() const
-{
-    return _selectedMethod.get();
-}
-
-VCMProtectionMethodEnum VCMLossProtectionLogic::SelectedType() const {
-  return _selectedMethod ? _selectedMethod->Type() : kNone;
-}
-
-void
-VCMLossProtectionLogic::Reset(int64_t nowMs)
-{
-    _lastPrUpdateT = nowMs;
-    _lastPacketPerFrameUpdateT = nowMs;
-    _lastPacketPerFrameUpdateTKey = nowMs;
-    _lossPr255.Reset(0.9999f);
-    _packetsPerFrame.Reset(0.9999f);
-    _fecRateDelta = _fecRateKey = 0;
-    for (int32_t i = 0; i < kLossPrHistorySize; i++)
-    {
-        _lossPrHistory[i].lossPr255 = 0;
-        _lossPrHistory[i].timeMs = -1;
-    }
-    _shortMaxLossPr255 = 0;
-    Release();
-}
-
-void VCMLossProtectionLogic::Release() {
-  _selectedMethod.reset();
-}
-
-}  // namespace media_optimization
-}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/media_opt_util.h b/webrtc/modules/video_coding/main/source/media_opt_util.h
deleted file mode 100644
index 2085bbcde9..0000000000
--- a/webrtc/modules/video_coding/main/source/media_opt_util.h
+++ /dev/null
@@ -1,364 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "webrtc/base/exp_filter.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-namespace media_optimization {
-
-// Number of time periods used for (max) window filter for packet loss
-// TODO (marpan): set reasonable window size for filtered packet loss,
-// adjustment should be based on logged/real data of loss stats/correlation.
-enum { kLossPrHistorySize = 10 };
-
-// 1000 ms, total filter length is (kLossPrHistorySize * 1000) ms
-enum { kLossPrShortFilterWinMs = 1000 };
-
-// The type of filter used on the received packet loss reports.
-enum FilterPacketLossMode {
-  kNoFilter,    // No filtering on received loss.
-  kAvgFilter,   // Recursive average filter.
-  kMaxFilter    // Max-window filter, over the time interval of:
-                // (kLossPrHistorySize * kLossPrShortFilterWinMs) ms.
-};
-
-// Thresholds for hybrid NACK/FEC
-// common to media optimization and the jitter buffer.
-const int64_t kLowRttNackMs = 20;
-
-struct VCMProtectionParameters
-{
-    VCMProtectionParameters() : rtt(0), lossPr(0.0f), bitRate(0.0f),
-        packetsPerFrame(0.0f), packetsPerFrameKey(0.0f), frameRate(0.0f),
-        keyFrameSize(0.0f), fecRateDelta(0), fecRateKey(0),
-        codecWidth(0), codecHeight(0),
-        numLayers(1)
-        {}
-
-    int64_t             rtt;
-    float               lossPr;
-    float               bitRate;
-    float               packetsPerFrame;
-    float               packetsPerFrameKey;
-    float               frameRate;
-    float               keyFrameSize;
-    uint8_t       fecRateDelta;
-    uint8_t       fecRateKey;
-    uint16_t      codecWidth;
-    uint16_t      codecHeight;
-    int                 numLayers;
-};
-
-
-/******************************/
-/* VCMProtectionMethod class  */
-/******************************/
-
-enum VCMProtectionMethodEnum
-{
-    kNack,
-    kFec,
-    kNackFec,
-    kNone
-};
-
-class VCMLossProbabilitySample
-{
-public:
-    VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {};
-
-    uint8_t     lossPr255;
-    int64_t     timeMs;
-};
-
-
-class VCMProtectionMethod
-{
-public:
-    VCMProtectionMethod();
-    virtual ~VCMProtectionMethod();
-
-    // Updates the efficiency of the method using the parameters provided
-    //
-    // Input:
-    //         - parameters         : Parameters used to calculate efficiency
-    //
-    // Return value                 : True if this method is recommended in
-    //                                the given conditions.
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
-
-    // Returns the protection type
-    //
-    // Return value                 : The protection type
-    enum VCMProtectionMethodEnum Type() const { return _type; }
-
-    // Returns the effective packet loss for ER, required by this protection method
-    //
-    // Return value                 : Required effective packet loss
-    virtual uint8_t RequiredPacketLossER() { return _effectivePacketLoss; }
-
-    // Extracts the FEC protection factor for Key frame, required by this protection method
-    //
-    // Return value                 : Required protectionFactor for Key frame
-    virtual uint8_t RequiredProtectionFactorK() { return _protectionFactorK; }
-
-    // Extracts the FEC protection factor for Delta frame, required by this protection method
-    //
-    // Return value                 : Required protectionFactor for delta frame
-    virtual uint8_t RequiredProtectionFactorD() { return _protectionFactorD; }
-
-    // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
-    //
-    // Return value                 : Required Unequal protection on/off state.
-    virtual bool RequiredUepProtectionK() { return _useUepProtectionK; }
-
-    // Extracts whether the the FEC Unequal protection (UEP) is used for Delta frame.
-    //
-    // Return value                 : Required Unequal protection on/off state.
-    virtual bool RequiredUepProtectionD() { return _useUepProtectionD; }
-
-    virtual int MaxFramesFec() const { return 1; }
-
-    // Updates content metrics
-    void UpdateContentMetrics(const VideoContentMetrics* contentMetrics);
-
-protected:
-
-    uint8_t                        _effectivePacketLoss;
-    uint8_t                        _protectionFactorK;
-    uint8_t                        _protectionFactorD;
-    // Estimation of residual loss after the FEC
-    float                                _scaleProtKey;
-    int32_t                        _maxPayloadSize;
-
-    VCMQmRobustness*                     _qmRobustness;
-    bool                                 _useUepProtectionK;
-    bool                                 _useUepProtectionD;
-    float                                _corrFecCost;
-    enum VCMProtectionMethodEnum         _type;
-};
-
-class VCMNackMethod : public VCMProtectionMethod
-{
-public:
-    VCMNackMethod();
-    virtual ~VCMNackMethod();
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
-    // Get the effective packet loss
-    bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
-};
-
-class VCMFecMethod : public VCMProtectionMethod
-{
-public:
-    VCMFecMethod();
-    virtual ~VCMFecMethod();
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
-    // Get the effective packet loss for ER
-    bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
-    // Get the FEC protection factors
-    bool ProtectionFactor(const VCMProtectionParameters* parameters);
-    // Get the boost for key frame protection
-    uint8_t BoostCodeRateKey(uint8_t packetFrameDelta,
-                                   uint8_t packetFrameKey) const;
-    // Convert the rates: defined relative to total# packets or source# packets
-    uint8_t ConvertFECRate(uint8_t codeRate) const;
-    // Get the average effective recovery from FEC: for random loss model
-    float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
-    // Update FEC with protectionFactorD
-    void UpdateProtectionFactorD(uint8_t protectionFactorD);
-    // Update FEC with protectionFactorK
-    void UpdateProtectionFactorK(uint8_t protectionFactorK);
-    // Compute the bits per frame. Account for temporal layers when applicable.
-    int BitsPerFrame(const VCMProtectionParameters* parameters);
-
-protected:
-    enum { kUpperLimitFramesFec = 6 };
-    // Thresholds values for the bytes/frame and round trip time, below which we
-    // may turn off FEC, depending on |_numLayers| and |_maxFramesFec|.
-    // Max bytes/frame for VGA, corresponds to ~140k at 25fps.
-    enum { kMaxBytesPerFrameForFec = 700 };
-    // Max bytes/frame for CIF and lower: corresponds to ~80k at 25fps.
-    enum { kMaxBytesPerFrameForFecLow = 400 };
-    // Max bytes/frame for frame size larger than VGA, ~200k at 25fps.
-    enum { kMaxBytesPerFrameForFecHigh = 1000 };
-};
-
-
-class VCMNackFecMethod : public VCMFecMethod
-{
-public:
-    VCMNackFecMethod(int64_t lowRttNackThresholdMs,
-                     int64_t highRttNackThresholdMs);
-    virtual ~VCMNackFecMethod();
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
-    // Get the effective packet loss for ER
-    bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
-    // Get the protection factors
-    bool ProtectionFactor(const VCMProtectionParameters* parameters);
-    // Get the max number of frames the FEC is allowed to be based on.
-    int MaxFramesFec() const;
-    // Turn off the FEC based on low bitrate and other factors.
-    bool BitRateTooLowForFec(const VCMProtectionParameters* parameters);
-private:
-    int ComputeMaxFramesFec(const VCMProtectionParameters* parameters);
-
-    int64_t _lowRttNackMs;
-    int64_t _highRttNackMs;
-    int _maxFramesFec;
-};
-
-class VCMLossProtectionLogic
-{
-public:
-    VCMLossProtectionLogic(int64_t nowMs);
-    ~VCMLossProtectionLogic();
-
-    // Set the protection method to be used
-    //
-    // Input:
-    //        - newMethodType    : New requested protection method type. If one
-    //                           is already set, it will be deleted and replaced
-    void SetMethod(VCMProtectionMethodEnum newMethodType);
-
-    // Update the round-trip time
-    //
-    // Input:
-    //          - rtt           : Round-trip time in seconds.
-    void UpdateRtt(int64_t rtt);
-
-    // Update the filtered packet loss.
-    //
-    // Input:
-    //          - packetLossEnc :  The reported packet loss filtered
-    //                             (max window or average)
-    void UpdateFilteredLossPr(uint8_t packetLossEnc);
-
-    // Update the current target bit rate.
-    //
-    // Input:
-    //          - bitRate          : The current target bit rate in kbits/s
-    void UpdateBitRate(float bitRate);
-
-    // Update the number of packets per frame estimate, for delta frames
-    //
-    // Input:
-    //          - nPackets         : Number of packets in the latest sent frame.
-    void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
-
-   // Update the number of packets per frame estimate, for key frames
-    //
-    // Input:
-    //          - nPackets         : umber of packets in the latest sent frame.
-    void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
-
-    // Update the keyFrameSize estimate
-    //
-    // Input:
-    //          - keyFrameSize     : The size of the latest sent key frame.
-    void UpdateKeyFrameSize(float keyFrameSize);
-
-    // Update the frame rate
-    //
-    // Input:
-    //          - frameRate        : The current target frame rate.
-    void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
-
-    // Update the frame size
-    //
-    // Input:
-    //          - width        : The codec frame width.
-    //          - height       : The codec frame height.
-    void UpdateFrameSize(uint16_t width, uint16_t height);
-
-    // Update the number of active layers
-    //
-    // Input:
-    //          - numLayers    : Number of layers used.
-    void UpdateNumLayers(int numLayers);
-
-    // The amount of packet loss to cover for with FEC.
-    //
-    // Input:
-    //          - fecRateKey      : Packet loss to cover for with FEC when
-    //                              sending key frames.
-    //          - fecRateDelta    : Packet loss to cover for with FEC when
-    //                              sending delta frames.
-    void UpdateFECRates(uint8_t fecRateKey, uint8_t fecRateDelta)
-                       { _fecRateKey = fecRateKey;
-                         _fecRateDelta = fecRateDelta; }
-
-    // Update the protection methods with the current VCMProtectionParameters
-    // and set the requested protection settings.
-    // Return value     : Returns true on update
-    bool UpdateMethod();
-
-    // Returns the method currently selected.
-    //
-    // Return value                 : The protection method currently selected.
-    VCMProtectionMethod* SelectedMethod() const;
-
-    // Return the protection type of the currently selected method
-    VCMProtectionMethodEnum SelectedType() const;
-
-    // Updates the filtered loss for the average and max window packet loss,
-    // and returns the filtered loss probability in the interval [0, 255].
-    // The returned filtered loss value depends on the parameter |filter_mode|.
-    // The input parameter |lossPr255| is the received packet loss.
-
-    // Return value                 : The filtered loss probability
-    uint8_t FilteredLoss(int64_t nowMs, FilterPacketLossMode filter_mode,
-                               uint8_t lossPr255);
-
-    void Reset(int64_t nowMs);
-
-    void Release();
-
-private:
-    // Sets the available loss protection methods.
-    void UpdateMaxLossHistory(uint8_t lossPr255, int64_t now);
-    uint8_t MaxFilteredLossPr(int64_t nowMs) const;
-    rtc::scoped_ptr<VCMProtectionMethod> _selectedMethod;
-    VCMProtectionParameters _currentParameters;
-    int64_t _rtt;
-    float _lossPr;
-    float _bitRate;
-    float _frameRate;
-    float _keyFrameSize;
-    uint8_t _fecRateKey;
-    uint8_t _fecRateDelta;
-    int64_t _lastPrUpdateT;
-    int64_t _lastPacketPerFrameUpdateT;
-    int64_t _lastPacketPerFrameUpdateTKey;
-    rtc::ExpFilter _lossPr255;
-    VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize];
-    uint8_t _shortMaxLossPr255;
-    rtc::ExpFilter _packetsPerFrame;
-    rtc::ExpFilter _packetsPerFrameKey;
-    uint16_t _codecWidth;
-    uint16_t _codecHeight;
-    int _numLayers;
-};
-
-}  // namespace media_optimization
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
diff --git a/webrtc/modules/video_coding/main/source/nack_fec_tables.h b/webrtc/modules/video_coding/main/source/nack_fec_tables.h
deleted file mode 100644
index b82bb1b4ba..0000000000
--- a/webrtc/modules/video_coding/main/source/nack_fec_tables.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
-
-namespace webrtc
-{
-
-// Table for adjusting FEC rate for NACK/FEC protection method
-// Table values are built as a sigmoid function, ranging from 0 to 100, based on
-// the HybridNackTH values defined in media_opt_util.h.
-const uint16_t VCMNackFecTable[100] = {
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-1,
-1,
-1,
-1,
-1,
-2,
-2,
-2,
-3,
-3,
-4,
-5,
-6,
-7,
-9,
-10,
-12,
-15,
-18,
-21,
-24,
-28,
-32,
-37,
-41,
-46,
-51,
-56,
-61,
-66,
-70,
-74,
-78,
-81,
-84,
-86,
-89,
-90,
-92,
-93,
-95,
-95,
-96,
-97,
-97,
-98,
-98,
-99,
-99,
-99,
-99,
-99,
-99,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
diff --git a/webrtc/modules/video_coding/main/source/packet.h b/webrtc/modules/video_coding/main/source/packet.h
deleted file mode 100644
index 80bf532502..0000000000
--- a/webrtc/modules/video_coding/main/source/packet.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
-#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class VCMPacket {
-public:
-    VCMPacket();
-    VCMPacket(const uint8_t* ptr,
-              const size_t size,
-              const WebRtcRTPHeader& rtpHeader);
-    VCMPacket(const uint8_t* ptr,
-              size_t size,
-              uint16_t seqNum,
-              uint32_t timestamp,
-              bool markerBit);
-
-    void Reset();
-
-    uint8_t           payloadType;
-    uint32_t          timestamp;
-    // NTP time of the capture time in local timebase in milliseconds.
-    int64_t ntp_time_ms_;
-    uint16_t          seqNum;
-    const uint8_t*    dataPtr;
-    size_t          sizeBytes;
-    bool                    markerBit;
-
-    FrameType               frameType;
-    VideoCodecType          codec;
-
-    bool isFirstPacket;                 // Is this first packet in a frame.
-    VCMNaluCompleteness completeNALU;   // Default is kNaluIncomplete.
-    bool insertStartCode;               // True if a start code should be inserted before this
-                                        // packet.
-    int width;
-    int height;
-    RTPVideoHeader codecSpecificHeader;
-
-protected:
-    void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
-};
-
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
diff --git a/webrtc/modules/video_coding/main/source/rtt_filter.cc b/webrtc/modules/video_coding/main/source/rtt_filter.cc
deleted file mode 100644
index 5742e8fa89..0000000000
--- a/webrtc/modules/video_coding/main/source/rtt_filter.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/rtt_filter.h"
-
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-
-namespace webrtc {
-
-VCMRttFilter::VCMRttFilter()
-    : _filtFactMax(35),
-      _jumpStdDevs(2.5),
-      _driftStdDevs(3.5),
-      _detectThreshold(kMaxDriftJumpCount) {
-    Reset();
-}
-
-VCMRttFilter&
-VCMRttFilter::operator=(const VCMRttFilter& rhs)
-{
-    if (this != &rhs)
-    {
-        _gotNonZeroUpdate = rhs._gotNonZeroUpdate;
-        _avgRtt = rhs._avgRtt;
-        _varRtt = rhs._varRtt;
-        _maxRtt = rhs._maxRtt;
-        _filtFactCount = rhs._filtFactCount;
-        _jumpCount = rhs._jumpCount;
-        _driftCount = rhs._driftCount;
-        memcpy(_jumpBuf, rhs._jumpBuf, sizeof(_jumpBuf));
-        memcpy(_driftBuf, rhs._driftBuf, sizeof(_driftBuf));
-    }
-    return *this;
-}
-
-void
-VCMRttFilter::Reset()
-{
-    _gotNonZeroUpdate = false;
-    _avgRtt = 0;
-    _varRtt = 0;
-    _maxRtt = 0;
-    _filtFactCount = 1;
-    _jumpCount = 0;
-    _driftCount = 0;
-    memset(_jumpBuf, 0, kMaxDriftJumpCount);
-    memset(_driftBuf, 0, kMaxDriftJumpCount);
-}
-
-void
-VCMRttFilter::Update(int64_t rttMs)
-{
-    if (!_gotNonZeroUpdate)
-    {
-        if (rttMs == 0)
-        {
-            return;
-        }
-        _gotNonZeroUpdate = true;
-    }
-
-    // Sanity check
-    if (rttMs > 3000)
-    {
-        rttMs = 3000;
-    }
-
-    double filtFactor = 0;
-    if (_filtFactCount > 1)
-    {
-        filtFactor = static_cast<double>(_filtFactCount - 1) / _filtFactCount;
-    }
-    _filtFactCount++;
-    if (_filtFactCount > _filtFactMax)
-    {
-        // This prevents filtFactor from going above
-        // (_filtFactMax - 1) / _filtFactMax,
-        // e.g., _filtFactMax = 50 => filtFactor = 49/50 = 0.98
-        _filtFactCount = _filtFactMax;
-    }
-    double oldAvg = _avgRtt;
-    double oldVar = _varRtt;
-    _avgRtt = filtFactor * _avgRtt + (1 - filtFactor) * rttMs;
-    _varRtt = filtFactor * _varRtt + (1 - filtFactor) *
-                (rttMs - _avgRtt) * (rttMs - _avgRtt);
-    _maxRtt = VCM_MAX(rttMs, _maxRtt);
-    if (!JumpDetection(rttMs) || !DriftDetection(rttMs))
-    {
-        // In some cases we don't want to update the statistics
-        _avgRtt = oldAvg;
-        _varRtt = oldVar;
-    }
-}
-
-bool
-VCMRttFilter::JumpDetection(int64_t rttMs)
-{
-    double diffFromAvg = _avgRtt - rttMs;
-    if (fabs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt))
-    {
-        int diffSign = (diffFromAvg >= 0) ? 1 : -1;
-        int jumpCountSign = (_jumpCount >= 0) ? 1 : -1;
-        if (diffSign != jumpCountSign)
-        {
-            // Since the signs differ the samples currently
-            // in the buffer is useless as they represent a
-            // jump in a different direction.
-            _jumpCount = 0;
-        }
-        if (abs(_jumpCount) < kMaxDriftJumpCount)
-        {
-            // Update the buffer used for the short time
-            // statistics.
-            // The sign of the diff is used for updating the counter since
-            // we want to use the same buffer for keeping track of when
-            // the RTT jumps down and up.
-            _jumpBuf[abs(_jumpCount)] = rttMs;
-            _jumpCount += diffSign;
-        }
-        if (abs(_jumpCount) >= _detectThreshold)
-        {
-            // Detected an RTT jump
-            ShortRttFilter(_jumpBuf, abs(_jumpCount));
-            _filtFactCount = _detectThreshold + 1;
-            _jumpCount = 0;
-        }
-        else
-        {
-            return false;
-        }
-    }
-    else
-    {
-        _jumpCount = 0;
-    }
-    return true;
-}
-
-bool
-VCMRttFilter::DriftDetection(int64_t rttMs)
-{
-    if (_maxRtt - _avgRtt > _driftStdDevs * sqrt(_varRtt))
-    {
-        if (_driftCount < kMaxDriftJumpCount)
-        {
-            // Update the buffer used for the short time
-            // statistics.
-            _driftBuf[_driftCount] = rttMs;
-            _driftCount++;
-        }
-        if (_driftCount >= _detectThreshold)
-        {
-            // Detected an RTT drift
-            ShortRttFilter(_driftBuf, _driftCount);
-            _filtFactCount = _detectThreshold + 1;
-            _driftCount = 0;
-        }
-    }
-    else
-    {
-        _driftCount = 0;
-    }
-    return true;
-}
-
-void
-VCMRttFilter::ShortRttFilter(int64_t* buf, uint32_t length)
-{
-    if (length == 0)
-    {
-        return;
-    }
-    _maxRtt = 0;
-    _avgRtt = 0;
-    for (uint32_t i=0; i < length; i++)
-    {
-        if (buf[i] > _maxRtt)
-        {
-            _maxRtt = buf[i];
-        }
-        _avgRtt += buf[i];
-    }
-    _avgRtt = _avgRtt / static_cast<double>(length);
-}
-
-int64_t
-VCMRttFilter::RttMs() const
-{
-    return static_cast<int64_t>(_maxRtt + 0.5);
-}
-
-}
diff --git a/webrtc/modules/video_coding/main/source/rtt_filter.h b/webrtc/modules/video_coding/main/source/rtt_filter.h
deleted file mode 100644
index 9e14a1ab39..0000000000
--- a/webrtc/modules/video_coding/main/source/rtt_filter.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class VCMRttFilter
-{
-public:
-    VCMRttFilter();
-
-    VCMRttFilter& operator=(const VCMRttFilter& rhs);
-
-    // Resets the filter.
-    void Reset();
-    // Updates the filter with a new sample.
-    void Update(int64_t rttMs);
-    // A getter function for the current RTT level in ms.
-    int64_t RttMs() const;
-
-private:
-    // The size of the drift and jump memory buffers
-    // and thus also the detection threshold for these
-    // detectors in number of samples.
-    enum { kMaxDriftJumpCount = 5 };
-    // Detects RTT jumps by comparing the difference between
-    // samples and average to the standard deviation.
-    // Returns true if the long time statistics should be updated
-    // and false otherwise
-    bool JumpDetection(int64_t rttMs);
-    // Detects RTT drifts by comparing the difference between
-    // max and average to the standard deviation.
-    // Returns true if the long time statistics should be updated
-    // and false otherwise
-    bool DriftDetection(int64_t rttMs);
-    // Computes the short time average and maximum of the vector buf.
-    void ShortRttFilter(int64_t* buf, uint32_t length);
-
-    bool                  _gotNonZeroUpdate;
-    double                _avgRtt;
-    double                _varRtt;
-    int64_t         _maxRtt;
-    uint32_t        _filtFactCount;
-    const uint32_t  _filtFactMax;
-    const double          _jumpStdDevs;
-    const double          _driftStdDevs;
-    int32_t         _jumpCount;
-    int32_t         _driftCount;
-    const int32_t   _detectThreshold;
-    int64_t         _jumpBuf[kMaxDriftJumpCount];
-    int64_t         _driftBuf[kMaxDriftJumpCount];
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
diff --git a/webrtc/modules/video_coding/main/test/video_source.h b/webrtc/modules/video_coding/main/test/video_source.h
deleted file mode 100644
index 05deb4a39b..0000000000
--- a/webrtc/modules/video_coding/main/test/video_source.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/typedefs.h"
-
-#include <string>
-
-enum VideoSize
-    {
-        kUndefined,
-        kSQCIF,     // 128*96       = 12 288
-        kQQVGA,     // 160*120      = 19 200
-        kQCIF,      // 176*144      = 25 344
-        kCGA,       // 320*200      = 64 000
-        kQVGA,      // 320*240      = 76 800
-        kSIF,       // 352*240      = 84 480
-        kWQVGA,     // 400*240      = 96 000
-        kCIF,       // 352*288      = 101 376
-        kW288p,     // 512*288      = 147 456 (WCIF)
-        k448p,      // 576*448      = 281 088
-        kVGA,       // 640*480      = 307 200
-        k432p,      // 720*432      = 311 040
-        kW432p,     // 768*432      = 331 776
-        k4SIF,      // 704*480      = 337 920
-        kW448p,     // 768*448      = 344 064
-        kNTSC,		// 720*480      = 345 600
-        kFW448p,    // 800*448      = 358 400
-        kWVGA,      // 800*480      = 384 000
-        k4CIF,      // 704*576      = 405 504
-        kSVGA,      // 800*600      = 480 000
-        kW544p,     // 960*544      = 522 240
-        kW576p,     // 1024*576     = 589 824 (W4CIF)
-        kHD,        // 960*720      = 691 200
-        kXGA,       // 1024*768     = 786 432
-        kWHD,       // 1280*720     = 921 600
-        kFullHD,   // 1440*1080    = 1 555 200
-        kWFullHD,  // 1920*1080    = 2 073 600
-
-        kNumberOfVideoSizes
-    };
-
-
-class VideoSource
-{
-public:
-  VideoSource();
-  VideoSource(std::string fileName, VideoSize size, float frameRate, webrtc::VideoType type = webrtc::kI420);
-  VideoSource(std::string fileName, uint16_t width, uint16_t height,
-      float frameRate = 30, webrtc::VideoType type = webrtc::kI420);
-
-    std::string GetFileName() const { return _fileName; }
-    uint16_t  GetWidth() const { return _width; }
-    uint16_t GetHeight() const { return _height; }
-    webrtc::VideoType GetType() const { return _type; }
-    float GetFrameRate() const { return _frameRate; }
-    int GetWidthHeight( VideoSize size);
-
-    // Returns the filename with the path (including the leading slash) removed.
-    std::string GetName() const;
-
-    size_t GetFrameLength() const;
-
-private:
-    std::string         _fileName;
-    uint16_t      _width;
-    uint16_t      _height;
-    webrtc::VideoType   _type;
-    float               _frameRate;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
diff --git a/webrtc/modules/video_coding/media_opt_util.cc b/webrtc/modules/video_coding/media_opt_util.cc
new file mode 100644
index 0000000000..d57e9c8dd2
--- /dev/null
+++ b/webrtc/modules/video_coding/media_opt_util.cc
@@ -0,0 +1,682 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/media_opt_util.h"
+
+#include <float.h>
+#include <limits.h>
+#include <math.h>
+
+#include <algorithm>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/fec_tables_xor.h"
+#include "webrtc/modules/video_coding/nack_fec_tables.h"
+
+namespace webrtc {
+// Max value of loss rates in off-line model
+static const int kPacketLossMax = 129;
+
+namespace media_optimization {
+
+VCMProtectionMethod::VCMProtectionMethod()
+    : _effectivePacketLoss(0),
+      _protectionFactorK(0),
+      _protectionFactorD(0),
+      _scaleProtKey(2.0f),
+      _maxPayloadSize(1460),
+      _qmRobustness(new VCMQmRobustness()),
+      _useUepProtectionK(false),
+      _useUepProtectionD(true),
+      _corrFecCost(1.0),
+      _type(kNone) {}
+
+VCMProtectionMethod::~VCMProtectionMethod() {
+  delete _qmRobustness;
+}
+void VCMProtectionMethod::UpdateContentMetrics(
+    const VideoContentMetrics* contentMetrics) {
+  _qmRobustness->UpdateContent(contentMetrics);
+}
+
+VCMNackFecMethod::VCMNackFecMethod(int64_t lowRttNackThresholdMs,
+                                   int64_t highRttNackThresholdMs)
+    : VCMFecMethod(),
+      _lowRttNackMs(lowRttNackThresholdMs),
+      _highRttNackMs(highRttNackThresholdMs),
+      _maxFramesFec(1) {
+  assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
+  assert(highRttNackThresholdMs == -1 ||
+         lowRttNackThresholdMs <= highRttNackThresholdMs);
+  assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
+  _type = kNackFec;
+}
+
+VCMNackFecMethod::~VCMNackFecMethod() {
+  //
+}
+bool VCMNackFecMethod::ProtectionFactor(
+    const VCMProtectionParameters* parameters) {
+  // Hybrid Nack FEC has three operational modes:
+  // 1. Low RTT (below kLowRttNackMs) - Nack only: Set FEC rate
+  //    (_protectionFactorD) to zero. -1 means no FEC.
+  // 2. High RTT (above _highRttNackMs) - FEC Only: Keep FEC factors.
+  //    -1 means always allow NACK.
+  // 3. Medium RTT values - Hybrid mode: We will only nack the
+  //    residual following the decoding of the FEC (refer to JB logic). FEC
+  //    delta protection factor will be adjusted based on the RTT.
+
+  // Otherwise: we count on FEC; if the RTT is below a threshold, then we
+  // nack the residual, based on a decision made in the JB.
+
+  // Compute the protection factors
+  VCMFecMethod::ProtectionFactor(parameters);
+  if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs) {
+    _protectionFactorD = 0;
+    VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+
+    // When in Hybrid mode (RTT range), adjust FEC rates based on the
+    // RTT (NACK effectiveness) - adjustment factor is in the range [0,1].
+  } else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs) {
+    // TODO(mikhal): Disabling adjustment temporarily.
+    // uint16_t rttIndex = (uint16_t) parameters->rtt;
+    float adjustRtt = 1.0f;  // (float)VCMNackFecTable[rttIndex] / 100.0f;
+
+    // Adjust FEC with NACK on (for delta frame only)
+    // table depends on RTT relative to rttMax (NACK Threshold)
+    _protectionFactorD = static_cast<uint8_t>(
+        adjustRtt * static_cast<float>(_protectionFactorD));
+    // update FEC rates after applying adjustment
+    VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+  }
+
+  return true;
+}
+
+int VCMNackFecMethod::ComputeMaxFramesFec(
+    const VCMProtectionParameters* parameters) {
+  if (parameters->numLayers > 2) {
+    // For more than 2 temporal layers we will only have FEC on the base layer,
+    // and the base layers will be pretty far apart. Therefore we force one
+    // frame FEC.
+    return 1;
+  }
+  // We set the max number of frames to base the FEC on so that on average
+  // we will have complete frames in one RTT. Note that this is an upper
+  // bound, and that the actual number of frames used for FEC is decided by the
+  // RTP module based on the actual number of packets and the protection factor.
+  float base_layer_framerate =
+      parameters->frameRate /
+      static_cast<float>(1 << (parameters->numLayers - 1));
+  int max_frames_fec = std::max(
+      static_cast<int>(2.0f * base_layer_framerate * parameters->rtt / 1000.0f +
+                       0.5f),
+      1);
+  // |kUpperLimitFramesFec| is the upper limit on how many frames we
+  // allow any FEC to be based on.
+  if (max_frames_fec > kUpperLimitFramesFec) {
+    max_frames_fec = kUpperLimitFramesFec;
+  }
+  return max_frames_fec;
+}
+
+int VCMNackFecMethod::MaxFramesFec() const {
+  return _maxFramesFec;
+}
+
+bool VCMNackFecMethod::BitRateTooLowForFec(
+    const VCMProtectionParameters* parameters) {
+  // Bitrate below which we turn off FEC, regardless of reported packet loss.
+  // The condition should depend on resolution and content. For now, use
+  // threshold on bytes per frame, with some effect for the frame size.
+  // The condition for turning off FEC is also based on other factors,
+  // such as |_numLayers|, |_maxFramesFec|, and |_rtt|.
+  int estimate_bytes_per_frame = 1000 * BitsPerFrame(parameters) / 8;
+  int max_bytes_per_frame = kMaxBytesPerFrameForFec;
+  int num_pixels = parameters->codecWidth * parameters->codecHeight;
+  if (num_pixels <= 352 * 288) {
+    max_bytes_per_frame = kMaxBytesPerFrameForFecLow;
+  } else if (num_pixels > 640 * 480) {
+    max_bytes_per_frame = kMaxBytesPerFrameForFecHigh;
+  }
+  // TODO(marpan): add condition based on maximum frames used for FEC,
+  // and expand condition based on frame size.
+  // Max round trip time threshold in ms.
+  const int64_t kMaxRttTurnOffFec = 200;
+  if (estimate_bytes_per_frame < max_bytes_per_frame &&
+      parameters->numLayers < 3 && parameters->rtt < kMaxRttTurnOffFec) {
+    return true;
+  }
+  return false;
+}
+
+bool VCMNackFecMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameters) {
+  // Set the effective packet loss for encoder (based on FEC code).
+  // Compute the effective packet loss and residual packet loss due to FEC.
+  VCMFecMethod::EffectivePacketLoss(parameters);
+  return true;
+}
+
+bool VCMNackFecMethod::UpdateParameters(
+    const VCMProtectionParameters* parameters) {
+  ProtectionFactor(parameters);
+  EffectivePacketLoss(parameters);
+  _maxFramesFec = ComputeMaxFramesFec(parameters);
+  if (BitRateTooLowForFec(parameters)) {
+    _protectionFactorK = 0;
+    _protectionFactorD = 0;
+  }
+
+  // Protection/fec rates obtained above are defined relative to total number
+  // of packets (total rate: source + fec) FEC in RTP module assumes
+  // protection factor is defined relative to source number of packets so we
+  // should convert the factor to reduce mismatch between mediaOpt's rate and
+  // the actual one
+  _protectionFactorK = VCMFecMethod::ConvertFECRate(_protectionFactorK);
+  _protectionFactorD = VCMFecMethod::ConvertFECRate(_protectionFactorD);
+
+  return true;
+}
+
+VCMNackMethod::VCMNackMethod() : VCMProtectionMethod() {
+  _type = kNack;
+}
+
+VCMNackMethod::~VCMNackMethod() {
+  //
+}
+
+bool VCMNackMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameter) {
+  // Effective Packet Loss, NA in current version.
+  _effectivePacketLoss = 0;
+  return true;
+}
+
+bool VCMNackMethod::UpdateParameters(
+    const VCMProtectionParameters* parameters) {
+  // Compute the effective packet loss
+  EffectivePacketLoss(parameters);
+
+  // nackCost  = (bitRate - nackCost) * (lossPr)
+  return true;
+}
+
+VCMFecMethod::VCMFecMethod() : VCMProtectionMethod() {
+  _type = kFec;
+}
+VCMFecMethod::~VCMFecMethod() {
+  //
+}
+
+uint8_t VCMFecMethod::BoostCodeRateKey(uint8_t packetFrameDelta,
+                                       uint8_t packetFrameKey) const {
+  uint8_t boostRateKey = 2;
+  // Default: ratio scales the FEC protection up for I frames
+  uint8_t ratio = 1;
+
+  if (packetFrameDelta > 0) {
+    ratio = (int8_t)(packetFrameKey / packetFrameDelta);
+  }
+  ratio = VCM_MAX(boostRateKey, ratio);
+
+  return ratio;
+}
+
+uint8_t VCMFecMethod::ConvertFECRate(uint8_t codeRateRTP) const {
+  return static_cast<uint8_t>(VCM_MIN(
+      255,
+      (0.5 + 255.0 * codeRateRTP / static_cast<float>(255 - codeRateRTP))));
+}
+
+// Update FEC with protectionFactorD
+void VCMFecMethod::UpdateProtectionFactorD(uint8_t protectionFactorD) {
+  _protectionFactorD = protectionFactorD;
+}
+
+// Update FEC with protectionFactorK
+void VCMFecMethod::UpdateProtectionFactorK(uint8_t protectionFactorK) {
+  _protectionFactorK = protectionFactorK;
+}
+
+bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
+  // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
+
+  // No protection if (filtered) packetLoss is 0
+  uint8_t packetLoss = (uint8_t)(255 * parameters->lossPr);
+  if (packetLoss == 0) {
+    _protectionFactorK = 0;
+    _protectionFactorD = 0;
+    return true;
+  }
+
+  // Parameters for FEC setting:
+  // first partition size, thresholds, table pars, spatial resoln fac.
+
+  // First partition protection: ~ 20%
+  uint8_t firstPartitionProt = (uint8_t)(255 * 0.20);
+
+  // Minimum protection level needed to generate one FEC packet for one
+  // source packet/frame (in RTP sender)
+  uint8_t minProtLevelFec = 85;
+
+  // Threshold on packetLoss and bitRrate/frameRate (=average #packets),
+  // above which we allocate protection to cover at least first partition.
+  uint8_t lossThr = 0;
+  uint8_t packetNumThr = 1;
+
+  // Parameters for range of rate index of table.
+  const uint8_t ratePar1 = 5;
+  const uint8_t ratePar2 = 49;
+
+  // Spatial resolution size, relative to a reference size.
+  float spatialSizeToRef =
+      static_cast<float>(parameters->codecWidth * parameters->codecHeight) /
+      (static_cast<float>(704 * 576));
+  // resolnFac: This parameter will generally increase/decrease the FEC rate
+  // (for fixed bitRate and packetLoss) based on system size.
+  // Use a smaller exponent (< 1) to control/soften system size effect.
+  const float resolnFac = 1.0 / powf(spatialSizeToRef, 0.3f);
+
+  const int bitRatePerFrame = BitsPerFrame(parameters);
+
+  // Average number of packets per frame (source and fec):
+  const uint8_t avgTotPackets =
+      1 + (uint8_t)(static_cast<float>(bitRatePerFrame) * 1000.0 /
+                        static_cast<float>(8.0 * _maxPayloadSize) +
+                    0.5);
+
+  // FEC rate parameters: for P and I frame
+  uint8_t codeRateDelta = 0;
+  uint8_t codeRateKey = 0;
+
+  // Get index for table: the FEC protection depends on an effective rate.
+  // The range on the rate index corresponds to rates (bps)
+  // from ~200k to ~8000k, for 30fps
+  const uint16_t effRateFecTable =
+      static_cast<uint16_t>(resolnFac * bitRatePerFrame);
+  uint8_t rateIndexTable = (uint8_t)VCM_MAX(
+      VCM_MIN((effRateFecTable - ratePar1) / ratePar1, ratePar2), 0);
+
+  // Restrict packet loss range to 50:
+  // current tables defined only up to 50%
+  if (packetLoss >= kPacketLossMax) {
+    packetLoss = kPacketLossMax - 1;
+  }
+  uint16_t indexTable = rateIndexTable * kPacketLossMax + packetLoss;
+
+  // Check on table index
+  assert(indexTable < kSizeCodeRateXORTable);
+
+  // Protection factor for P frame
+  codeRateDelta = kCodeRateXORTable[indexTable];
+
+  if (packetLoss > lossThr && avgTotPackets > packetNumThr) {
+    // Set a minimum based on first partition size.
+    if (codeRateDelta < firstPartitionProt) {
+      codeRateDelta = firstPartitionProt;
+    }
+  }
+
+  // Check limit on amount of protection for P frame; 50% is max.
+  if (codeRateDelta >= kPacketLossMax) {
+    codeRateDelta = kPacketLossMax - 1;
+  }
+
+  float adjustFec = 1.0f;
+  // Avoid additional adjustments when layers are active.
+  // TODO(mikhal/marco): Update adjusmtent based on layer info.
+  if (parameters->numLayers == 1) {
+    adjustFec = _qmRobustness->AdjustFecFactor(
+        codeRateDelta, parameters->bitRate, parameters->frameRate,
+        parameters->rtt, packetLoss);
+  }
+
+  codeRateDelta = static_cast<uint8_t>(codeRateDelta * adjustFec);
+
+  // For Key frame:
+  // Effectively at a higher rate, so we scale/boost the rate
+  // The boost factor may depend on several factors: ratio of packet
+  // number of I to P frames, how much protection placed on P frames, etc.
+  const uint8_t packetFrameDelta = (uint8_t)(0.5 + parameters->packetsPerFrame);
+  const uint8_t packetFrameKey =
+      (uint8_t)(0.5 + parameters->packetsPerFrameKey);
+  const uint8_t boostKey = BoostCodeRateKey(packetFrameDelta, packetFrameKey);
+
+  rateIndexTable = (uint8_t)VCM_MAX(
+      VCM_MIN(1 + (boostKey * effRateFecTable - ratePar1) / ratePar1, ratePar2),
+      0);
+  uint16_t indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
+
+  indexTableKey = VCM_MIN(indexTableKey, kSizeCodeRateXORTable);
+
+  // Check on table index
+  assert(indexTableKey < kSizeCodeRateXORTable);
+
+  // Protection factor for I frame
+  codeRateKey = kCodeRateXORTable[indexTableKey];
+
+  // Boosting for Key frame.
+  int boostKeyProt = _scaleProtKey * codeRateDelta;
+  if (boostKeyProt >= kPacketLossMax) {
+    boostKeyProt = kPacketLossMax - 1;
+  }
+
+  // Make sure I frame protection is at least larger than P frame protection,
+  // and at least as high as filtered packet loss.
+  codeRateKey = static_cast<uint8_t>(
+      VCM_MAX(packetLoss, VCM_MAX(boostKeyProt, codeRateKey)));
+
+  // Check limit on amount of protection for I frame: 50% is max.
+  if (codeRateKey >= kPacketLossMax) {
+    codeRateKey = kPacketLossMax - 1;
+  }
+
+  _protectionFactorK = codeRateKey;
+  _protectionFactorD = codeRateDelta;
+
+  // Generally there is a rate mis-match between the FEC cost estimated
+  // in mediaOpt and the actual FEC cost sent out in RTP module.
+  // This is more significant at low rates (small # of source packets), where
+  // the granularity of the FEC decreases. In this case, non-zero protection
+  // in mediaOpt may generate 0 FEC packets in RTP sender (since actual #FEC
+  // is based on rounding off protectionFactor on actual source packet number).
+  // The correction factor (_corrFecCost) attempts to corrects this, at least
+  // for cases of low rates (small #packets) and low protection levels.
+
+  float numPacketsFl = 1.0f + (static_cast<float>(bitRatePerFrame) * 1000.0 /
+                                   static_cast<float>(8.0 * _maxPayloadSize) +
+                               0.5);
+
+  const float estNumFecGen =
+      0.5f + static_cast<float>(_protectionFactorD * numPacketsFl / 255.0f);
+
+  // We reduce cost factor (which will reduce overhead for FEC and
+  // hybrid method) and not the protectionFactor.
+  _corrFecCost = 1.0f;
+  if (estNumFecGen < 1.1f && _protectionFactorD < minProtLevelFec) {
+    _corrFecCost = 0.5f;
+  }
+  if (estNumFecGen < 0.9f && _protectionFactorD < minProtLevelFec) {
+    _corrFecCost = 0.0f;
+  }
+
+  // TODO(marpan): Set the UEP protection on/off for Key and Delta frames
+  _useUepProtectionK = _qmRobustness->SetUepProtection(
+      codeRateKey, parameters->bitRate, packetLoss, 0);
+
+  _useUepProtectionD = _qmRobustness->SetUepProtection(
+      codeRateDelta, parameters->bitRate, packetLoss, 1);
+
+  // DONE WITH FEC PROTECTION SETTINGS
+  return true;
+}
+
+int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
+  // When temporal layers are available FEC will only be applied on the base
+  // layer.
+  const float bitRateRatio =
+      kVp8LayerRateAlloction[parameters->numLayers - 1][0];
+  float frameRateRatio = powf(1 / 2.0, parameters->numLayers - 1);
+  float bitRate = parameters->bitRate * bitRateRatio;
+  float frameRate = parameters->frameRate * frameRateRatio;
+
+  // TODO(mikhal): Update factor following testing.
+  float adjustmentFactor = 1;
+
+  // Average bits per frame (units of kbits)
+  return static_cast<int>(adjustmentFactor * bitRate / frameRate);
+}
+
+bool VCMFecMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameters) {
+  // Effective packet loss to encoder is based on RPL (residual packet loss)
+  // this is a soft setting based on degree of FEC protection
+  // RPL = received/input packet loss - average_FEC_recovery
+  // note: received/input packet loss may be filtered based on FilteredLoss
+
+  // Effective Packet Loss, NA in current version.
+  _effectivePacketLoss = 0;
+
+  return true;
+}
+
+bool VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters) {
+  // Compute the protection factor
+  ProtectionFactor(parameters);
+
+  // Compute the effective packet loss
+  EffectivePacketLoss(parameters);
+
+  // Protection/fec rates obtained above is defined relative to total number
+  // of packets (total rate: source+fec) FEC in RTP module assumes protection
+  // factor is defined relative to source number of packets so we should
+  // convert the factor to reduce mismatch between mediaOpt suggested rate and
+  // the actual rate
+  _protectionFactorK = ConvertFECRate(_protectionFactorK);
+  _protectionFactorD = ConvertFECRate(_protectionFactorD);
+
+  return true;
+}
+VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs)
+    : _currentParameters(),
+      _rtt(0),
+      _lossPr(0.0f),
+      _bitRate(0.0f),
+      _frameRate(0.0f),
+      _keyFrameSize(0.0f),
+      _fecRateKey(0),
+      _fecRateDelta(0),
+      _lastPrUpdateT(0),
+      _lossPr255(0.9999f),
+      _lossPrHistory(),
+      _shortMaxLossPr255(0),
+      _packetsPerFrame(0.9999f),
+      _packetsPerFrameKey(0.9999f),
+      _codecWidth(0),
+      _codecHeight(0),
+      _numLayers(1) {
+  Reset(nowMs);
+}
+
+VCMLossProtectionLogic::~VCMLossProtectionLogic() {
+  Release();
+}
+
+void VCMLossProtectionLogic::SetMethod(
+    enum VCMProtectionMethodEnum newMethodType) {
+  if (_selectedMethod && _selectedMethod->Type() == newMethodType)
+    return;
+
+  switch (newMethodType) {
+    case kNack:
+      _selectedMethod.reset(new VCMNackMethod());
+      break;
+    case kFec:
+      _selectedMethod.reset(new VCMFecMethod());
+      break;
+    case kNackFec:
+      _selectedMethod.reset(new VCMNackFecMethod(kLowRttNackMs, -1));
+      break;
+    case kNone:
+      _selectedMethod.reset();
+      break;
+  }
+  UpdateMethod();
+}
+
+void VCMLossProtectionLogic::UpdateRtt(int64_t rtt) {
+  _rtt = rtt;
+}
+
+void VCMLossProtectionLogic::UpdateMaxLossHistory(uint8_t lossPr255,
+                                                  int64_t now) {
+  if (_lossPrHistory[0].timeMs >= 0 &&
+      now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs) {
+    if (lossPr255 > _shortMaxLossPr255) {
+      _shortMaxLossPr255 = lossPr255;
+    }
+  } else {
+    // Only add a new value to the history once a second
+    if (_lossPrHistory[0].timeMs == -1) {
+      // First, no shift
+      _shortMaxLossPr255 = lossPr255;
+    } else {
+      // Shift
+      for (int32_t i = (kLossPrHistorySize - 2); i >= 0; i--) {
+        _lossPrHistory[i + 1].lossPr255 = _lossPrHistory[i].lossPr255;
+        _lossPrHistory[i + 1].timeMs = _lossPrHistory[i].timeMs;
+      }
+    }
+    if (_shortMaxLossPr255 == 0) {
+      _shortMaxLossPr255 = lossPr255;
+    }
+
+    _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
+    _lossPrHistory[0].timeMs = now;
+    _shortMaxLossPr255 = 0;
+  }
+}
+
+uint8_t VCMLossProtectionLogic::MaxFilteredLossPr(int64_t nowMs) const {
+  uint8_t maxFound = _shortMaxLossPr255;
+  if (_lossPrHistory[0].timeMs == -1) {
+    return maxFound;
+  }
+  for (int32_t i = 0; i < kLossPrHistorySize; i++) {
+    if (_lossPrHistory[i].timeMs == -1) {
+      break;
+    }
+    if (nowMs - _lossPrHistory[i].timeMs >
+        kLossPrHistorySize * kLossPrShortFilterWinMs) {
+      // This sample (and all samples after this) is too old
+      break;
+    }
+    if (_lossPrHistory[i].lossPr255 > maxFound) {
+      // This sample is the largest one this far into the history
+      maxFound = _lossPrHistory[i].lossPr255;
+    }
+  }
+  return maxFound;
+}
+
+uint8_t VCMLossProtectionLogic::FilteredLoss(int64_t nowMs,
+                                             FilterPacketLossMode filter_mode,
+                                             uint8_t lossPr255) {
+  // Update the max window filter.
+  UpdateMaxLossHistory(lossPr255, nowMs);
+
+  // Update the recursive average filter.
+  _lossPr255.Apply(static_cast<float>(nowMs - _lastPrUpdateT),
+                   static_cast<float>(lossPr255));
+  _lastPrUpdateT = nowMs;
+
+  // Filtered loss: default is received loss (no filtering).
+  uint8_t filtered_loss = lossPr255;
+
+  switch (filter_mode) {
+    case kNoFilter:
+      break;
+    case kAvgFilter:
+      filtered_loss = static_cast<uint8_t>(_lossPr255.filtered() + 0.5);
+      break;
+    case kMaxFilter:
+      filtered_loss = MaxFilteredLossPr(nowMs);
+      break;
+  }
+
+  return filtered_loss;
+}
+
+void VCMLossProtectionLogic::UpdateFilteredLossPr(uint8_t packetLossEnc) {
+  _lossPr = static_cast<float>(packetLossEnc) / 255.0;
+}
+
+void VCMLossProtectionLogic::UpdateBitRate(float bitRate) {
+  _bitRate = bitRate;
+}
+
+void VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets,
+                                                   int64_t nowMs) {
+  _packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
+                         nPackets);
+  _lastPacketPerFrameUpdateT = nowMs;
+}
+
+void VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets,
+                                                      int64_t nowMs) {
+  _packetsPerFrameKey.Apply(
+      static_cast<float>(nowMs - _lastPacketPerFrameUpdateTKey), nPackets);
+  _lastPacketPerFrameUpdateTKey = nowMs;
+}
+
+void VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize) {
+  _keyFrameSize = keyFrameSize;
+}
+
+void VCMLossProtectionLogic::UpdateFrameSize(uint16_t width, uint16_t height) {
+  _codecWidth = width;
+  _codecHeight = height;
+}
+
+void VCMLossProtectionLogic::UpdateNumLayers(int numLayers) {
+  _numLayers = (numLayers == 0) ? 1 : numLayers;
+}
+
+bool VCMLossProtectionLogic::UpdateMethod() {
+  if (!_selectedMethod)
+    return false;
+  _currentParameters.rtt = _rtt;
+  _currentParameters.lossPr = _lossPr;
+  _currentParameters.bitRate = _bitRate;
+  _currentParameters.frameRate = _frameRate;  // rename actual frame rate?
+  _currentParameters.keyFrameSize = _keyFrameSize;
+  _currentParameters.fecRateDelta = _fecRateDelta;
+  _currentParameters.fecRateKey = _fecRateKey;
+  _currentParameters.packetsPerFrame = _packetsPerFrame.filtered();
+  _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.filtered();
+  _currentParameters.codecWidth = _codecWidth;
+  _currentParameters.codecHeight = _codecHeight;
+  _currentParameters.numLayers = _numLayers;
+  return _selectedMethod->UpdateParameters(&_currentParameters);
+}
+
+VCMProtectionMethod* VCMLossProtectionLogic::SelectedMethod() const {
+  return _selectedMethod.get();
+}
+
+VCMProtectionMethodEnum VCMLossProtectionLogic::SelectedType() const {
+  return _selectedMethod ? _selectedMethod->Type() : kNone;
+}
+
+void VCMLossProtectionLogic::Reset(int64_t nowMs) {
+  _lastPrUpdateT = nowMs;
+  _lastPacketPerFrameUpdateT = nowMs;
+  _lastPacketPerFrameUpdateTKey = nowMs;
+  _lossPr255.Reset(0.9999f);
+  _packetsPerFrame.Reset(0.9999f);
+  _fecRateDelta = _fecRateKey = 0;
+  for (int32_t i = 0; i < kLossPrHistorySize; i++) {
+    _lossPrHistory[i].lossPr255 = 0;
+    _lossPrHistory[i].timeMs = -1;
+  }
+  _shortMaxLossPr255 = 0;
+  Release();
+}
+
+void VCMLossProtectionLogic::Release() {
+  _selectedMethod.reset();
+}
+
+}  // namespace media_optimization
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/media_opt_util.h b/webrtc/modules/video_coding/media_opt_util.h
new file mode 100644
index 0000000000..a016a03eab
--- /dev/null
+++ b/webrtc/modules/video_coding/media_opt_util.h
@@ -0,0 +1,361 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "webrtc/base/exp_filter.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/qm_select.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace media_optimization {
+
+// Number of time periods used for (max) window filter for packet loss
+// TODO(marpan): set reasonable window size for filtered packet loss,
+// adjustment should be based on logged/real data of loss stats/correlation.
+enum { kLossPrHistorySize = 10 };
+
+// 1000 ms, total filter length is (kLossPrHistorySize * 1000) ms
+enum { kLossPrShortFilterWinMs = 1000 };
+
+// The type of filter used on the received packet loss reports.
+enum FilterPacketLossMode {
+  kNoFilter,   // No filtering on received loss.
+  kAvgFilter,  // Recursive average filter.
+  kMaxFilter   // Max-window filter, over the time interval of:
+               // (kLossPrHistorySize * kLossPrShortFilterWinMs) ms.
+};
+
+// Thresholds for hybrid NACK/FEC
+// common to media optimization and the jitter buffer.
+const int64_t kLowRttNackMs = 20;
+
+struct VCMProtectionParameters {
+  VCMProtectionParameters()
+      : rtt(0),
+        lossPr(0.0f),
+        bitRate(0.0f),
+        packetsPerFrame(0.0f),
+        packetsPerFrameKey(0.0f),
+        frameRate(0.0f),
+        keyFrameSize(0.0f),
+        fecRateDelta(0),
+        fecRateKey(0),
+        codecWidth(0),
+        codecHeight(0),
+        numLayers(1) {}
+
+  int64_t rtt;
+  float lossPr;
+  float bitRate;
+  float packetsPerFrame;
+  float packetsPerFrameKey;
+  float frameRate;
+  float keyFrameSize;
+  uint8_t fecRateDelta;
+  uint8_t fecRateKey;
+  uint16_t codecWidth;
+  uint16_t codecHeight;
+  int numLayers;
+};
+
+/******************************/
+/* VCMProtectionMethod class  */
+/******************************/
+
+enum VCMProtectionMethodEnum { kNack, kFec, kNackFec, kNone };
+
+class VCMLossProbabilitySample {
+ public:
+  VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {}
+
+  uint8_t lossPr255;
+  int64_t timeMs;
+};
+
+class VCMProtectionMethod {
+ public:
+  VCMProtectionMethod();
+  virtual ~VCMProtectionMethod();
+
+  // Updates the efficiency of the method using the parameters provided
+  //
+  // Input:
+  //         - parameters         : Parameters used to calculate efficiency
+  //
+  // Return value                 : True if this method is recommended in
+  //                                the given conditions.
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
+
+  // Returns the protection type
+  //
+  // Return value                 : The protection type
+  enum VCMProtectionMethodEnum Type() const { return _type; }
+
+  // Returns the effective packet loss for ER, required by this protection
+  // method
+  //
+  // Return value                 : Required effective packet loss
+  virtual uint8_t RequiredPacketLossER() { return _effectivePacketLoss; }
+
+  // Extracts the FEC protection factor for Key frame, required by this
+  // protection method
+  //
+  // Return value                 : Required protectionFactor for Key frame
+  virtual uint8_t RequiredProtectionFactorK() { return _protectionFactorK; }
+
+  // Extracts the FEC protection factor for Delta frame, required by this
+  // protection method
+  //
+  // Return value                 : Required protectionFactor for delta frame
+  virtual uint8_t RequiredProtectionFactorD() { return _protectionFactorD; }
+
+  // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
+  //
+  // Return value                 : Required Unequal protection on/off state.
+  virtual bool RequiredUepProtectionK() { return _useUepProtectionK; }
+
+  // Extracts whether the the FEC Unequal protection (UEP) is used for Delta
+  // frame.
+  //
+  // Return value                 : Required Unequal protection on/off state.
+  virtual bool RequiredUepProtectionD() { return _useUepProtectionD; }
+
+  virtual int MaxFramesFec() const { return 1; }
+
+  // Updates content metrics
+  void UpdateContentMetrics(const VideoContentMetrics* contentMetrics);
+
+ protected:
+  uint8_t _effectivePacketLoss;
+  uint8_t _protectionFactorK;
+  uint8_t _protectionFactorD;
+  // Estimation of residual loss after the FEC
+  float _scaleProtKey;
+  int32_t _maxPayloadSize;
+
+  VCMQmRobustness* _qmRobustness;
+  bool _useUepProtectionK;
+  bool _useUepProtectionD;
+  float _corrFecCost;
+  enum VCMProtectionMethodEnum _type;
+};
+
+class VCMNackMethod : public VCMProtectionMethod {
+ public:
+  VCMNackMethod();
+  virtual ~VCMNackMethod();
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+  // Get the effective packet loss
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
+};
+
+class VCMFecMethod : public VCMProtectionMethod {
+ public:
+  VCMFecMethod();
+  virtual ~VCMFecMethod();
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+  // Get the effective packet loss for ER
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+  // Get the FEC protection factors
+  bool ProtectionFactor(const VCMProtectionParameters* parameters);
+  // Get the boost for key frame protection
+  uint8_t BoostCodeRateKey(uint8_t packetFrameDelta,
+                           uint8_t packetFrameKey) const;
+  // Convert the rates: defined relative to total# packets or source# packets
+  uint8_t ConvertFECRate(uint8_t codeRate) const;
+  // Get the average effective recovery from FEC: for random loss model
+  float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
+  // Update FEC with protectionFactorD
+  void UpdateProtectionFactorD(uint8_t protectionFactorD);
+  // Update FEC with protectionFactorK
+  void UpdateProtectionFactorK(uint8_t protectionFactorK);
+  // Compute the bits per frame. Account for temporal layers when applicable.
+  int BitsPerFrame(const VCMProtectionParameters* parameters);
+
+ protected:
+  enum { kUpperLimitFramesFec = 6 };
+  // Thresholds values for the bytes/frame and round trip time, below which we
+  // may turn off FEC, depending on |_numLayers| and |_maxFramesFec|.
+  // Max bytes/frame for VGA, corresponds to ~140k at 25fps.
+  enum { kMaxBytesPerFrameForFec = 700 };
+  // Max bytes/frame for CIF and lower: corresponds to ~80k at 25fps.
+  enum { kMaxBytesPerFrameForFecLow = 400 };
+  // Max bytes/frame for frame size larger than VGA, ~200k at 25fps.
+  enum { kMaxBytesPerFrameForFecHigh = 1000 };
+};
+
+class VCMNackFecMethod : public VCMFecMethod {
+ public:
+  VCMNackFecMethod(int64_t lowRttNackThresholdMs,
+                   int64_t highRttNackThresholdMs);
+  virtual ~VCMNackFecMethod();
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+  // Get the effective packet loss for ER
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+  // Get the protection factors
+  bool ProtectionFactor(const VCMProtectionParameters* parameters);
+  // Get the max number of frames the FEC is allowed to be based on.
+  int MaxFramesFec() const;
+  // Turn off the FEC based on low bitrate and other factors.
+  bool BitRateTooLowForFec(const VCMProtectionParameters* parameters);
+
+ private:
+  int ComputeMaxFramesFec(const VCMProtectionParameters* parameters);
+
+  int64_t _lowRttNackMs;
+  int64_t _highRttNackMs;
+  int _maxFramesFec;
+};
+
+class VCMLossProtectionLogic {
+ public:
+  explicit VCMLossProtectionLogic(int64_t nowMs);
+  ~VCMLossProtectionLogic();
+
+  // Set the protection method to be used
+  //
+  // Input:
+  //        - newMethodType    : New requested protection method type. If one
+  //                           is already set, it will be deleted and replaced
+  void SetMethod(VCMProtectionMethodEnum newMethodType);
+
+  // Update the round-trip time
+  //
+  // Input:
+  //          - rtt           : Round-trip time in seconds.
+  void UpdateRtt(int64_t rtt);
+
+  // Update the filtered packet loss.
+  //
+  // Input:
+  //          - packetLossEnc :  The reported packet loss filtered
+  //                             (max window or average)
+  void UpdateFilteredLossPr(uint8_t packetLossEnc);
+
+  // Update the current target bit rate.
+  //
+  // Input:
+  //          - bitRate          : The current target bit rate in kbits/s
+  void UpdateBitRate(float bitRate);
+
+  // Update the number of packets per frame estimate, for delta frames
+  //
+  // Input:
+  //          - nPackets         : Number of packets in the latest sent frame.
+  void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
+
+  // Update the number of packets per frame estimate, for key frames
+  //
+  // Input:
+  //          - nPackets         : umber of packets in the latest sent frame.
+  void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
+
+  // Update the keyFrameSize estimate
+  //
+  // Input:
+  //          - keyFrameSize     : The size of the latest sent key frame.
+  void UpdateKeyFrameSize(float keyFrameSize);
+
+  // Update the frame rate
+  //
+  // Input:
+  //          - frameRate        : The current target frame rate.
+  void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
+
+  // Update the frame size
+  //
+  // Input:
+  //          - width        : The codec frame width.
+  //          - height       : The codec frame height.
+  void UpdateFrameSize(uint16_t width, uint16_t height);
+
+  // Update the number of active layers
+  //
+  // Input:
+  //          - numLayers    : Number of layers used.
+  void UpdateNumLayers(int numLayers);
+
+  // The amount of packet loss to cover for with FEC.
+  //
+  // Input:
+  //          - fecRateKey      : Packet loss to cover for with FEC when
+  //                              sending key frames.
+  //          - fecRateDelta    : Packet loss to cover for with FEC when
+  //                              sending delta frames.
+  void UpdateFECRates(uint8_t fecRateKey, uint8_t fecRateDelta) {
+    _fecRateKey = fecRateKey;
+    _fecRateDelta = fecRateDelta;
+  }
+
+  // Update the protection methods with the current VCMProtectionParameters
+  // and set the requested protection settings.
+  // Return value     : Returns true on update
+  bool UpdateMethod();
+
+  // Returns the method currently selected.
+  //
+  // Return value                 : The protection method currently selected.
+  VCMProtectionMethod* SelectedMethod() const;
+
+  // Return the protection type of the currently selected method
+  VCMProtectionMethodEnum SelectedType() const;
+
+  // Updates the filtered loss for the average and max window packet loss,
+  // and returns the filtered loss probability in the interval [0, 255].
+  // The returned filtered loss value depends on the parameter |filter_mode|.
+  // The input parameter |lossPr255| is the received packet loss.
+
+  // Return value                 : The filtered loss probability
+  uint8_t FilteredLoss(int64_t nowMs,
+                       FilterPacketLossMode filter_mode,
+                       uint8_t lossPr255);
+
+  void Reset(int64_t nowMs);
+
+  void Release();
+
+ private:
+  // Sets the available loss protection methods.
+  void UpdateMaxLossHistory(uint8_t lossPr255, int64_t now);
+  uint8_t MaxFilteredLossPr(int64_t nowMs) const;
+  rtc::scoped_ptr<VCMProtectionMethod> _selectedMethod;
+  VCMProtectionParameters _currentParameters;
+  int64_t _rtt;
+  float _lossPr;
+  float _bitRate;
+  float _frameRate;
+  float _keyFrameSize;
+  uint8_t _fecRateKey;
+  uint8_t _fecRateDelta;
+  int64_t _lastPrUpdateT;
+  int64_t _lastPacketPerFrameUpdateT;
+  int64_t _lastPacketPerFrameUpdateTKey;
+  rtc::ExpFilter _lossPr255;
+  VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize];
+  uint8_t _shortMaxLossPr255;
+  rtc::ExpFilter _packetsPerFrame;
+  rtc::ExpFilter _packetsPerFrameKey;
+  uint16_t _codecWidth;
+  uint16_t _codecHeight;
+  int _numLayers;
+};
+
+}  // namespace media_optimization
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.cc b/webrtc/modules/video_coding/media_optimization.cc
index cc73d3803d..a234a06f9b 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.cc
+++ b/webrtc/modules/video_coding/media_optimization.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
 
-#include "webrtc/modules/video_coding/main/source/content_metrics_processing.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/content_metrics_processing.h"
+#include "webrtc/modules/video_coding/qm_select.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 namespace media_optimization {
@@ -53,11 +53,9 @@ void UpdateProtectionCallback(
   key_fec_params.fec_mask_type = kFecMaskRandom;
 
   // TODO(Marco): Pass FEC protection values per layer.
-  video_protection_callback->ProtectionRequest(&delta_fec_params,
-                                               &key_fec_params,
-                                               video_rate_bps,
-                                               nack_overhead_rate_bps,
-                                               fec_overhead_rate_bps);
+  video_protection_callback->ProtectionRequest(
+      &delta_fec_params, &key_fec_params, video_rate_bps,
+      nack_overhead_rate_bps, fec_overhead_rate_bps);
 }
 }  // namespace
 
@@ -115,8 +113,8 @@ MediaOptimization::~MediaOptimization(void) {
 
 void MediaOptimization::Reset() {
   CriticalSectionScoped lock(crit_sect_.get());
-  SetEncodingDataInternal(
-      kVideoCodecUnknown, 0, 0, 0, 0, 0, 0, max_payload_size_);
+  SetEncodingDataInternal(kVideoCodecUnknown, 0, 0, 0, 0, 0, 0,
+                          max_payload_size_);
   memset(incoming_frame_times_, -1, sizeof(incoming_frame_times_));
   incoming_frame_rate_ = 0.0;
   frame_dropper_->Reset();
@@ -149,14 +147,8 @@ void MediaOptimization::SetEncodingData(VideoCodecType send_codec_type,
                                         int num_layers,
                                         int32_t mtu) {
   CriticalSectionScoped lock(crit_sect_.get());
-  SetEncodingDataInternal(send_codec_type,
-                          max_bit_rate,
-                          frame_rate,
-                          target_bitrate,
-                          width,
-                          height,
-                          num_layers,
-                          mtu);
+  SetEncodingDataInternal(send_codec_type, max_bit_rate, frame_rate,
+                          target_bitrate, width, height, num_layers, mtu);
 }
 
 void MediaOptimization::SetEncodingDataInternal(VideoCodecType send_codec_type,
@@ -190,11 +182,8 @@ void MediaOptimization::SetEncodingDataInternal(VideoCodecType send_codec_type,
   codec_height_ = height;
   num_layers_ = (num_layers <= 1) ? 1 : num_layers;  // Can also be zero.
   max_payload_size_ = mtu;
-  qm_resolution_->Initialize(target_bitrate_kbps,
-                             user_frame_rate_,
-                             codec_width_,
-                             codec_height_,
-                             num_layers_);
+  qm_resolution_->Initialize(target_bitrate_kbps, user_frame_rate_,
+                             codec_width_, codec_height_, num_layers_);
 }
 
 uint32_t MediaOptimization::SetTargetRates(
@@ -256,10 +245,8 @@ uint32_t MediaOptimization::SetTargetRates(
     // overhead data actually transmitted (including headers) the last
     // second.
     if (protection_callback) {
-      UpdateProtectionCallback(selected_method,
-                               &sent_video_rate_bps,
-                               &sent_nack_rate_bps,
-                               &sent_fec_rate_bps,
+      UpdateProtectionCallback(selected_method, &sent_video_rate_bps,
+                               &sent_nack_rate_bps, &sent_fec_rate_bps,
                                protection_callback);
     }
     uint32_t sent_total_rate_bps =
@@ -296,10 +283,8 @@ uint32_t MediaOptimization::SetTargetRates(
 
   if (enable_qm_ && qmsettings_callback) {
     // Update QM with rates.
-    qm_resolution_->UpdateRates(target_video_bitrate_kbps,
-                                sent_video_rate_kbps,
-                                incoming_frame_rate_,
-                                fraction_lost_);
+    qm_resolution_->UpdateRates(target_video_bitrate_kbps, sent_video_rate_kbps,
+                                incoming_frame_rate_, fraction_lost_);
     // Check for QM selection.
     bool select_qm = CheckStatusForQMchange();
     if (select_qm) {
@@ -514,8 +499,7 @@ void MediaOptimization::UpdateSentBitrate(int64_t now_ms) {
   }
   size_t framesize_sum = 0;
   for (FrameSampleList::iterator it = encoded_frame_samples_.begin();
-       it != encoded_frame_samples_.end();
-       ++it) {
+       it != encoded_frame_samples_.end(); ++it) {
     framesize_sum += it->size_bytes;
   }
   float denom = static_cast<float>(
@@ -565,7 +549,8 @@ bool MediaOptimization::QMUpdate(
   }
 
   LOG(LS_INFO) << "Media optimizer requests the video resolution to be changed "
-                  "to " << qm->codec_width << "x" << qm->codec_height << "@"
+                  "to "
+               << qm->codec_width << "x" << qm->codec_height << "@"
                << qm->frame_rate;
 
   // Update VPM with new target frame rate and frame size.
@@ -574,11 +559,11 @@ bool MediaOptimization::QMUpdate(
   // will vary/fluctuate, and since we don't want to change the state of the
   // VPM frame dropper, unless a temporal action was selected, we use the
   // quantity |qm->frame_rate| for updating.
-  video_qmsettings_callback->SetVideoQMSettings(
-      qm->frame_rate, codec_width_, codec_height_);
+  video_qmsettings_callback->SetVideoQMSettings(qm->frame_rate, codec_width_,
+                                                codec_height_);
   content_->UpdateFrameRate(qm->frame_rate);
-  qm_resolution_->UpdateCodecParameters(
-      qm->frame_rate, codec_width_, codec_height_);
+  qm_resolution_->UpdateCodecParameters(qm->frame_rate, codec_width_,
+                                        codec_height_);
   return true;
 }
 
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.h b/webrtc/modules/video_coding/media_optimization.h
index c4feeff743..54389bf5b5 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.h
+++ b/webrtc/modules/video_coding/media_optimization.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MEDIA_OPTIMIZATION_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MEDIA_OPTIMIZATION_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
 
 #include <list>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/media_opt_util.h"
+#include "webrtc/modules/video_coding/qm_select.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
@@ -85,15 +85,9 @@ class MediaOptimization {
   uint32_t SentBitRate();
 
  private:
-  enum {
-    kFrameCountHistorySize = 90
-  };
-  enum {
-    kFrameHistoryWinMs = 2000
-  };
-  enum {
-    kBitrateAverageWinMs = 1000
-  };
+  enum { kFrameCountHistorySize = 90 };
+  enum { kFrameHistoryWinMs = 2000 };
+  enum { kBitrateAverageWinMs = 1000 };
 
   struct EncodedFrameSample;
   typedef std::list<EncodedFrameSample> FrameSampleList;
@@ -177,4 +171,4 @@ class MediaOptimization {
 }  // namespace media_optimization
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MEDIA_OPTIMIZATION_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
diff --git a/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc b/webrtc/modules/video_coding/media_optimization_unittest.cc
index be528d9932..3f8ac5d075 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc
+++ b/webrtc/modules/video_coding/media_optimization_unittest.cc
@@ -9,7 +9,7 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -51,7 +51,6 @@ class TestMediaOptimization : public ::testing::Test {
   uint32_t next_timestamp_;
 };
 
-
 TEST_F(TestMediaOptimization, VerifyMuting) {
   // Enable video suspension with these limits.
   // Suspend the video when the rate is below 50 kbps and resume when it gets
diff --git a/webrtc/modules/video_coding/nack_fec_tables.h b/webrtc/modules/video_coding/nack_fec_tables.h
new file mode 100644
index 0000000000..f9f5ad97ac
--- /dev/null
+++ b/webrtc/modules/video_coding/nack_fec_tables.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_NACK_FEC_TABLES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_NACK_FEC_TABLES_H_
+
+namespace webrtc {
+
+// Table for adjusting FEC rate for NACK/FEC protection method
+// Table values are built as a sigmoid function, ranging from 0 to 100, based on
+// the HybridNackTH values defined in media_opt_util.h.
+const uint16_t VCMNackFecTable[100] = {
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   1,   1,   1,   1,
+    1,   2,   2,   2,   3,   3,   4,   5,   6,   7,   9,   10,  12,  15,  18,
+    21,  24,  28,  32,  37,  41,  46,  51,  56,  61,  66,  70,  74,  78,  81,
+    84,  86,  89,  90,  92,  93,  95,  95,  96,  97,  97,  98,  98,  99,  99,
+    99,  99,  99,  99,  100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100,
+    100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100,
+    100, 100, 100, 100, 100, 100, 100, 100, 100, 100,
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_NACK_FEC_TABLES_H_
diff --git a/webrtc/modules/video_coding/main/source/packet.cc b/webrtc/modules/video_coding/packet.cc
index fd5a6abb8c..e25de2ed6c 100644
--- a/webrtc/modules/video_coding/main/source/packet.cc
+++ b/webrtc/modules/video_coding/packet.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 #include <assert.h>
 
+#include "webrtc/modules/include/module_common_types.h"
+
 namespace webrtc {
 
 VCMPacket::VCMPacket()
@@ -34,49 +35,47 @@ VCMPacket::VCMPacket()
 
 VCMPacket::VCMPacket(const uint8_t* ptr,
                      const size_t size,
-                     const WebRtcRTPHeader& rtpHeader) :
-    payloadType(rtpHeader.header.payloadType),
-    timestamp(rtpHeader.header.timestamp),
-    ntp_time_ms_(rtpHeader.ntp_time_ms),
-    seqNum(rtpHeader.header.sequenceNumber),
-    dataPtr(ptr),
-    sizeBytes(size),
-    markerBit(rtpHeader.header.markerBit),
+                     const WebRtcRTPHeader& rtpHeader)
+    : payloadType(rtpHeader.header.payloadType),
+      timestamp(rtpHeader.header.timestamp),
+      ntp_time_ms_(rtpHeader.ntp_time_ms),
+      seqNum(rtpHeader.header.sequenceNumber),
+      dataPtr(ptr),
+      sizeBytes(size),
+      markerBit(rtpHeader.header.markerBit),
 
-    frameType(rtpHeader.frameType),
-    codec(kVideoCodecUnknown),
-    isFirstPacket(rtpHeader.type.Video.isFirstPacket),
-    completeNALU(kNaluComplete),
-    insertStartCode(false),
-    width(rtpHeader.type.Video.width),
-    height(rtpHeader.type.Video.height),
-    codecSpecificHeader(rtpHeader.type.Video)
-{
-    CopyCodecSpecifics(rtpHeader.type.Video);
+      frameType(rtpHeader.frameType),
+      codec(kVideoCodecUnknown),
+      isFirstPacket(rtpHeader.type.Video.isFirstPacket),
+      completeNALU(kNaluComplete),
+      insertStartCode(false),
+      width(rtpHeader.type.Video.width),
+      height(rtpHeader.type.Video.height),
+      codecSpecificHeader(rtpHeader.type.Video) {
+  CopyCodecSpecifics(rtpHeader.type.Video);
 }
 
 VCMPacket::VCMPacket(const uint8_t* ptr,
                      size_t size,
                      uint16_t seq,
                      uint32_t ts,
-                     bool mBit) :
-    payloadType(0),
-    timestamp(ts),
-    ntp_time_ms_(0),
-    seqNum(seq),
-    dataPtr(ptr),
-    sizeBytes(size),
-    markerBit(mBit),
+                     bool mBit)
+    : payloadType(0),
+      timestamp(ts),
+      ntp_time_ms_(0),
+      seqNum(seq),
+      dataPtr(ptr),
+      sizeBytes(size),
+      markerBit(mBit),
 
-    frameType(kVideoFrameDelta),
-    codec(kVideoCodecUnknown),
-    isFirstPacket(false),
-    completeNALU(kNaluComplete),
-    insertStartCode(false),
-    width(0),
-    height(0),
-    codecSpecificHeader()
-{}
+      frameType(kVideoFrameDelta),
+      codec(kVideoCodecUnknown),
+      isFirstPacket(false),
+      completeNALU(kNaluComplete),
+      insertStartCode(false),
+      width(0),
+      height(0),
+      codecSpecificHeader() {}
 
 void VCMPacket::Reset() {
   payloadType = 0;
diff --git a/webrtc/modules/video_coding/packet.h b/webrtc/modules/video_coding/packet.h
new file mode 100644
index 0000000000..b77c1df039
--- /dev/null
+++ b/webrtc/modules/video_coding/packet.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class VCMPacket {
+ public:
+  VCMPacket();
+  VCMPacket(const uint8_t* ptr,
+            const size_t size,
+            const WebRtcRTPHeader& rtpHeader);
+  VCMPacket(const uint8_t* ptr,
+            size_t size,
+            uint16_t seqNum,
+            uint32_t timestamp,
+            bool markerBit);
+
+  void Reset();
+
+  uint8_t payloadType;
+  uint32_t timestamp;
+  // NTP time of the capture time in local timebase in milliseconds.
+  int64_t ntp_time_ms_;
+  uint16_t seqNum;
+  const uint8_t* dataPtr;
+  size_t sizeBytes;
+  bool markerBit;
+
+  FrameType frameType;
+  VideoCodecType codec;
+
+  bool isFirstPacket;                // Is this first packet in a frame.
+  VCMNaluCompleteness completeNALU;  // Default is kNaluIncomplete.
+  bool insertStartCode;  // True if a start code should be inserted before this
+                         // packet.
+  int width;
+  int height;
+  RTPVideoHeader codecSpecificHeader;
+
+ protected:
+  void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
diff --git a/webrtc/modules/video_coding/main/source/qm_select.cc b/webrtc/modules/video_coding/qm_select.cc
index e86d0755c0..9da42bb33c 100644
--- a/webrtc/modules/video_coding/main/source/qm_select.cc
+++ b/webrtc/modules/video_coding/qm_select.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
+#include "webrtc/modules/video_coding/qm_select.h"
 
 #include <math.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/qm_select_data.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/qm_select_data.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -36,8 +36,7 @@ VCMQmMethod::VCMQmMethod()
   ResetQM();
 }
 
-VCMQmMethod::~VCMQmMethod() {
-}
+VCMQmMethod::~VCMQmMethod() {}
 
 void VCMQmMethod::ResetQM() {
   aspect_ratio_ = 1.0f;
@@ -52,7 +51,7 @@ uint8_t VCMQmMethod::ComputeContentClass() {
   return content_class_ = 3 * motion_.level + spatial_.level;
 }
 
-void VCMQmMethod::UpdateContent(const VideoContentMetrics*  contentMetrics) {
+void VCMQmMethod::UpdateContent(const VideoContentMetrics* contentMetrics) {
   content_metrics_ = contentMetrics;
 }
 
@@ -64,7 +63,7 @@ void VCMQmMethod::ComputeMotionNFD() {
   if (motion_.value < kLowMotionNfd) {
     motion_.level = kLow;
   } else if (motion_.value > kHighMotionNfd) {
-    motion_.level  = kHigh;
+    motion_.level = kHigh;
   } else {
     motion_.level = kDefault;
   }
@@ -75,7 +74,7 @@ void VCMQmMethod::ComputeSpatial() {
   float spatial_err_h = 0.0;
   float spatial_err_v = 0.0;
   if (content_metrics_) {
-    spatial_err =  content_metrics_->spatial_pred_err;
+    spatial_err = content_metrics_->spatial_pred_err;
     spatial_err_h = content_metrics_->spatial_pred_err_h;
     spatial_err_v = content_metrics_->spatial_pred_err_v;
   }
@@ -94,8 +93,7 @@ void VCMQmMethod::ComputeSpatial() {
   }
 }
 
-ImageType VCMQmMethod::GetImageType(uint16_t width,
-                                    uint16_t height) {
+ImageType VCMQmMethod::GetImageType(uint16_t width, uint16_t height) {
   // Get the image type for the encoder frame size.
   uint32_t image_size = width * height;
   if (image_size == kSizeOfImageType[kQCIF]) {
@@ -142,7 +140,7 @@ FrameRateLevelClass VCMQmMethod::FrameRateLevel(float avg_framerate) {
   } else if (avg_framerate <= kMiddleFrameRate) {
     return kFrameRateMiddle1;
   } else if (avg_framerate <= kHighFrameRate) {
-     return kFrameRateMiddle2;
+    return kFrameRateMiddle2;
   } else {
     return kFrameRateHigh;
   }
@@ -150,8 +148,7 @@ FrameRateLevelClass VCMQmMethod::FrameRateLevel(float avg_framerate) {
 
 // RESOLUTION CLASS
 
-VCMQmResolution::VCMQmResolution()
-    :  qm_(new VCMResolutionScale()) {
+VCMQmResolution::VCMQmResolution() : qm_(new VCMResolutionScale()) {
   Reset();
 }
 
@@ -174,7 +171,7 @@ void VCMQmResolution::ResetRates() {
 
 void VCMQmResolution::ResetDownSamplingState() {
   state_dec_factor_spatial_ = 1.0;
-  state_dec_factor_temporal_  = 1.0;
+  state_dec_factor_temporal_ = 1.0;
   for (int i = 0; i < kDownActionHistorySize; i++) {
     down_action_history_[i].spatial = kNoChangeSpatial;
     down_action_history_[i].temporal = kNoChangeTemporal;
@@ -225,11 +222,12 @@ int VCMQmResolution::Initialize(float bitrate,
   buffer_level_ = kInitBufferLevel * target_bitrate_;
   // Per-frame bandwidth.
   per_frame_bandwidth_ = target_bitrate_ / user_framerate;
-  init_  = true;
+  init_ = true;
   return VCM_OK;
 }
 
-void VCMQmResolution::UpdateCodecParameters(float frame_rate, uint16_t width,
+void VCMQmResolution::UpdateCodecParameters(float frame_rate,
+                                            uint16_t width,
                                             uint16_t height) {
   width_ = width;
   height_ = height;
@@ -283,12 +281,12 @@ void VCMQmResolution::UpdateRates(float target_bitrate,
 
   // Update with the current new target and frame rate:
   // these values are ones the encoder will use for the current/next ~1sec.
-  target_bitrate_ =  target_bitrate;
+  target_bitrate_ = target_bitrate;
   incoming_framerate_ = incoming_framerate;
   sum_incoming_framerate_ += incoming_framerate_;
   // Update the per_frame_bandwidth:
   // this is the per_frame_bw for the current/next ~1sec.
-  per_frame_bandwidth_  = 0.0f;
+  per_frame_bandwidth_ = 0.0f;
   if (incoming_framerate_ > 0.0f) {
     per_frame_bandwidth_ = target_bitrate_ / incoming_framerate_;
   }
@@ -313,7 +311,7 @@ int VCMQmResolution::SelectResolution(VCMResolutionScale** qm) {
   }
   if (content_metrics_ == NULL) {
     Reset();
-    *qm =  qm_;
+    *qm = qm_;
     return VCM_OK;
   }
 
@@ -376,31 +374,31 @@ void VCMQmResolution::ComputeRatesForSelection() {
   avg_rate_mismatch_sgn_ = 0.0f;
   avg_packet_loss_ = 0.0f;
   if (frame_cnt_ > 0) {
-    avg_ratio_buffer_low_ = static_cast<float>(low_buffer_cnt_) /
-        static_cast<float>(frame_cnt_);
+    avg_ratio_buffer_low_ =
+        static_cast<float>(low_buffer_cnt_) / static_cast<float>(frame_cnt_);
   }
   if (update_rate_cnt_ > 0) {
-    avg_rate_mismatch_ = static_cast<float>(sum_rate_MM_) /
-        static_cast<float>(update_rate_cnt_);
+    avg_rate_mismatch_ =
+        static_cast<float>(sum_rate_MM_) / static_cast<float>(update_rate_cnt_);
     avg_rate_mismatch_sgn_ = static_cast<float>(sum_rate_MM_sgn_) /
-        static_cast<float>(update_rate_cnt_);
+                             static_cast<float>(update_rate_cnt_);
     avg_target_rate_ = static_cast<float>(sum_target_rate_) /
-        static_cast<float>(update_rate_cnt_);
+                       static_cast<float>(update_rate_cnt_);
     avg_incoming_framerate_ = static_cast<float>(sum_incoming_framerate_) /
-        static_cast<float>(update_rate_cnt_);
-    avg_packet_loss_ =  static_cast<float>(sum_packet_loss_) /
-        static_cast<float>(update_rate_cnt_);
+                              static_cast<float>(update_rate_cnt_);
+    avg_packet_loss_ = static_cast<float>(sum_packet_loss_) /
+                       static_cast<float>(update_rate_cnt_);
   }
   // For selection we may want to weight some quantities more heavily
   // with the current (i.e., next ~1sec) rate values.
-  avg_target_rate_ = kWeightRate * avg_target_rate_ +
-      (1.0 - kWeightRate) * target_bitrate_;
+  avg_target_rate_ =
+      kWeightRate * avg_target_rate_ + (1.0 - kWeightRate) * target_bitrate_;
   avg_incoming_framerate_ = kWeightRate * avg_incoming_framerate_ +
-      (1.0 - kWeightRate) * incoming_framerate_;
+                            (1.0 - kWeightRate) * incoming_framerate_;
   // Use base layer frame rate for temporal layers: this will favor spatial.
   assert(num_layers_ > 0);
-  framerate_level_ = FrameRateLevel(
-      avg_incoming_framerate_ / static_cast<float>(1 << (num_layers_ - 1)));
+  framerate_level_ = FrameRateLevel(avg_incoming_framerate_ /
+                                    static_cast<float>(1 << (num_layers_ - 1)));
 }
 
 void VCMQmResolution::ComputeEncoderState() {
@@ -412,7 +410,7 @@ void VCMQmResolution::ComputeEncoderState() {
   // 2) rate mis-match is high, and consistent over-shooting by encoder.
   if ((avg_ratio_buffer_low_ > kMaxBufferLow) ||
       ((avg_rate_mismatch_ > kMaxRateMisMatch) &&
-          (avg_rate_mismatch_sgn_ < -kRateOverShoot))) {
+       (avg_rate_mismatch_sgn_ < -kRateOverShoot))) {
     encoder_state_ = kStressedEncoding;
   }
   // Assign easy state if:
@@ -435,9 +433,9 @@ bool VCMQmResolution::GoingUpResolution() {
   // Modify the fac_width/height for this case.
   if (down_action_history_[0].spatial == kOneQuarterSpatialUniform) {
     fac_width = kFactorWidthSpatial[kOneQuarterSpatialUniform] /
-        kFactorWidthSpatial[kOneHalfSpatialUniform];
+                kFactorWidthSpatial[kOneHalfSpatialUniform];
     fac_height = kFactorHeightSpatial[kOneQuarterSpatialUniform] /
-        kFactorHeightSpatial[kOneHalfSpatialUniform];
+                 kFactorHeightSpatial[kOneHalfSpatialUniform];
   }
 
   // Check if we should go up both spatially and temporally.
@@ -459,8 +457,8 @@ bool VCMQmResolution::GoingUpResolution() {
                                               kTransRateScaleUpSpatial);
   }
   if (down_action_history_[0].temporal != kNoChangeTemporal) {
-    selected_up_temporal = ConditionForGoingUp(1.0f, 1.0f, fac_temp,
-                                               kTransRateScaleUpTemp);
+    selected_up_temporal =
+        ConditionForGoingUp(1.0f, 1.0f, fac_temp, kTransRateScaleUpTemp);
   }
   if (selected_up_spatial && !selected_up_temporal) {
     action_.spatial = down_action_history_[0].spatial;
@@ -484,13 +482,13 @@ bool VCMQmResolution::ConditionForGoingUp(float fac_width,
                                           float fac_height,
                                           float fac_temp,
                                           float scale_fac) {
-  float estimated_transition_rate_up = GetTransitionRate(fac_width, fac_height,
-                                                         fac_temp, scale_fac);
+  float estimated_transition_rate_up =
+      GetTransitionRate(fac_width, fac_height, fac_temp, scale_fac);
   // Go back up if:
   // 1) target rate is above threshold and current encoder state is stable, or
   // 2) encoder state is easy (encoder is significantly under-shooting target).
   if (((avg_target_rate_ > estimated_transition_rate_up) &&
-      (encoder_state_ == kStableEncoding)) ||
+       (encoder_state_ == kStableEncoding)) ||
       (encoder_state_ == kEasyEncoding)) {
     return true;
   } else {
@@ -505,7 +503,7 @@ bool VCMQmResolution::GoingDownResolution() {
   // Resolution reduction if:
   // (1) target rate is below transition rate, or
   // (2) encoder is in stressed state and target rate below a max threshold.
-  if ((avg_target_rate_ < estimated_transition_rate_down ) ||
+  if ((avg_target_rate_ < estimated_transition_rate_down) ||
       (encoder_state_ == kStressedEncoding && avg_target_rate_ < max_rate)) {
     // Get the down-sampling action: based on content class, and how low
     // average target rate is relative to transition rate.
@@ -529,9 +527,7 @@ bool VCMQmResolution::GoingDownResolution() {
         action_.spatial = kNoChangeSpatial;
         break;
       }
-      default: {
-        assert(false);
-      }
+      default: { assert(false); }
     }
     switch (temp_fact) {
       case 3: {
@@ -546,9 +542,7 @@ bool VCMQmResolution::GoingDownResolution() {
         action_.temporal = kNoChangeTemporal;
         break;
       }
-      default: {
-        assert(false);
-      }
+      default: { assert(false); }
     }
     // Only allow for one action (spatial or temporal) at a given time.
     assert(action_.temporal == kNoChangeTemporal ||
@@ -572,9 +566,9 @@ float VCMQmResolution::GetTransitionRate(float fac_width,
                                          float fac_height,
                                          float fac_temp,
                                          float scale_fac) {
-  ImageType image_type = GetImageType(
-      static_cast<uint16_t>(fac_width * width_),
-      static_cast<uint16_t>(fac_height * height_));
+  ImageType image_type =
+      GetImageType(static_cast<uint16_t>(fac_width * width_),
+                   static_cast<uint16_t>(fac_height * height_));
 
   FrameRateLevelClass framerate_level =
       FrameRateLevel(fac_temp * avg_incoming_framerate_);
@@ -589,13 +583,13 @@ float VCMQmResolution::GetTransitionRate(float fac_width,
   // Nominal values based on image format (frame size and frame rate).
   float max_rate = kFrameRateFac[framerate_level] * kMaxRateQm[image_type];
 
-  uint8_t image_class = image_type > kVGA ? 1: 0;
+  uint8_t image_class = image_type > kVGA ? 1 : 0;
   uint8_t table_index = image_class * 9 + content_class_;
   // Scale factor for down-sampling transition threshold:
   // factor based on the content class and the image size.
   float scaleTransRate = kScaleTransRateQm[table_index];
   // Threshold bitrate for resolution action.
-  return static_cast<float> (scale_fac * scaleTransRate * max_rate);
+  return static_cast<float>(scale_fac * scaleTransRate * max_rate);
 }
 
 void VCMQmResolution::UpdateDownsamplingState(UpDownAction up_down) {
@@ -605,9 +599,9 @@ void VCMQmResolution::UpdateDownsamplingState(UpDownAction up_down) {
     // If last spatial action was 1/2x1/2, we undo it in two steps, so the
     // spatial scale factor in this first step is modified as (4.0/3.0 / 2.0).
     if (action_.spatial == kOneQuarterSpatialUniform) {
-      qm_->spatial_width_fact =
-          1.0f * kFactorWidthSpatial[kOneHalfSpatialUniform] /
-          kFactorWidthSpatial[kOneQuarterSpatialUniform];
+      qm_->spatial_width_fact = 1.0f *
+                                kFactorWidthSpatial[kOneHalfSpatialUniform] /
+                                kFactorWidthSpatial[kOneQuarterSpatialUniform];
       qm_->spatial_height_fact =
           1.0f * kFactorHeightSpatial[kOneHalfSpatialUniform] /
           kFactorHeightSpatial[kOneQuarterSpatialUniform];
@@ -628,17 +622,18 @@ void VCMQmResolution::UpdateDownsamplingState(UpDownAction up_down) {
   }
   UpdateCodecResolution();
   state_dec_factor_spatial_ = state_dec_factor_spatial_ *
-      qm_->spatial_width_fact * qm_->spatial_height_fact;
+                              qm_->spatial_width_fact *
+                              qm_->spatial_height_fact;
   state_dec_factor_temporal_ = state_dec_factor_temporal_ * qm_->temporal_fact;
 }
 
-void  VCMQmResolution::UpdateCodecResolution() {
+void VCMQmResolution::UpdateCodecResolution() {
   if (action_.spatial != kNoChangeSpatial) {
     qm_->change_resolution_spatial = true;
-    qm_->codec_width = static_cast<uint16_t>(width_ /
-                                             qm_->spatial_width_fact + 0.5f);
-    qm_->codec_height = static_cast<uint16_t>(height_ /
-                                              qm_->spatial_height_fact + 0.5f);
+    qm_->codec_width =
+        static_cast<uint16_t>(width_ / qm_->spatial_width_fact + 0.5f);
+    qm_->codec_height =
+        static_cast<uint16_t>(height_ / qm_->spatial_height_fact + 0.5f);
     // Size should not exceed native sizes.
     assert(qm_->codec_width <= native_width_);
     assert(qm_->codec_height <= native_height_);
@@ -662,8 +657,9 @@ void  VCMQmResolution::UpdateCodecResolution() {
 }
 
 uint8_t VCMQmResolution::RateClass(float transition_rate) {
-  return avg_target_rate_ < (kFacLowRate * transition_rate) ? 0:
-  (avg_target_rate_ >= transition_rate ? 2 : 1);
+  return avg_target_rate_ < (kFacLowRate * transition_rate)
+             ? 0
+             : (avg_target_rate_ >= transition_rate ? 2 : 1);
 }
 
 // TODO(marpan): Would be better to capture these frame rate adjustments by
@@ -698,15 +694,14 @@ void VCMQmResolution::AdjustAction() {
   }
   // Never use temporal action if number of temporal layers is above 2.
   if (num_layers_ > 2) {
-    if (action_.temporal !=  kNoChangeTemporal) {
+    if (action_.temporal != kNoChangeTemporal) {
       action_.spatial = kOneHalfSpatialUniform;
     }
     action_.temporal = kNoChangeTemporal;
   }
   // If spatial action was selected, we need to make sure the frame sizes
   // are multiples of two. Otherwise switch to 2/3 temporal.
-  if (action_.spatial != kNoChangeSpatial &&
-      !EvenFrameSize()) {
+  if (action_.spatial != kNoChangeSpatial && !EvenFrameSize()) {
     action_.spatial = kNoChangeSpatial;
     // Only one action (spatial or temporal) is allowed at a given time, so need
     // to check whether temporal action is currently selected.
@@ -722,35 +717,36 @@ void VCMQmResolution::ConvertSpatialFractionalToWhole() {
     bool found = false;
     int isel = kDownActionHistorySize;
     for (int i = 0; i < kDownActionHistorySize; ++i) {
-      if (down_action_history_[i].spatial ==  kOneHalfSpatialUniform) {
+      if (down_action_history_[i].spatial == kOneHalfSpatialUniform) {
         isel = i;
         found = true;
         break;
       }
     }
     if (found) {
-       action_.spatial = kOneQuarterSpatialUniform;
-       state_dec_factor_spatial_ = state_dec_factor_spatial_ /
-           (kFactorWidthSpatial[kOneHalfSpatialUniform] *
-            kFactorHeightSpatial[kOneHalfSpatialUniform]);
-       // Check if switching to 1/2x1/2 (=1/4) spatial is allowed.
-       ConstrainAmountOfDownSampling();
-       if (action_.spatial == kNoChangeSpatial) {
-         // Not allowed. Go back to 3/4x3/4 spatial.
-         action_.spatial = kOneHalfSpatialUniform;
-         state_dec_factor_spatial_ = state_dec_factor_spatial_ *
-             kFactorWidthSpatial[kOneHalfSpatialUniform] *
-             kFactorHeightSpatial[kOneHalfSpatialUniform];
-       } else {
-         // Switching is allowed. Remove 3/4x3/4 from the history, and update
-         // the frame size.
-         for (int i = isel; i < kDownActionHistorySize - 1; ++i) {
-           down_action_history_[i].spatial =
-               down_action_history_[i + 1].spatial;
-         }
-         width_ = width_ * kFactorWidthSpatial[kOneHalfSpatialUniform];
-         height_ = height_ * kFactorHeightSpatial[kOneHalfSpatialUniform];
-       }
+      action_.spatial = kOneQuarterSpatialUniform;
+      state_dec_factor_spatial_ =
+          state_dec_factor_spatial_ /
+          (kFactorWidthSpatial[kOneHalfSpatialUniform] *
+           kFactorHeightSpatial[kOneHalfSpatialUniform]);
+      // Check if switching to 1/2x1/2 (=1/4) spatial is allowed.
+      ConstrainAmountOfDownSampling();
+      if (action_.spatial == kNoChangeSpatial) {
+        // Not allowed. Go back to 3/4x3/4 spatial.
+        action_.spatial = kOneHalfSpatialUniform;
+        state_dec_factor_spatial_ =
+            state_dec_factor_spatial_ *
+            kFactorWidthSpatial[kOneHalfSpatialUniform] *
+            kFactorHeightSpatial[kOneHalfSpatialUniform];
+      } else {
+        // Switching is allowed. Remove 3/4x3/4 from the history, and update
+        // the frame size.
+        for (int i = isel; i < kDownActionHistorySize - 1; ++i) {
+          down_action_history_[i].spatial = down_action_history_[i + 1].spatial;
+        }
+        width_ = width_ * kFactorWidthSpatial[kOneHalfSpatialUniform];
+        height_ = height_ * kFactorHeightSpatial[kOneHalfSpatialUniform];
+      }
     }
   }
 }
@@ -815,8 +811,8 @@ void VCMQmResolution::ConstrainAmountOfDownSampling() {
   float spatial_width_fact = kFactorWidthSpatial[action_.spatial];
   float spatial_height_fact = kFactorHeightSpatial[action_.spatial];
   float temporal_fact = kFactorTemporal[action_.temporal];
-  float new_dec_factor_spatial = state_dec_factor_spatial_ *
-      spatial_width_fact * spatial_height_fact;
+  float new_dec_factor_spatial =
+      state_dec_factor_spatial_ * spatial_width_fact * spatial_height_fact;
   float new_dec_factor_temp = state_dec_factor_temporal_ * temporal_fact;
 
   // No spatial sampling if current frame size is too small, or if the
@@ -908,8 +904,7 @@ VCMQmRobustness::VCMQmRobustness() {
   Reset();
 }
 
-VCMQmRobustness::~VCMQmRobustness() {
-}
+VCMQmRobustness::~VCMQmRobustness() {}
 
 void VCMQmRobustness::Reset() {
   prev_total_rate_ = 0.0f;
@@ -928,7 +923,7 @@ float VCMQmRobustness::AdjustFecFactor(uint8_t code_rate_delta,
                                        int64_t rtt_time,
                                        uint8_t packet_loss) {
   // Default: no adjustment
-  float adjust_fec =  1.0f;
+  float adjust_fec = 1.0f;
   if (content_metrics_ == NULL) {
     return adjust_fec;
   }
@@ -955,4 +950,4 @@ bool VCMQmRobustness::SetUepProtection(uint8_t code_rate_delta,
   // Default.
   return false;
 }
-}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/qm_select.h b/webrtc/modules/video_coding/qm_select.h
index 079e7f8879..764b5ed8e3 100644
--- a/webrtc/modules/video_coding/main/source/qm_select.h
+++ b/webrtc/modules/video_coding/qm_select.h
@@ -30,8 +30,7 @@ struct VCMResolutionScale {
         spatial_height_fact(1.0f),
         temporal_fact(1.0f),
         change_resolution_spatial(false),
-        change_resolution_temporal(false) {
-  }
+        change_resolution_temporal(false) {}
   uint16_t codec_width;
   uint16_t codec_height;
   float frame_rate;
@@ -43,20 +42,20 @@ struct VCMResolutionScale {
 };
 
 enum ImageType {
-  kQCIF = 0,            // 176x144
-  kHCIF,                // 264x216 = half(~3/4x3/4) CIF.
-  kQVGA,                // 320x240 = quarter VGA.
-  kCIF,                 // 352x288
-  kHVGA,                // 480x360 = half(~3/4x3/4) VGA.
-  kVGA,                 // 640x480
-  kQFULLHD,             // 960x540 = quarter FULLHD, and half(~3/4x3/4) WHD.
-  kWHD,                 // 1280x720
-  kFULLHD,              // 1920x1080
+  kQCIF = 0,  // 176x144
+  kHCIF,      // 264x216 = half(~3/4x3/4) CIF.
+  kQVGA,      // 320x240 = quarter VGA.
+  kCIF,       // 352x288
+  kHVGA,      // 480x360 = half(~3/4x3/4) VGA.
+  kVGA,       // 640x480
+  kQFULLHD,   // 960x540 = quarter FULLHD, and half(~3/4x3/4) WHD.
+  kWHD,       // 1280x720
+  kFULLHD,    // 1920x1080
   kNumImageTypes
 };
 
-const uint32_t kSizeOfImageType[kNumImageTypes] =
-{ 25344, 57024, 76800, 101376, 172800, 307200, 518400, 921600, 2073600 };
+const uint32_t kSizeOfImageType[kNumImageTypes] = {
+    25344, 57024, 76800, 101376, 172800, 307200, 518400, 921600, 2073600};
 
 enum FrameRateLevelClass {
   kFrameRateLow,
@@ -65,17 +64,10 @@ enum FrameRateLevelClass {
   kFrameRateHigh
 };
 
-enum ContentLevelClass {
-  kLow,
-  kHigh,
-  kDefault
-};
+enum ContentLevelClass { kLow, kHigh, kDefault };
 
 struct VCMContFeature {
-  VCMContFeature()
-      : value(0.0f),
-        level(kDefault) {
-  }
+  VCMContFeature() : value(0.0f), level(kDefault) {}
   void Reset() {
     value = 0.0f;
     level = kDefault;
@@ -84,43 +76,34 @@ struct VCMContFeature {
   ContentLevelClass level;
 };
 
-enum UpDownAction {
-  kUpResolution,
-  kDownResolution
-};
+enum UpDownAction { kUpResolution, kDownResolution };
 
 enum SpatialAction {
   kNoChangeSpatial,
-  kOneHalfSpatialUniform,        // 3/4 x 3/4: 9/6 ~1/2 pixel reduction.
-  kOneQuarterSpatialUniform,     // 1/2 x 1/2: 1/4 pixel reduction.
+  kOneHalfSpatialUniform,     // 3/4 x 3/4: 9/6 ~1/2 pixel reduction.
+  kOneQuarterSpatialUniform,  // 1/2 x 1/2: 1/4 pixel reduction.
   kNumModesSpatial
 };
 
 enum TemporalAction {
   kNoChangeTemporal,
-  kTwoThirdsTemporal,     // 2/3 frame rate reduction
-  kOneHalfTemporal,       // 1/2 frame rate reduction
+  kTwoThirdsTemporal,  // 2/3 frame rate reduction
+  kOneHalfTemporal,    // 1/2 frame rate reduction
   kNumModesTemporal
 };
 
 struct ResolutionAction {
-  ResolutionAction()
-      : spatial(kNoChangeSpatial),
-        temporal(kNoChangeTemporal) {
-  }
+  ResolutionAction() : spatial(kNoChangeSpatial), temporal(kNoChangeTemporal) {}
   SpatialAction spatial;
   TemporalAction temporal;
 };
 
 // Down-sampling factors for spatial (width and height), and temporal.
-const float kFactorWidthSpatial[kNumModesSpatial] =
-    { 1.0f, 4.0f / 3.0f, 2.0f };
+const float kFactorWidthSpatial[kNumModesSpatial] = {1.0f, 4.0f / 3.0f, 2.0f};
 
-const float kFactorHeightSpatial[kNumModesSpatial] =
-    { 1.0f, 4.0f / 3.0f, 2.0f };
+const float kFactorHeightSpatial[kNumModesSpatial] = {1.0f, 4.0f / 3.0f, 2.0f};
 
-const float kFactorTemporal[kNumModesTemporal] =
-    { 1.0f, 1.5f, 2.0f };
+const float kFactorTemporal[kNumModesTemporal] = {1.0f, 1.5f, 2.0f};
 
 enum EncoderState {
   kStableEncoding,    // Low rate mis-match, stable buffer levels.
@@ -297,7 +280,7 @@ class VCMQmResolution : public VCMQmMethod {
   // Select the directional (1x2 or 2x1) spatial down-sampling action.
   void SelectSpatialDirectionMode(float transition_rate);
 
-  enum { kDownActionHistorySize = 10};
+  enum { kDownActionHistorySize = 10 };
 
   VCMResolutionScale* qm_;
   // Encoder rate control parameters.
diff --git a/webrtc/modules/video_coding/main/source/qm_select_data.h b/webrtc/modules/video_coding/qm_select_data.h
index dc6bce4811..49190ef53b 100644
--- a/webrtc/modules/video_coding/main/source/qm_select_data.h
+++ b/webrtc/modules/video_coding/qm_select_data.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
+#define WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
 
 /***************************************************************
 *QMSelectData.h
@@ -69,36 +69,36 @@ const uint16_t kMaxRateQm[9] = {
 
 // Frame rate scale for maximum transition rate.
 const float kFrameRateFac[4] = {
-    0.5f,    // Low
-    0.7f,    // Middle level 1
-    0.85f,   // Middle level 2
-    1.0f,    // High
+    0.5f,   // Low
+    0.7f,   // Middle level 1
+    0.85f,  // Middle level 2
+    1.0f,   // High
 };
 
 // Scale for transitional rate: based on content class
 // motion=L/H/D,spatial==L/H/D: for low, high, middle levels
 const float kScaleTransRateQm[18] = {
     // VGA and lower
-    0.40f,       // L, L
-    0.50f,       // L, H
-    0.40f,       // L, D
-    0.60f,       // H ,L
-    0.60f,       // H, H
-    0.60f,       // H, D
-    0.50f,       // D, L
-    0.50f,       // D, D
-    0.50f,       // D, H
+    0.40f,  // L, L
+    0.50f,  // L, H
+    0.40f,  // L, D
+    0.60f,  // H ,L
+    0.60f,  // H, H
+    0.60f,  // H, D
+    0.50f,  // D, L
+    0.50f,  // D, D
+    0.50f,  // D, H
 
     // over VGA
-    0.40f,       // L, L
-    0.50f,       // L, H
-    0.40f,       // L, D
-    0.60f,       // H ,L
-    0.60f,       // H, H
-    0.60f,       // H, D
-    0.50f,       // D, L
-    0.50f,       // D, D
-    0.50f,       // D, H
+    0.40f,  // L, L
+    0.50f,  // L, H
+    0.40f,  // L, D
+    0.60f,  // H ,L
+    0.60f,  // H, H
+    0.60f,  // H, D
+    0.50f,  // D, L
+    0.50f,  // D, D
+    0.50f,  // D, H
 };
 
 // Threshold on the target rate relative to transitional rate.
@@ -108,73 +108,73 @@ const float kFacLowRate = 0.5f;
 // motion=L/H/D,spatial==L/H/D, for low, high, middle levels;
 // rate = 0/1/2, for target rate state relative to transition rate.
 const uint8_t kSpatialAction[27] = {
-// rateClass = 0:
-    1,       // L, L
-    1,       // L, H
-    1,       // L, D
-    4,       // H ,L
-    1,       // H, H
-    4,       // H, D
-    4,       // D, L
-    1,       // D, H
-    2,       // D, D
-
-// rateClass = 1:
-    1,       // L, L
-    1,       // L, H
-    1,       // L, D
-    2,       // H ,L
-    1,       // H, H
-    2,       // H, D
-    2,       // D, L
-    1,       // D, H
-    2,       // D, D
-
-// rateClass = 2:
-    1,       // L, L
-    1,       // L, H
-    1,       // L, D
-    2,       // H ,L
-    1,       // H, H
-    2,       // H, D
-    2,       // D, L
-    1,       // D, H
-    2,       // D, D
+    // rateClass = 0:
+    1,  // L, L
+    1,  // L, H
+    1,  // L, D
+    4,  // H ,L
+    1,  // H, H
+    4,  // H, D
+    4,  // D, L
+    1,  // D, H
+    2,  // D, D
+
+    // rateClass = 1:
+    1,  // L, L
+    1,  // L, H
+    1,  // L, D
+    2,  // H ,L
+    1,  // H, H
+    2,  // H, D
+    2,  // D, L
+    1,  // D, H
+    2,  // D, D
+
+    // rateClass = 2:
+    1,  // L, L
+    1,  // L, H
+    1,  // L, D
+    2,  // H ,L
+    1,  // H, H
+    2,  // H, D
+    2,  // D, L
+    1,  // D, H
+    2,  // D, D
 };
 
 const uint8_t kTemporalAction[27] = {
-// rateClass = 0:
-    3,       // L, L
-    2,       // L, H
-    2,       // L, D
-    1,       // H ,L
-    3,       // H, H
-    1,       // H, D
-    1,       // D, L
-    2,       // D, H
-    1,       // D, D
-
-// rateClass = 1:
-    3,       // L, L
-    3,       // L, H
-    3,       // L, D
-    1,       // H ,L
-    3,       // H, H
-    1,       // H, D
-    1,       // D, L
-    3,       // D, H
-    1,       // D, D
-
-// rateClass = 2:
-    1,       // L, L
-    3,       // L, H
-    3,       // L, D
-    1,       // H ,L
-    3,       // H, H
-    1,       // H, D
-    1,       // D, L
-    3,       // D, H
-    1,       // D, D
+    // rateClass = 0:
+    3,  // L, L
+    2,  // L, H
+    2,  // L, D
+    1,  // H ,L
+    3,  // H, H
+    1,  // H, D
+    1,  // D, L
+    2,  // D, H
+    1,  // D, D
+
+    // rateClass = 1:
+    3,  // L, L
+    3,  // L, H
+    3,  // L, D
+    1,  // H ,L
+    3,  // H, H
+    1,  // H, D
+    1,  // D, L
+    3,  // D, H
+    1,  // D, D
+
+    // rateClass = 2:
+    1,  // L, L
+    3,  // L, H
+    3,  // L, D
+    1,  // H ,L
+    3,  // H, H
+    1,  // H, D
+    1,  // D, L
+    3,  // D, H
+    1,  // D, D
 };
 
 // Control the total amount of down-sampling allowed.
@@ -224,4 +224,4 @@ const float kSpatialErrVertVsHoriz = 0.1f;  // percentage to favor H over V
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
diff --git a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc b/webrtc/modules/video_coding/qm_select_unittest.cc
index 6abc0d3099..f8542ec676 100644
--- a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
+++ b/webrtc/modules/video_coding/qm_select_unittest.cc
@@ -15,8 +15,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/qm_select.h"
 
 namespace webrtc {
 
@@ -32,10 +32,9 @@ const float kTemporalHigh = 0.1f;
 class QmSelectTest : public ::testing::Test {
  protected:
   QmSelectTest()
-      :  qm_resolution_(new VCMQmResolution()),
-         content_metrics_(new VideoContentMetrics()),
-         qm_scale_(NULL) {
-  }
+      : qm_resolution_(new VCMQmResolution()),
+        content_metrics_(new VideoContentMetrics()),
+        qm_scale_(NULL) {}
   VCMQmResolution* qm_resolution_;
   VideoContentMetrics* content_metrics_;
   VCMResolutionScale* qm_scale_;
@@ -87,8 +86,8 @@ TEST_F(QmSelectTest, HandleInputs) {
   qm_resolution_->UpdateContent(content_metrics);
   // Content metrics are NULL: Expect success and no down-sampling action.
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0, 1.0, 1.0, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0, 1.0, 1.0, 640, 480, 30.0f));
 }
 
 // TODO(marpan): Add a test for number of temporal layers > 1.
@@ -118,8 +117,8 @@ TEST_F(QmSelectTest, NoActionHighRate) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 }
 
 // Rate is well below transition, down-sampling action is taken,
@@ -149,40 +148,40 @@ TEST_F(QmSelectTest, DownActionLowRate) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, low spatial: 2/3 temporal is expected.
   UpdateQmContentData(kTemporalLow, kSpatialLow, kSpatialLow, kSpatialLow);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Medium motion, low spatial: 2x2 spatial expected.
   UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // High motion, high spatial: 2/3 temporal expected.
   UpdateQmContentData(kTemporalHigh, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(4, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial: 1/2 temporal expected.
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Medium motion, high spatial: 1/2 temporal expected.
@@ -190,8 +189,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
                       kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // High motion, medium spatial: 2x2 spatial expected.
@@ -200,8 +199,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
   // Target frame rate for frame dropper should be the same as previous == 15.
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, medium spatial: high frame rate, so 1/2 temporal expected.
@@ -209,8 +208,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
                       kSpatialMedium);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Medium motion, medium spatial: high frame rate, so 2/3 temporal expected.
@@ -218,8 +217,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
                       kSpatialMedium);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(8, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 }
 
 // Rate mis-match is high, and we have over-shooting.
@@ -249,16 +248,16 @@ TEST_F(QmSelectTest, DownActionHighRateMMOvershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 }
 
 // Rate mis-match is high, target rate is below max for down-sampling,
@@ -288,16 +287,16 @@ TEST_F(QmSelectTest, NoActionHighRateMMUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 }
 
 // Buffer is underflowing, and target rate is below max for down-sampling,
@@ -332,16 +331,16 @@ TEST_F(QmSelectTest, DownActionBufferUnderflow) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 }
 
 // Target rate is below max for down-sampling, but buffer level is stable,
@@ -376,16 +375,16 @@ TEST_F(QmSelectTest, NoActionBufferStable) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 }
 
 // Very low rate, but no spatial down-sampling below some size (QCIF).
@@ -414,8 +413,8 @@ TEST_F(QmSelectTest, LimitDownSpatialAction) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 176, 144,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 176, 144, 30.0f));
 }
 
 // Very low rate, but no frame reduction below some frame_rate (8fps).
@@ -445,8 +444,8 @@ TEST_F(QmSelectTest, LimitDownTemporalAction) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      8.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 8.0f));
 }
 
 // Two stages: spatial down-sample and then back up spatially,
@@ -468,7 +467,7 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -476,8 +475,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Reset and go up in rate: expected to go back up, in 2 stages of 3/4.
   qm_resolution_->ResetRates();
@@ -493,8 +492,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -522,7 +521,7 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -530,8 +529,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Reset rates and simulate under-shooting scenario.: expect to go back up.
   // Goes up spatially in two stages for 1/2x1/2 down-sampling.
@@ -548,8 +547,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -577,7 +576,7 @@ TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -585,8 +584,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Reset and simulate large rate mis-match: expect no action to go back up.
   qm_resolution_->ResetRates();
@@ -601,8 +600,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 320, 240, 30.0f));
 }
 
 // Two stages: temporally down-sample and then back up temporally,
@@ -632,8 +631,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporal) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   // Reset rates and go up in rate: expect to go back up.
   qm_resolution_->ResetRates();
@@ -646,8 +645,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporal) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480, 30.0f));
 }
 
 // Two stages: temporal down-sample and then back up temporally, since encoder
@@ -669,7 +668,7 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // Low motion, high spatial.
@@ -677,8 +676,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   // Reset rates and simulate under-shooting scenario.: expect to go back up.
   qm_resolution_->ResetRates();
@@ -691,8 +690,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480, 30.0f));
 }
 
 // Two stages: temporal down-sample and then no action to go up,
@@ -736,8 +735,8 @@ TEST_F(QmSelectTest, 2StageDownTemporalNoActionUp) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      15.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 15.0f));
 }
 // 3 stages: spatial down-sample, followed by temporal down-sample,
 // and then go up to full state, as encoding rate has increased.
@@ -766,8 +765,8 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Change content data: expect temporal down-sample.
   qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
@@ -780,7 +779,7 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
   uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
   UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                    fraction_lost2, 5);
+                   fraction_lost2, 5);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // Low motion, high spatial.
@@ -788,8 +787,8 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
 
   // Reset rates and go high up in rate: expect to go back up both spatial
   // and temporally. The 1/2x1/2 spatial is undone in two stages.
@@ -806,8 +805,8 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f,
-                                      480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f, 480,
+                                      360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -842,8 +841,8 @@ TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 640, 360,
-              30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 640, 360, 30.0f));
 
   // Reset and lower rates to get another spatial action (3/4x3/4).
   // Lower the frame rate for spatial to be selected again.
@@ -865,8 +864,8 @@ TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 270, 10.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 270, 10.0f));
 
   // Reset and go to very low rate: no action should be taken,
   // we went down too much already.
@@ -883,8 +882,8 @@ TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 480, 270,
-                                      10.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 480, 270, 10.0f));
 }
 
 // Multiple down-sampling stages and then undo all of them.
@@ -917,8 +916,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
   // Go down 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -936,8 +935,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360, 20.5f));
 
   // Go down 3/4x3/4 spatial:
   qm_resolution_->UpdateCodecParameters(20.0f, 480, 360);
@@ -947,7 +946,7 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   int incoming_frame_rate3[] = {20, 20, 20, 20, 20};
   uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
   UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
-                    fraction_lost3, 5);
+                   fraction_lost3, 5);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -957,8 +956,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
   // The two spatial actions of 3/4x3/4 are converted to 1/2x1/2,
   // so scale factor is 2.0.
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      20.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 20.0f));
 
   // Reset rates and go high up in rate: expect to go up:
   // 1/2x1x2 spatial and 1/2 temporally.
@@ -1018,8 +1017,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Go down 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
@@ -1039,8 +1038,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
 
   // Go up 2/3 temporally.
   qm_resolution_->UpdateCodecParameters(20.0f, 320, 240);
@@ -1076,8 +1075,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
 
   // Go up spatial and temporal. Spatial undoing is done in 2 stages.
   qm_resolution_->UpdateCodecParameters(20.5f, 320, 240);
@@ -1092,8 +1091,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
 
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f,
-                                      480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f, 480,
+                                      360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -1131,8 +1130,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory3) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   // Go down 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
@@ -1151,8 +1150,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory3) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360, 20.5f));
 
   // Go up 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(20.5f, 480, 360);
@@ -1184,8 +1183,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory3) {
 
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f,
-                                      1.0f, 640, 480, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
+                                      640, 480, 30.0f));
 }
 
 // Two stages of 3/4x3/4 converted to one stage of 1/2x1/2.
@@ -1215,8 +1214,8 @@ TEST_F(QmSelectTest, ConvertThreeQuartersToOneHalf) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   // Set rates to go down another 3/4 spatial. Should be converted ton 1/2.
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
@@ -1235,8 +1234,8 @@ TEST_F(QmSelectTest, ConvertThreeQuartersToOneHalf) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 }
 
 void QmSelectTest::InitQmNativeData(float initial_bit_rate,
@@ -1244,11 +1243,9 @@ void QmSelectTest::InitQmNativeData(float initial_bit_rate,
                                     int native_width,
                                     int native_height,
                                     int num_layers) {
-  EXPECT_EQ(0, qm_resolution_->Initialize(initial_bit_rate,
-                                          user_frame_rate,
-                                          native_width,
-                                          native_height,
-                                          num_layers));
+  EXPECT_EQ(
+      0, qm_resolution_->Initialize(initial_bit_rate, user_frame_rate,
+                                    native_width, native_height, num_layers));
 }
 
 void QmSelectTest::UpdateQmContentData(float motion_metric,
@@ -1281,8 +1278,7 @@ void QmSelectTest::UpdateQmRateData(int* target_rate,
     float encoder_sent_rate_update = encoder_sent_rate[i];
     float incoming_frame_rate_update = incoming_frame_rate[i];
     uint8_t fraction_lost_update = fraction_lost[i];
-    qm_resolution_->UpdateRates(target_rate_update,
-                                encoder_sent_rate_update,
+    qm_resolution_->UpdateRates(target_rate_update, encoder_sent_rate_update,
                                 incoming_frame_rate_update,
                                 fraction_lost_update);
   }
diff --git a/webrtc/modules/video_coding/main/source/receiver.cc b/webrtc/modules/video_coding/receiver.cc
index 0707a9c3cd..fa2a2dca29 100644
--- a/webrtc/modules/video_coding/main/source/receiver.cc
+++ b/webrtc/modules/video_coding/receiver.cc
@@ -8,18 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/receiver.h"
+#include "webrtc/modules/video_coding/receiver.h"
 
 #include <assert.h>
 
 #include <cstdlib>
+#include <utility>
+#include <vector>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/media_opt_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -40,9 +42,9 @@ VCMReceiver::VCMReceiver(VCMTiming* timing,
                          rtc::scoped_ptr<EventWrapper> jitter_buffer_event)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       clock_(clock),
-      jitter_buffer_(clock_, jitter_buffer_event.Pass()),
+      jitter_buffer_(clock_, std::move(jitter_buffer_event)),
       timing_(timing),
-      render_wait_event_(receiver_event.Pass()),
+      render_wait_event_(std::move(receiver_event)),
       max_video_delay_ms_(kMaxVideoDelayMs) {
   Reset();
 }
@@ -71,8 +73,8 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
   // Insert the packet into the jitter buffer. The packet can either be empty or
   // contain media at this point.
   bool retransmitted = false;
-  const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
-                                                             &retransmitted);
+  const VCMFrameBufferEnum ret =
+      jitter_buffer_.InsertPacket(packet, &retransmitted);
   if (ret == kOldPacket) {
     return VCM_OK;
   } else if (ret == kFlushIndicator) {
@@ -95,13 +97,13 @@ void VCMReceiver::TriggerDecoderShutdown() {
 }
 
 VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
-                                               int64_t& next_render_time_ms,
-                                               bool render_timing) {
+                                               int64_t* next_render_time_ms,
+                                               bool prefer_late_decoding) {
   const int64_t start_time_ms = clock_->TimeInMilliseconds();
   uint32_t frame_timestamp = 0;
   // Exhaust wait time to get a complete frame for decoding.
-  bool found_frame = jitter_buffer_.NextCompleteTimestamp(
-      max_wait_time_ms, &frame_timestamp);
+  bool found_frame =
+      jitter_buffer_.NextCompleteTimestamp(max_wait_time_ms, &frame_timestamp);
 
   if (!found_frame)
     found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
@@ -113,14 +115,14 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
   timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
   const int64_t now_ms = clock_->TimeInMilliseconds();
   timing_->UpdateCurrentDelay(frame_timestamp);
-  next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
+  *next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
   // Check render timing.
   bool timing_error = false;
   // Assume that render timing errors are due to changes in the video stream.
-  if (next_render_time_ms < 0) {
+  if (*next_render_time_ms < 0) {
     timing_error = true;
-  } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
-    int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms));
+  } else if (std::abs(*next_render_time_ms - now_ms) > max_video_delay_ms_) {
+    int frame_delay = static_cast<int>(std::abs(*next_render_time_ms - now_ms));
     LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
                     << "delay bounds (" << frame_delay << " > "
                     << max_video_delay_ms_
@@ -140,14 +142,15 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
     return NULL;
   }
 
-  if (!render_timing) {
+  if (prefer_late_decoding) {
     // Decode frame as close as possible to the render timestamp.
-    const int32_t available_wait_time = max_wait_time_ms -
+    const int32_t available_wait_time =
+        max_wait_time_ms -
         static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
-    uint16_t new_max_wait_time = static_cast<uint16_t>(
-        VCM_MAX(available_wait_time, 0));
+    uint16_t new_max_wait_time =
+        static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
     uint32_t wait_time_ms = timing_->MaxWaitingTime(
-        next_render_time_ms, clock_->TimeInMilliseconds());
+        *next_render_time_ms, clock_->TimeInMilliseconds());
     if (new_max_wait_time < wait_time_ms) {
       // We're not allowed to wait until the frame is supposed to be rendered,
       // waiting as long as we're allowed to avoid busy looping, and then return
@@ -164,9 +167,9 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
   if (frame == NULL) {
     return NULL;
   }
-  frame->SetRenderTime(next_render_time_ms);
-  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
-                          "SetRenderTS", "render_time", next_render_time_ms);
+  frame->SetRenderTime(*next_render_time_ms);
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
+                          "render_time", *next_render_time_ms);
   if (!frame->Complete()) {
     // Update stats for incomplete frames.
     bool retransmitted = false;
@@ -186,8 +189,7 @@ void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
   jitter_buffer_.ReleaseFrame(frame);
 }
 
-void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
-                                    uint32_t* framerate) {
+void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
   assert(bitrate);
   assert(framerate);
   jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
@@ -209,8 +211,7 @@ void VCMReceiver::SetNackMode(VCMNackMode nackMode,
 void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
                                   int max_packet_age_to_nack,
                                   int max_incomplete_time_ms) {
-  jitter_buffer_.SetNackSettings(max_nack_list_size,
-                                 max_packet_age_to_nack,
+  jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
                                  max_incomplete_time_ms);
 }
 
diff --git a/webrtc/modules/video_coding/main/source/receiver.h b/webrtc/modules/video_coding/receiver.h
index e2515d438f..ff0eef8a6a 100644
--- a/webrtc/modules/video_coding/main/source/receiver.h
+++ b/webrtc/modules/video_coding/receiver.h
@@ -8,15 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_RECEIVER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_RECEIVER_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
 
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
+#include <vector>
+
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/timing.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 
@@ -25,9 +27,7 @@ class VCMEncodedFrame;
 
 class VCMReceiver {
  public:
-  VCMReceiver(VCMTiming* timing,
-              Clock* clock,
-              EventFactory* event_factory);
+  VCMReceiver(VCMTiming* timing, Clock* clock, EventFactory* event_factory);
 
   // Using this constructor, you can specify a different event factory for the
   // jitter buffer. Useful for unit tests when you want to simulate incoming
@@ -46,8 +46,8 @@ class VCMReceiver {
                        uint16_t frame_width,
                        uint16_t frame_height);
   VCMEncodedFrame* FrameForDecoding(uint16_t max_wait_time_ms,
-                                    int64_t& next_render_time_ms,
-                                    bool render_timing = true);
+                                    int64_t* next_render_time_ms,
+                                    bool prefer_late_decoding);
   void ReleaseFrame(VCMEncodedFrame* frame);
   void ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate);
   uint32_t DiscardedPackets() const;
@@ -89,4 +89,4 @@ class VCMReceiver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_RECEIVER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
diff --git a/webrtc/modules/video_coding/main/source/receiver_unittest.cc b/webrtc/modules/video_coding/receiver_unittest.cc
index 359b241e72..1f3a144bad 100644
--- a/webrtc/modules/video_coding/main/source/receiver_unittest.cc
+++ b/webrtc/modules/video_coding/receiver_unittest.cc
@@ -11,14 +11,16 @@
 
 #include <list>
 #include <queue>
+#include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/receiver.h"
-#include "webrtc/modules/video_coding/main/source/test/stream_generator.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/receiver.h"
+#include "webrtc/modules/video_coding/test/stream_generator.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
@@ -33,14 +35,11 @@ class TestVCMReceiver : public ::testing::Test {
       : clock_(new SimulatedClock(0)),
         timing_(clock_.get()),
         receiver_(&timing_, clock_.get(), &event_factory_) {
-
-    stream_generator_.reset(new
-        StreamGenerator(0, clock_->TimeInMilliseconds()));
+    stream_generator_.reset(
+        new StreamGenerator(0, clock_->TimeInMilliseconds()));
   }
 
-  virtual void SetUp() {
-    receiver_.Reset();
-  }
+  virtual void SetUp() { receiver_.Reset(); }
 
   int32_t InsertPacket(int index) {
     VCMPacket packet;
@@ -78,7 +77,7 @@ class TestVCMReceiver : public ::testing::Test {
   bool DecodeNextFrame() {
     int64_t render_time_ms = 0;
     VCMEncodedFrame* frame =
-        receiver_.FrameForDecoding(0, render_time_ms, false);
+        receiver_.FrameForDecoding(0, &render_time_ms, false);
     if (!frame)
       return false;
     receiver_.ReleaseFrame(frame);
@@ -115,7 +114,7 @@ TEST_F(TestVCMReceiver, RenderBufferSize_SkipToKeyFrame) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
   }
   EXPECT_EQ((kNumOfFrames - 1) * kDefaultFramePeriodMs,
-      receiver_.RenderBufferSizeMs());
+            receiver_.RenderBufferSizeMs());
 }
 
 TEST_F(TestVCMReceiver, RenderBufferSize_NotAllComplete) {
@@ -131,7 +130,7 @@ TEST_F(TestVCMReceiver, RenderBufferSize_NotAllComplete) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
   }
   EXPECT_EQ((num_of_frames - 1) * kDefaultFramePeriodMs,
-      receiver_.RenderBufferSizeMs());
+            receiver_.RenderBufferSizeMs());
 }
 
 TEST_F(TestVCMReceiver, RenderBufferSize_NoKeyFrame) {
@@ -141,7 +140,8 @@ TEST_F(TestVCMReceiver, RenderBufferSize_NoKeyFrame) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
   }
   int64_t next_render_time_ms = 0;
-  VCMEncodedFrame* frame = receiver_.FrameForDecoding(10, next_render_time_ms);
+  VCMEncodedFrame* frame =
+      receiver_.FrameForDecoding(10, &next_render_time_ms, false);
   EXPECT_TRUE(frame == NULL);
   receiver_.ReleaseFrame(frame);
   EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
@@ -159,7 +159,7 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_Empty) {
   const int kMaxNonDecodableDuration = 500;
   const int kMinDelayMs = 500;
   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
+                            kMaxNonDecodableDuration);
   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
   // Advance time until it's time to decode the key frame.
   clock_->AdvanceTimeMilliseconds(kMinDelayMs);
@@ -176,7 +176,7 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_NoKeyFrame) {
   const int kMaxPacketAgeToNack = 1000;
   const int kMaxNonDecodableDuration = 500;
   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
+                            kMaxNonDecodableDuration);
   const int kNumFrames = kDefaultFrameRate * kMaxNonDecodableDuration / 1000;
   for (int i = 0; i < kNumFrames; ++i) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
@@ -192,24 +192,23 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_OneIncomplete) {
   const size_t kMaxNackListSize = 1000;
   const int kMaxPacketAgeToNack = 1000;
   const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
   const int kMinDelayMs = 500;
   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
+                            kMaxNonDecodableDuration);
   receiver_.SetMinReceiverDelay(kMinDelayMs);
   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
   // Insert an incomplete frame.
   EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
   // Insert enough frames to have too long non-decodable sequence.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames;
-       ++i) {
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
   }
   // Advance time until it's time to decode the key frame.
   clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
+                                  key_frame_inserted);
   EXPECT_TRUE(DecodeNextFrame());
   // Make sure we get a key frame request.
   bool request_key_frame = false;
@@ -223,11 +222,11 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger) {
   const size_t kMaxNackListSize = 1000;
   const int kMaxPacketAgeToNack = 1000;
   const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
   const int kMinDelayMs = 500;
   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
+                            kMaxNonDecodableDuration);
   receiver_.SetMinReceiverDelay(kMinDelayMs);
   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
@@ -235,13 +234,12 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger) {
   EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
   // Insert all but one frame to not trigger a key frame request due to
   // too long duration of non-decodable frames.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames - 1;
-       ++i) {
+  for (int i = 0; i < kMaxNonDecodableDurationFrames - 1; ++i) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
   }
   // Advance time until it's time to decode the key frame.
   clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
+                                  key_frame_inserted);
   EXPECT_TRUE(DecodeNextFrame());
   // Make sure we don't get a key frame request since we haven't generated
   // enough frames.
@@ -256,25 +254,24 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger2) {
   const size_t kMaxNackListSize = 1000;
   const int kMaxPacketAgeToNack = 1000;
   const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
   const int kMinDelayMs = 500;
   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
+                            kMaxNonDecodableDuration);
   receiver_.SetMinReceiverDelay(kMinDelayMs);
   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
   // Insert enough frames to have too long non-decodable sequence, except that
   // we don't have any losses.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames;
-       ++i) {
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
   }
   // Insert an incomplete frame.
   EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
   // Advance time until it's time to decode the key frame.
   clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
+                                  key_frame_inserted);
   EXPECT_TRUE(DecodeNextFrame());
   // Make sure we don't get a key frame request since the non-decodable duration
   // is only one frame.
@@ -289,25 +286,24 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_KeyFrameAfterIncompleteFrames) {
   const size_t kMaxNackListSize = 1000;
   const int kMaxPacketAgeToNack = 1000;
   const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
   const int kMinDelayMs = 500;
   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
+                            kMaxNonDecodableDuration);
   receiver_.SetMinReceiverDelay(kMinDelayMs);
   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
   // Insert an incomplete frame.
   EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
   // Insert enough frames to have too long non-decodable sequence.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames;
-       ++i) {
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
     EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
   }
   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
   // Advance time until it's time to decode the key frame.
   clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
+                                  key_frame_inserted);
   EXPECT_TRUE(DecodeNextFrame());
   // Make sure we don't get a key frame request since we have a key frame
   // in the list.
@@ -338,7 +334,7 @@ class SimulatedClockWithFrames : public SimulatedClock {
   // Return true if some frame arrives between now and now+|milliseconds|.
   bool AdvanceTimeMilliseconds(int64_t milliseconds, bool stop_on_frame) {
     return AdvanceTimeMicroseconds(milliseconds * 1000, stop_on_frame);
-  };
+  }
 
   bool AdvanceTimeMicroseconds(int64_t microseconds, bool stop_on_frame) {
     int64_t start_time = TimeInMicroseconds();
@@ -362,7 +358,7 @@ class SimulatedClockWithFrames : public SimulatedClock {
       SimulatedClock::AdvanceTimeMicroseconds(end_time - TimeInMicroseconds());
     }
     return frame_injected;
-  };
+  }
 
   // Input timestamps are in unit Milliseconds.
   // And |arrive_timestamps| must be positive and in increasing order.
@@ -429,7 +425,7 @@ class FrameInjectEvent : public EventWrapper {
 
   bool Set() override { return true; }
 
-  EventTypeWrapper Wait(unsigned long max_time) override {
+  EventTypeWrapper Wait(unsigned long max_time) override {  // NOLINT
     if (clock_->AdvanceTimeMilliseconds(max_time, stop_on_frame_) &&
         stop_on_frame_) {
       return EventTypeWrapper::kEventSignaled;
@@ -445,7 +441,6 @@ class FrameInjectEvent : public EventWrapper {
 
 class VCMReceiverTimingTest : public ::testing::Test {
  protected:
-
   VCMReceiverTimingTest()
 
       : clock_(&stream_generator_, &receiver_),
@@ -458,7 +453,6 @@ class VCMReceiverTimingTest : public ::testing::Test {
             rtc::scoped_ptr<EventWrapper>(
                 new FrameInjectEvent(&clock_, true))) {}
 
-
   virtual void SetUp() { receiver_.Reset(); }
 
   SimulatedClockWithFrames clock_;
@@ -504,7 +498,7 @@ TEST_F(VCMReceiverTimingTest, FrameForDecoding) {
   while (num_frames_return < kNumFrames) {
     int64_t start_time = clock_.TimeInMilliseconds();
     VCMEncodedFrame* frame =
-        receiver_.FrameForDecoding(kMaxWaitTime, next_render_time, false);
+        receiver_.FrameForDecoding(kMaxWaitTime, &next_render_time, false);
     int64_t end_time = clock_.TimeInMilliseconds();
 
     // In any case the FrameForDecoding should not wait longer than
@@ -523,4 +517,59 @@ TEST_F(VCMReceiverTimingTest, FrameForDecoding) {
   }
 }
 
+// Test whether VCMReceiver::FrameForDecoding handles parameter
+// |prefer_late_decoding| and |max_wait_time_ms| correctly:
+// 1. The function execution should never take more than |max_wait_time_ms|.
+// 2. If the function exit before now + |max_wait_time_ms|, a frame must be
+//    returned and the end time must be equal to the render timestamp - delay
+//    for decoding and rendering.
+TEST_F(VCMReceiverTimingTest, FrameForDecodingPreferLateDecoding) {
+  const size_t kNumFrames = 100;
+  const int kFramePeriod = 40;
+
+  int64_t arrive_timestamps[kNumFrames];
+  int64_t render_timestamps[kNumFrames];
+  int64_t next_render_time;
+
+  int render_delay_ms;
+  int max_decode_ms;
+  int dummy;
+  timing_.GetTimings(&dummy, &max_decode_ms, &dummy, &dummy, &dummy, &dummy,
+                     &render_delay_ms);
+
+  // Construct test samples.
+  // render_timestamps are the timestamps stored in the Frame;
+  // arrive_timestamps controls when the Frame packet got received.
+  for (size_t i = 0; i < kNumFrames; i++) {
+    // Preset frame rate to 25Hz.
+    // But we add a reasonable deviation to arrive_timestamps to mimic Internet
+    // fluctuation.
+    arrive_timestamps[i] =
+        (i + 1) * kFramePeriod + (i % 10) * ((i % 2) ? 1 : -1);
+    render_timestamps[i] = (i + 1) * kFramePeriod;
+  }
+
+  clock_.SetFrames(arrive_timestamps, render_timestamps, kNumFrames);
+
+  // Record how many frames we finally get out of the receiver.
+  size_t num_frames_return = 0;
+  const int64_t kMaxWaitTime = 30;
+  bool prefer_late_decoding = true;
+  while (num_frames_return < kNumFrames) {
+    int64_t start_time = clock_.TimeInMilliseconds();
+
+    VCMEncodedFrame* frame = receiver_.FrameForDecoding(
+        kMaxWaitTime, &next_render_time, prefer_late_decoding);
+    int64_t end_time = clock_.TimeInMilliseconds();
+    if (frame) {
+      EXPECT_EQ(frame->RenderTimeMs() - max_decode_ms - render_delay_ms,
+                end_time);
+      receiver_.ReleaseFrame(frame);
+      ++num_frames_return;
+    } else {
+      EXPECT_EQ(kMaxWaitTime, end_time - start_time);
+    }
+  }
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/rtt_filter.cc b/webrtc/modules/video_coding/rtt_filter.cc
new file mode 100644
index 0000000000..742f70f1c1
--- /dev/null
+++ b/webrtc/modules/video_coding/rtt_filter.cc
@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/rtt_filter.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "webrtc/modules/video_coding/internal_defines.h"
+
+namespace webrtc {
+
+VCMRttFilter::VCMRttFilter()
+    : _filtFactMax(35),
+      _jumpStdDevs(2.5),
+      _driftStdDevs(3.5),
+      _detectThreshold(kMaxDriftJumpCount) {
+  Reset();
+}
+
+VCMRttFilter& VCMRttFilter::operator=(const VCMRttFilter& rhs) {
+  if (this != &rhs) {
+    _gotNonZeroUpdate = rhs._gotNonZeroUpdate;
+    _avgRtt = rhs._avgRtt;
+    _varRtt = rhs._varRtt;
+    _maxRtt = rhs._maxRtt;
+    _filtFactCount = rhs._filtFactCount;
+    _jumpCount = rhs._jumpCount;
+    _driftCount = rhs._driftCount;
+    memcpy(_jumpBuf, rhs._jumpBuf, sizeof(_jumpBuf));
+    memcpy(_driftBuf, rhs._driftBuf, sizeof(_driftBuf));
+  }
+  return *this;
+}
+
+void VCMRttFilter::Reset() {
+  _gotNonZeroUpdate = false;
+  _avgRtt = 0;
+  _varRtt = 0;
+  _maxRtt = 0;
+  _filtFactCount = 1;
+  _jumpCount = 0;
+  _driftCount = 0;
+  memset(_jumpBuf, 0, kMaxDriftJumpCount);
+  memset(_driftBuf, 0, kMaxDriftJumpCount);
+}
+
+void VCMRttFilter::Update(int64_t rttMs) {
+  if (!_gotNonZeroUpdate) {
+    if (rttMs == 0) {
+      return;
+    }
+    _gotNonZeroUpdate = true;
+  }
+
+  // Sanity check
+  if (rttMs > 3000) {
+    rttMs = 3000;
+  }
+
+  double filtFactor = 0;
+  if (_filtFactCount > 1) {
+    filtFactor = static_cast<double>(_filtFactCount - 1) / _filtFactCount;
+  }
+  _filtFactCount++;
+  if (_filtFactCount > _filtFactMax) {
+    // This prevents filtFactor from going above
+    // (_filtFactMax - 1) / _filtFactMax,
+    // e.g., _filtFactMax = 50 => filtFactor = 49/50 = 0.98
+    _filtFactCount = _filtFactMax;
+  }
+  double oldAvg = _avgRtt;
+  double oldVar = _varRtt;
+  _avgRtt = filtFactor * _avgRtt + (1 - filtFactor) * rttMs;
+  _varRtt = filtFactor * _varRtt +
+            (1 - filtFactor) * (rttMs - _avgRtt) * (rttMs - _avgRtt);
+  _maxRtt = VCM_MAX(rttMs, _maxRtt);
+  if (!JumpDetection(rttMs) || !DriftDetection(rttMs)) {
+    // In some cases we don't want to update the statistics
+    _avgRtt = oldAvg;
+    _varRtt = oldVar;
+  }
+}
+
+bool VCMRttFilter::JumpDetection(int64_t rttMs) {
+  double diffFromAvg = _avgRtt - rttMs;
+  if (fabs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt)) {
+    int diffSign = (diffFromAvg >= 0) ? 1 : -1;
+    int jumpCountSign = (_jumpCount >= 0) ? 1 : -1;
+    if (diffSign != jumpCountSign) {
+      // Since the signs differ the samples currently
+      // in the buffer is useless as they represent a
+      // jump in a different direction.
+      _jumpCount = 0;
+    }
+    if (abs(_jumpCount) < kMaxDriftJumpCount) {
+      // Update the buffer used for the short time
+      // statistics.
+      // The sign of the diff is used for updating the counter since
+      // we want to use the same buffer for keeping track of when
+      // the RTT jumps down and up.
+      _jumpBuf[abs(_jumpCount)] = rttMs;
+      _jumpCount += diffSign;
+    }
+    if (abs(_jumpCount) >= _detectThreshold) {
+      // Detected an RTT jump
+      ShortRttFilter(_jumpBuf, abs(_jumpCount));
+      _filtFactCount = _detectThreshold + 1;
+      _jumpCount = 0;
+    } else {
+      return false;
+    }
+  } else {
+    _jumpCount = 0;
+  }
+  return true;
+}
+
+bool VCMRttFilter::DriftDetection(int64_t rttMs) {
+  if (_maxRtt - _avgRtt > _driftStdDevs * sqrt(_varRtt)) {
+    if (_driftCount < kMaxDriftJumpCount) {
+      // Update the buffer used for the short time
+      // statistics.
+      _driftBuf[_driftCount] = rttMs;
+      _driftCount++;
+    }
+    if (_driftCount >= _detectThreshold) {
+      // Detected an RTT drift
+      ShortRttFilter(_driftBuf, _driftCount);
+      _filtFactCount = _detectThreshold + 1;
+      _driftCount = 0;
+    }
+  } else {
+    _driftCount = 0;
+  }
+  return true;
+}
+
+void VCMRttFilter::ShortRttFilter(int64_t* buf, uint32_t length) {
+  if (length == 0) {
+    return;
+  }
+  _maxRtt = 0;
+  _avgRtt = 0;
+  for (uint32_t i = 0; i < length; i++) {
+    if (buf[i] > _maxRtt) {
+      _maxRtt = buf[i];
+    }
+    _avgRtt += buf[i];
+  }
+  _avgRtt = _avgRtt / static_cast<double>(length);
+}
+
+int64_t VCMRttFilter::RttMs() const {
+  return static_cast<int64_t>(_maxRtt + 0.5);
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/rtt_filter.h b/webrtc/modules/video_coding/rtt_filter.h
new file mode 100644
index 0000000000..f5de532cfc
--- /dev/null
+++ b/webrtc/modules/video_coding/rtt_filter.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class VCMRttFilter {
+ public:
+  VCMRttFilter();
+
+  VCMRttFilter& operator=(const VCMRttFilter& rhs);
+
+  // Resets the filter.
+  void Reset();
+  // Updates the filter with a new sample.
+  void Update(int64_t rttMs);
+  // A getter function for the current RTT level in ms.
+  int64_t RttMs() const;
+
+ private:
+  // The size of the drift and jump memory buffers
+  // and thus also the detection threshold for these
+  // detectors in number of samples.
+  enum { kMaxDriftJumpCount = 5 };
+  // Detects RTT jumps by comparing the difference between
+  // samples and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool JumpDetection(int64_t rttMs);
+  // Detects RTT drifts by comparing the difference between
+  // max and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool DriftDetection(int64_t rttMs);
+  // Computes the short time average and maximum of the vector buf.
+  void ShortRttFilter(int64_t* buf, uint32_t length);
+
+  bool _gotNonZeroUpdate;
+  double _avgRtt;
+  double _varRtt;
+  int64_t _maxRtt;
+  uint32_t _filtFactCount;
+  const uint32_t _filtFactMax;
+  const double _jumpStdDevs;
+  const double _driftStdDevs;
+  int32_t _jumpCount;
+  int32_t _driftCount;
+  const int32_t _detectThreshold;
+  int64_t _jumpBuf[kMaxDriftJumpCount];
+  int64_t _driftBuf[kMaxDriftJumpCount];
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
diff --git a/webrtc/modules/video_coding/main/source/session_info.cc b/webrtc/modules/video_coding/session_info.cc
index 9a1bc54e52..8701098639 100644
--- a/webrtc/modules/video_coding/main/source/session_info.cc
+++ b/webrtc/modules/video_coding/session_info.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/session_info.h"
+#include "webrtc/modules/video_coding/session_info.h"
 
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 namespace webrtc {
 
@@ -32,8 +32,7 @@ VCMSessionInfo::VCMSessionInfo()
       empty_seq_num_low_(-1),
       empty_seq_num_high_(-1),
       first_packet_seq_num_(-1),
-      last_packet_seq_num_(-1) {
-}
+      last_packet_seq_num_(-1) {}
 
 void VCMSessionInfo::UpdateDataPointers(const uint8_t* old_base_ptr,
                                         const uint8_t* new_base_ptr) {
@@ -88,8 +87,8 @@ bool VCMSessionInfo::LayerSync() const {
   if (packets_.front().codecSpecificHeader.codec == kRtpVideoVp8) {
     return packets_.front().codecSpecificHeader.codecHeader.VP8.layerSync;
   } else if (packets_.front().codecSpecificHeader.codec == kRtpVideoVp9) {
-    return
-        packets_.front().codecSpecificHeader.codecHeader.VP9.temporal_up_switch;
+    return packets_.front()
+        .codecSpecificHeader.codecHeader.VP9.temporal_up_switch;
   } else {
     return false;
   }
@@ -126,7 +125,7 @@ void VCMSessionInfo::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
       gof_info.temporal_up_switch[idx];
   packets_.front().codecSpecificHeader.codecHeader.VP9.num_ref_pics =
       gof_info.num_ref_pics[idx];
-  for (size_t i = 0; i < gof_info.num_ref_pics[idx]; ++i) {
+  for (uint8_t i = 0; i < gof_info.num_ref_pics[idx]; ++i) {
     packets_.front().codecSpecificHeader.codecHeader.VP9.pid_diff[i] =
         gof_info.pid_diff[idx][i];
   }
@@ -193,9 +192,7 @@ size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
     while (nalu_ptr < packet_buffer + packet.sizeBytes) {
       size_t length = BufferToUWord16(nalu_ptr);
       nalu_ptr += kLengthFieldLength;
-      frame_buffer_ptr += Insert(nalu_ptr,
-                                 length,
-                                 packet.insertStartCode,
+      frame_buffer_ptr += Insert(nalu_ptr, length, packet.insertStartCode,
                                  const_cast<uint8_t*>(frame_buffer_ptr));
       nalu_ptr += length;
     }
@@ -203,14 +200,12 @@ size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
     return packet.sizeBytes;
   }
   ShiftSubsequentPackets(
-      packet_it,
-      packet.sizeBytes +
-          (packet.insertStartCode ? kH264StartCodeLengthBytes : 0));
-
-  packet.sizeBytes = Insert(packet_buffer,
-                            packet.sizeBytes,
-                            packet.insertStartCode,
-                            const_cast<uint8_t*>(packet.dataPtr));
+      packet_it, packet.sizeBytes +
+                     (packet.insertStartCode ? kH264StartCodeLengthBytes : 0));
+
+  packet.sizeBytes =
+      Insert(packet_buffer, packet.sizeBytes, packet.insertStartCode,
+             const_cast<uint8_t*>(packet.dataPtr));
   return packet.sizeBytes;
 }
 
@@ -223,8 +218,7 @@ size_t VCMSessionInfo::Insert(const uint8_t* buffer,
     memcpy(frame_buffer, startCode, kH264StartCodeLengthBytes);
   }
   memcpy(frame_buffer + (insert_start_code ? kH264StartCodeLengthBytes : 0),
-         buffer,
-         length);
+         buffer, length);
   length += (insert_start_code ? kH264StartCodeLengthBytes : 0);
 
   return length;
@@ -276,13 +270,12 @@ void VCMSessionInfo::UpdateDecodableSession(const FrameData& frame_data) {
   // thresholds.
   const float kLowPacketPercentageThreshold = 0.2f;
   const float kHighPacketPercentageThreshold = 0.8f;
-  if (frame_data.rtt_ms < kRttThreshold
-      || frame_type_ == kVideoFrameKey
-      || !HaveFirstPacket()
-      || (NumPackets() <= kHighPacketPercentageThreshold
-                          * frame_data.rolling_average_packets_per_frame
-          && NumPackets() > kLowPacketPercentageThreshold
-                            * frame_data.rolling_average_packets_per_frame))
+  if (frame_data.rtt_ms < kRttThreshold || frame_type_ == kVideoFrameKey ||
+      !HaveFirstPacket() ||
+      (NumPackets() <= kHighPacketPercentageThreshold *
+                           frame_data.rolling_average_packets_per_frame &&
+       NumPackets() > kLowPacketPercentageThreshold *
+                          frame_data.rolling_average_packets_per_frame))
     return;
 
   decodable_ = true;
@@ -308,7 +301,7 @@ VCMSessionInfo::PacketIterator VCMSessionInfo::FindNaluEnd(
   // Find the end of the NAL unit.
   for (; packet_it != packets_.end(); ++packet_it) {
     if (((*packet_it).completeNALU == kNaluComplete &&
-        (*packet_it).sizeBytes > 0) ||
+         (*packet_it).sizeBytes > 0) ||
         // Found next NALU.
         (*packet_it).completeNALU == kNaluStart)
       return --packet_it;
@@ -348,7 +341,7 @@ size_t VCMSessionInfo::BuildVP8FragmentationHeader(
   memset(fragmentation->fragmentationLength, 0,
          kMaxVP8Partitions * sizeof(size_t));
   if (packets_.empty())
-      return new_length;
+    return new_length;
   PacketIterator it = FindNextPartitionBeginning(packets_.begin());
   while (it != packets_.end()) {
     const int partition_id =
@@ -371,7 +364,7 @@ size_t VCMSessionInfo::BuildVP8FragmentationHeader(
   // Set all empty fragments to start where the previous fragment ends,
   // and have zero length.
   if (fragmentation->fragmentationLength[0] == 0)
-      fragmentation->fragmentationOffset[0] = 0;
+    fragmentation->fragmentationOffset[0] = 0;
   for (int i = 1; i < fragmentation->fragmentationVectorSize; ++i) {
     if (fragmentation->fragmentationLength[i] == 0)
       fragmentation->fragmentationOffset[i] =
@@ -379,7 +372,7 @@ size_t VCMSessionInfo::BuildVP8FragmentationHeader(
           fragmentation->fragmentationLength[i - 1];
     assert(i == 0 ||
            fragmentation->fragmentationOffset[i] >=
-           fragmentation->fragmentationOffset[i - 1]);
+               fragmentation->fragmentationOffset[i - 1]);
   }
   assert(new_length <= frame_buffer_length);
   return new_length;
@@ -424,8 +417,8 @@ bool VCMSessionInfo::InSequence(const PacketIterator& packet_it,
   // If the two iterators are pointing to the same packet they are considered
   // to be in sequence.
   return (packet_it == prev_packet_it ||
-      (static_cast<uint16_t>((*prev_packet_it).seqNum + 1) ==
-          (*packet_it).seqNum));
+          (static_cast<uint16_t>((*prev_packet_it).seqNum + 1) ==
+           (*packet_it).seqNum));
 }
 
 size_t VCMSessionInfo::MakeDecodable() {
@@ -435,8 +428,7 @@ size_t VCMSessionInfo::MakeDecodable() {
   }
   PacketIterator it = packets_.begin();
   // Make sure we remove the first NAL unit if it's not decodable.
-  if ((*it).completeNALU == kNaluIncomplete ||
-      (*it).completeNALU == kNaluEnd) {
+  if ((*it).completeNALU == kNaluIncomplete || (*it).completeNALU == kNaluEnd) {
     PacketIterator nalu_end = FindNaluEnd(it);
     return_length += DeletePacketData(it, nalu_end);
     it = nalu_end;
@@ -445,7 +437,7 @@ size_t VCMSessionInfo::MakeDecodable() {
   // Take care of the rest of the NAL units.
   for (; it != packets_.end(); ++it) {
     bool start_of_nalu = ((*it).completeNALU == kNaluStart ||
-        (*it).completeNALU == kNaluComplete);
+                          (*it).completeNALU == kNaluComplete);
     if (!start_of_nalu && !InSequence(it, prev_it)) {
       // Found a sequence number gap due to packet loss.
       PacketIterator nalu_end = FindNaluEnd(it);
@@ -463,18 +455,15 @@ void VCMSessionInfo::SetNotDecodableIfIncomplete() {
   decodable_ = false;
 }
 
-bool
-VCMSessionInfo::HaveFirstPacket() const {
+bool VCMSessionInfo::HaveFirstPacket() const {
   return !packets_.empty() && (first_packet_seq_num_ != -1);
 }
 
-bool
-VCMSessionInfo::HaveLastPacket() const {
+bool VCMSessionInfo::HaveLastPacket() const {
   return !packets_.empty() && (last_packet_seq_num_ != -1);
 }
 
-bool
-VCMSessionInfo::session_nack() const {
+bool VCMSessionInfo::session_nack() const {
   return session_nack_;
 }
 
@@ -502,8 +491,8 @@ int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
       break;
 
   // Check for duplicate packets.
-  if (rit != packets_.rend() &&
-      (*rit).seqNum == packet.seqNum && (*rit).sizeBytes > 0)
+  if (rit != packets_.rend() && (*rit).seqNum == packet.seqNum &&
+      (*rit).sizeBytes > 0)
     return -2;
 
   if (packet.codec == kVideoCodecH264) {
@@ -572,8 +561,8 @@ void VCMSessionInfo::InformOfEmptyPacket(uint16_t seq_num) {
     empty_seq_num_high_ = seq_num;
   else
     empty_seq_num_high_ = LatestSequenceNumber(seq_num, empty_seq_num_high_);
-  if (empty_seq_num_low_ == -1 || IsNewerSequenceNumber(empty_seq_num_low_,
-                                                        seq_num))
+  if (empty_seq_num_low_ == -1 ||
+      IsNewerSequenceNumber(empty_seq_num_low_, seq_num))
     empty_seq_num_low_ = seq_num;
 }
 
diff --git a/webrtc/modules/video_coding/main/source/session_info.h b/webrtc/modules/video_coding/session_info.h
index 88071e19d5..e9ff25166d 100644
--- a/webrtc/modules/video_coding/main/source/session_info.h
+++ b/webrtc/modules/video_coding/session_info.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_SESSION_INFO_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_SESSION_INFO_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
 
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/packet.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -116,8 +116,7 @@ class VCMSessionInfo {
   PacketIterator FindPartitionEnd(PacketIterator it) const;
   static bool InSequence(const PacketIterator& it,
                          const PacketIterator& prev_it);
-  size_t InsertBuffer(uint8_t* frame_buffer,
-                      PacketIterator packetIterator);
+  size_t InsertBuffer(uint8_t* frame_buffer, PacketIterator packetIterator);
   size_t Insert(const uint8_t* buffer,
                 size_t length,
                 bool insert_start_code,
@@ -126,8 +125,7 @@ class VCMSessionInfo {
   PacketIterator FindNaluEnd(PacketIterator packet_iter) const;
   // Deletes the data of all packets between |start| and |end|, inclusively.
   // Note that this function doesn't delete the actual packets.
-  size_t DeletePacketData(PacketIterator start,
-                          PacketIterator end);
+  size_t DeletePacketData(PacketIterator start, PacketIterator end);
   void UpdateCompleteSession();
 
   // When enabled, determine if session is decodable, i.e. incomplete but
@@ -169,4 +167,4 @@ class VCMSessionInfo {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_SESSION_INFO_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
diff --git a/webrtc/modules/video_coding/main/source/session_info_unittest.cc b/webrtc/modules/video_coding/session_info_unittest.cc
index 58c352d3fc..4019d63a5f 100644
--- a/webrtc/modules/video_coding/main/source/session_info_unittest.cc
+++ b/webrtc/modules/video_coding/session_info_unittest.cc
@@ -11,9 +11,9 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/session_info.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/session_info.h"
 
 namespace webrtc {
 
@@ -81,7 +81,7 @@ class TestVP8Partitions : public TestSessionInfo {
               fragmentation_.fragmentationLength[partition_id]);
     for (int i = 0; i < packets_expected; ++i) {
       size_t packet_index = fragmentation_.fragmentationOffset[partition_id] +
-          i * packet_buffer_size();
+                            i * packet_buffer_size();
       if (packet_index + packet_buffer_size() > frame_buffer_size())
         return false;
       VerifyPacket(frame_buffer_ + packet_index, start_value + i);
@@ -122,8 +122,7 @@ class TestNackList : public TestSessionInfo {
     memset(seq_num_list_, 0, sizeof(seq_num_list_));
   }
 
-  void BuildSeqNumList(uint16_t low,
-                       uint16_t high) {
+  void BuildSeqNumList(uint16_t low, uint16_t high) {
     size_t i = 0;
     while (low != high + 1) {
       EXPECT_LT(i, kMaxSeqNumListLength);
@@ -173,14 +172,11 @@ TEST_F(TestSessionInfo, TestSimpleAPIs) {
   // To make things more difficult we will make sure to have a wrap here.
   packet_.isFirstPacket = false;
   packet_.markerBit = true;
-  packet_.seqNum  = 2;
+  packet_.seqNum = 2;
   packet_.sizeBytes = 0;
   packet_.frameType = kEmptyFrame;
-  EXPECT_EQ(0,
-            session_.InsertPacket(packet_,
-                                  frame_buffer_,
-                                  kNoErrors,
-                                  frame_data));
+  EXPECT_EQ(
+      0, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
 }
 
@@ -198,9 +194,8 @@ TEST_F(TestSessionInfo, NormalOperation) {
     packet_.seqNum += 1;
     FillPacket(i);
     ASSERT_EQ(packet_buffer_size(),
-              static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                        kNoErrors,
-                                                        frame_data)));
+              static_cast<size_t>(session_.InsertPacket(
+                  packet_, frame_buffer_, kNoErrors, frame_data)));
   }
 
   packet_.seqNum += 1;
@@ -223,9 +218,8 @@ TEST_F(TestSessionInfo, ErrorsEqualDecodableState) {
   packet_.markerBit = false;
   FillPacket(3);
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kWithErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kWithErrors, frame_data)));
   EXPECT_TRUE(session_.decodable());
 }
 
@@ -237,18 +231,16 @@ TEST_F(TestSessionInfo, SelectiveDecodableState) {
   frame_data.rolling_average_packets_per_frame = 11;
   frame_data.rtt_ms = 150;
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kSelectiveErrors, frame_data)));
   EXPECT_FALSE(session_.decodable());
 
   packet_.seqNum -= 1;
   FillPacket(0);
   packet_.isFirstPacket = true;
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kSelectiveErrors, frame_data)));
   EXPECT_TRUE(session_.decodable());
 
   packet_.isFirstPacket = false;
@@ -256,19 +248,17 @@ TEST_F(TestSessionInfo, SelectiveDecodableState) {
   for (int i = 2; i < 8; ++i) {
     packet_.seqNum += 1;
     FillPacket(i);
-  EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+    EXPECT_EQ(packet_buffer_size(),
+              static_cast<size_t>(session_.InsertPacket(
+                  packet_, frame_buffer_, kSelectiveErrors, frame_data)));
     EXPECT_TRUE(session_.decodable());
   }
 
   packet_.seqNum += 1;
   FillPacket(8);
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kSelectiveErrors, frame_data)));
   EXPECT_TRUE(session_.decodable());
 }
 
@@ -285,18 +275,14 @@ TEST_F(TestSessionInfo, OutOfBoundsPackets1PacketFrame) {
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0000;
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, SetMarkerBitOnce) {
@@ -311,10 +297,8 @@ TEST_F(TestSessionInfo, SetMarkerBitOnce) {
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, OutOfBoundsPacketsBase) {
@@ -331,10 +315,8 @@ TEST_F(TestSessionInfo, OutOfBoundsPacketsBase) {
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0006;
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
@@ -346,10 +328,8 @@ TEST_F(TestSessionInfo, OutOfBoundsPacketsBase) {
   packet_.isFirstPacket = false;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, OutOfBoundsPacketsWrap) {
@@ -379,20 +359,14 @@ TEST_F(TestSessionInfo, OutOfBoundsPacketsWrap) {
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3,
-            session_.InsertPacket(packet_,
-                                  frame_buffer_,
-                                  kNoErrors,
-                                  frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0006;
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3,
-            session_.InsertPacket(packet_,
-                                  frame_buffer_,
-                                  kNoErrors,
-                                  frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, OutOfBoundsOutOfOrder) {
@@ -417,10 +391,8 @@ TEST_F(TestSessionInfo, OutOfBoundsOutOfOrder) {
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0010;
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
@@ -440,10 +412,8 @@ TEST_F(TestSessionInfo, OutOfBoundsOutOfOrder) {
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestVP8Partitions, TwoPartitionsOneLoss) {
@@ -455,8 +425,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsOneLoss) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -505,8 +475,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsOneLoss2) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 1;
   FillPacket(1);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -567,8 +537,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsNoLossWrap) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0xfffd;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -629,8 +599,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsLossWrap) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0xfffd;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -682,7 +652,6 @@ TEST_F(TestVP8Partitions, TwoPartitionsLossWrap) {
   EXPECT_TRUE(VerifyPartition(1, 1, 2));
 }
 
-
 TEST_F(TestVP8Partitions, ThreePartitionsOneMissing) {
   // Partition 1  |Partition 2    | Partition 3
   // [ 1 ] [ 2 ]  |               | [ 5 ] | [ 6 ]
@@ -692,8 +661,8 @@ TEST_F(TestVP8Partitions, ThreePartitionsOneMissing) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 1;
   FillPacket(1);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -754,8 +723,8 @@ TEST_F(TestVP8Partitions, ThreePartitionsLossInSecond) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 1;
   FillPacket(1);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -767,8 +736,7 @@ TEST_F(TestVP8Partitions, ThreePartitionsLossInSecond) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber += 1;
   FillPacket(2);
-  packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                         packet_header_);
+  packet = new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -841,8 +809,8 @@ TEST_F(TestVP8Partitions, AggregationOverTwoPackets) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -892,10 +860,8 @@ TEST_F(TestNalUnits, OnlyReceivedEmptyPacket) {
   packet_.sizeBytes = 0;
   packet_.seqNum = 0;
   packet_.markerBit = false;
-  EXPECT_EQ(0, session_.InsertPacket(packet_,
-                                     frame_buffer_,
-                                     kNoErrors,
-                                     frame_data));
+  EXPECT_EQ(
+      0, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 
   EXPECT_EQ(0U, session_.MakeDecodable());
   EXPECT_EQ(0U, session_.SessionLength());
diff --git a/webrtc/modules/video_coding/main/test/plotJitterEstimate.m b/webrtc/modules/video_coding/test/plotJitterEstimate.m
index d6185f55da..d6185f55da 100644
--- a/webrtc/modules/video_coding/main/test/plotJitterEstimate.m
+++ b/webrtc/modules/video_coding/test/plotJitterEstimate.m
diff --git a/webrtc/modules/video_coding/main/test/plotReceiveTrace.m b/webrtc/modules/video_coding/test/plotReceiveTrace.m
index 4d262aa165..4d262aa165 100644
--- a/webrtc/modules/video_coding/main/test/plotReceiveTrace.m
+++ b/webrtc/modules/video_coding/test/plotReceiveTrace.m
diff --git a/webrtc/modules/video_coding/main/test/plotTimingTest.m b/webrtc/modules/video_coding/test/plotTimingTest.m
index 52a6f303cd..52a6f303cd 100644
--- a/webrtc/modules/video_coding/main/test/plotTimingTest.m
+++ b/webrtc/modules/video_coding/test/plotTimingTest.m
diff --git a/webrtc/modules/video_coding/main/test/receiver_tests.h b/webrtc/modules/video_coding/test/receiver_tests.h
index 6d7b7beeb5..d6bac07392 100644
--- a/webrtc/modules/video_coding/main/test/receiver_tests.h
+++ b/webrtc/modules/video_coding/test/receiver_tests.h
@@ -11,20 +11,20 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
 #define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
 
-#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/modules/video_coding/main/test/video_source.h"
-#include "webrtc/typedefs.h"
-
 #include <stdio.h>
 #include <string>
 
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/modules/video_coding/test/video_source.h"
+#include "webrtc/typedefs.h"
+
 class RtpDataCallback : public webrtc::NullRtpData {
  public:
-  RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
+  explicit RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
   virtual ~RtpDataCallback() {}
 
   int32_t OnReceivedPayloadData(
@@ -40,4 +40,4 @@ class RtpDataCallback : public webrtc::NullRtpData {
 
 int RtpPlay(const CmdArgs& args);
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
diff --git a/webrtc/modules/video_coding/main/test/release_test.h b/webrtc/modules/video_coding/test/release_test.h
index e90dcaef01..ab9b2159d9 100644
--- a/webrtc/modules/video_coding/main/test/release_test.h
+++ b/webrtc/modules/video_coding/test/release_test.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef RELEASE_TEST_H
-#define RELEASE_TEST_H
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
 
 int ReleaseTest();
 int ReleaseTestPart2();
 
-#endif
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/test/rtp_player.cc
index 6717cf227d..9b6490618c 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/test/rtp_player.cc
@@ -8,27 +8,27 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/rtp_player.h"
+#include "webrtc/modules/video_coding/test/rtp_player.h"
 
 #include <stdio.h>
 
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 #if 1
-# define DEBUG_LOG1(text, arg)
+#define DEBUG_LOG1(text, arg)
 #else
-# define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
+#define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
 #endif
 
 namespace webrtc {
@@ -41,7 +41,9 @@ enum {
 
 class RawRtpPacket {
  public:
-  RawRtpPacket(const uint8_t* data, size_t length, uint32_t ssrc,
+  RawRtpPacket(const uint8_t* data,
+               size_t length,
+               uint32_t ssrc,
                uint16_t seq_num)
       : data_(new uint8_t[length]),
         length_(length),
@@ -140,7 +142,7 @@ class LostPackets {
     CriticalSectionScoped cs(crit_sect_.get());
     int count = 0;
     for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
-        ++it) {
+         ++it) {
       if ((*it)->resend_time_ms() >= 0) {
         count++;
       }
@@ -164,7 +166,7 @@ class LostPackets {
     printf("Packets still lost: %zd\n", packets_.size());
     printf("Sequence numbers:\n");
     for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
-        ++it) {
+         ++it) {
       printf("%u, ", (*it)->seq_num());
     }
     printf("\n");
@@ -231,17 +233,14 @@ class SsrcHandlers {
         kDefaultTransmissionTimeOffsetExtensionId);
 
     for (PayloadTypesIterator it = payload_types_.begin();
-        it != payload_types_.end(); ++it) {
+         it != payload_types_.end(); ++it) {
       VideoCodec codec;
       memset(&codec, 0, sizeof(codec));
-      strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName)-1);
+      strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName) - 1);
       codec.plType = it->payload_type();
       codec.codecType = it->codec_type();
-      if (handler->rtp_module_->RegisterReceivePayload(codec.plName,
-                                                       codec.plType,
-                                                       90000,
-                                                       0,
-                                                       codec.maxBitrate) < 0) {
+      if (handler->rtp_module_->RegisterReceivePayload(
+              codec.plName, codec.plType, 90000, 0, codec.maxBitrate) < 0) {
         return -1;
       }
     }
@@ -267,7 +266,8 @@ class SsrcHandlers {
  private:
   class Handler : public RtpStreamInterface {
    public:
-    Handler(uint32_t ssrc, const PayloadTypes& payload_types,
+    Handler(uint32_t ssrc,
+            const PayloadTypes& payload_types,
             LostPackets* lost_packets)
         : rtp_header_parser_(RtpHeaderParser::Create()),
           rtp_payload_registry_(new RTPPayloadRegistry(
@@ -290,9 +290,7 @@ class SsrcHandlers {
     }
 
     virtual uint32_t ssrc() const { return ssrc_; }
-    virtual const PayloadTypes& payload_types() const {
-      return payload_types_;
-    }
+    virtual const PayloadTypes& payload_types() const { return payload_types_; }
 
     rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
     rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
@@ -351,8 +349,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
   virtual int NextPacket(int64_t time_now) {
     // Send any packets ready to be resent.
     for (RawRtpPacket* packet = lost_packets_.NextPacketToResend(time_now);
-         packet != NULL;
-         packet = lost_packets_.NextPacketToResend(time_now)) {
+         packet != NULL; packet = lost_packets_.NextPacketToResend(time_now)) {
       int ret = SendPacket(packet->data(), packet->length());
       if (ret > 0) {
         printf("Resend: %08x:%u\n", packet->ssrc(), packet->seq_num());
@@ -392,8 +389,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
       if (!packet_source_->NextPacket(&next_packet_)) {
         end_of_file_ = true;
         return 0;
-      }
-      else if (next_packet_.length == 0) {
+      } else if (next_packet_.length == 0) {
         return 0;
       }
     }
@@ -406,7 +402,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 
   virtual uint32_t TimeUntilNextPacket() const {
     int64_t time_left = (next_rtp_time_ - first_packet_rtp_time_) -
-        (clock_->TimeInMilliseconds() - first_packet_time_ms_);
+                        (clock_->TimeInMilliseconds() - first_packet_time_ms_);
     if (time_left < 0) {
       return 0;
     }
@@ -438,7 +434,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 
       if (no_loss_startup_ > 0) {
         no_loss_startup_--;
-      } else if ((rand() + 1.0)/(RAND_MAX + 1.0) < loss_rate_) {
+      } else if ((rand() + 1.0) / (RAND_MAX + 1.0) < loss_rate_) {  // NOLINT
         uint16_t seq_num = header.sequenceNumber;
         lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
         DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
@@ -470,9 +466,12 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 };
 
 RtpPlayerInterface* Create(const std::string& input_filename,
-    PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock,
-    const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms,
-    bool reordering) {
+                           PayloadSinkFactoryInterface* payload_sink_factory,
+                           Clock* clock,
+                           const PayloadTypes& payload_types,
+                           float loss_rate,
+                           int64_t rtt_ms,
+                           bool reordering) {
   rtc::scoped_ptr<test::RtpFileReader> packet_source(
       test::RtpFileReader::Create(test::RtpFileReader::kRtpDump,
                                   input_filename));
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.h b/webrtc/modules/video_coding/test/rtp_player.h
index 7459231416..e50fb9ac70 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.h
+++ b/webrtc/modules/video_coding/test/rtp_player.h
@@ -14,8 +14,8 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 class Clock;
@@ -24,12 +24,12 @@ namespace rtpplayer {
 
 class PayloadCodecTuple {
  public:
-  PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name,
+  PayloadCodecTuple(uint8_t payload_type,
+                    const std::string& codec_name,
                     VideoCodecType codec_type)
       : name_(codec_name),
         payload_type_(payload_type),
-        codec_type_(codec_type) {
-  }
+        codec_type_(codec_type) {}
 
   const std::string& name() const { return name_; }
   uint8_t payload_type() const { return payload_type_; }
@@ -87,11 +87,14 @@ class RtpPlayerInterface {
 };
 
 RtpPlayerInterface* Create(const std::string& inputFilename,
-    PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock,
-    const PayloadTypes& payload_types, float lossRate, int64_t rttMs,
-    bool reordering);
+                           PayloadSinkFactoryInterface* payloadSinkFactory,
+                           Clock* clock,
+                           const PayloadTypes& payload_types,
+                           float lossRate,
+                           int64_t rttMs,
+                           bool reordering);
 
 }  // namespace rtpplayer
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
diff --git a/webrtc/modules/video_coding/main/source/test/stream_generator.cc b/webrtc/modules/video_coding/test/stream_generator.cc
index b365d96dc0..167d55faff 100644
--- a/webrtc/modules/video_coding/main/source/test/stream_generator.cc
+++ b/webrtc/modules/video_coding/test/stream_generator.cc
@@ -8,22 +8,21 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/test/stream_generator.h"
+#include "webrtc/modules/video_coding/test/stream_generator.h"
 
 #include <string.h>
 
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 StreamGenerator::StreamGenerator(uint16_t start_seq_num, int64_t current_time)
-    : packets_(), sequence_number_(start_seq_num), start_time_(current_time) {
-}
+    : packets_(), sequence_number_(start_seq_num), start_time_(current_time) {}
 
 void StreamGenerator::Init(uint16_t start_seq_num, int64_t current_time) {
   packets_.clear();
@@ -41,8 +40,8 @@ void StreamGenerator::GenerateFrame(FrameType type,
     const int packet_size =
         (kFrameSize + num_media_packets / 2) / num_media_packets;
     bool marker_bit = (i == num_media_packets - 1);
-    packets_.push_back(GeneratePacket(
-        sequence_number_, timestamp, packet_size, (i == 0), marker_bit, type));
+    packets_.push_back(GeneratePacket(sequence_number_, timestamp, packet_size,
+                                      (i == 0), marker_bit, type));
     ++sequence_number_;
   }
   for (int i = 0; i < num_empty_packets; ++i) {
@@ -104,7 +103,9 @@ bool StreamGenerator::NextPacket(VCMPacket* packet) {
   return true;
 }
 
-void StreamGenerator::DropLastPacket() { packets_.pop_back(); }
+void StreamGenerator::DropLastPacket() {
+  packets_.pop_back();
+}
 
 uint16_t StreamGenerator::NextSequenceNumber() const {
   if (packets_.empty())
@@ -112,7 +113,9 @@ uint16_t StreamGenerator::NextSequenceNumber() const {
   return packets_.front().seqNum;
 }
 
-int StreamGenerator::PacketsRemaining() const { return packets_.size(); }
+int StreamGenerator::PacketsRemaining() const {
+  return packets_.size();
+}
 
 std::list<VCMPacket>::iterator StreamGenerator::GetPacketIterator(int index) {
   std::list<VCMPacket>::iterator it = packets_.begin();
diff --git a/webrtc/modules/video_coding/main/source/test/stream_generator.h b/webrtc/modules/video_coding/test/stream_generator.h
index 7902d16706..36b26db92e 100644
--- a/webrtc/modules/video_coding/main/source/test/stream_generator.h
+++ b/webrtc/modules/video_coding/test/stream_generator.h
@@ -7,13 +7,13 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
 
 #include <list>
 
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -69,4 +69,4 @@ class StreamGenerator {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
diff --git a/webrtc/modules/video_coding/main/test/subfigure.m b/webrtc/modules/video_coding/test/subfigure.m
index eadfcb69bd..eadfcb69bd 100644
--- a/webrtc/modules/video_coding/main/test/subfigure.m
+++ b/webrtc/modules/video_coding/test/subfigure.m
diff --git a/webrtc/modules/video_coding/main/test/test_util.cc b/webrtc/modules/video_coding/test/test_util.cc
index cd858da288..7ff663e395 100644
--- a/webrtc/modules/video_coding/main/test/test_util.cc
+++ b/webrtc/modules/video_coding/test/test_util.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 
 #include <assert.h>
 #include <math.h>
@@ -17,7 +17,7 @@
 #include <sstream>
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 CmdArgs::CmdArgs()
@@ -28,12 +28,12 @@ CmdArgs::CmdArgs()
       rtt(0),
       inputFile(webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv"),
       outputFile(webrtc::test::OutputPath() +
-          "video_coding_test_output_352x288.yuv") {
-}
+                 "video_coding_test_output_352x288.yuv") {}
 
 namespace {
 
-void SplitFilename(const std::string& filename, std::string* basename,
+void SplitFilename(const std::string& filename,
+                   std::string* basename,
                    std::string* extension) {
   assert(basename);
   assert(extension);
@@ -41,7 +41,7 @@ void SplitFilename(const std::string& filename, std::string* basename,
   std::string::size_type idx;
   idx = filename.rfind('.');
 
-  if(idx != std::string::npos) {
+  if (idx != std::string::npos) {
     *basename = filename.substr(0, idx);
     *extension = filename.substr(idx + 1);
   } else {
@@ -50,21 +50,24 @@ void SplitFilename(const std::string& filename, std::string* basename,
   }
 }
 
-std::string AppendWidthHeightCount(const std::string& filename, int width,
-                                   int height, int count) {
+std::string AppendWidthHeightCount(const std::string& filename,
+                                   int width,
+                                   int height,
+                                   int count) {
   std::string basename;
   std::string extension;
   SplitFilename(filename, &basename, &extension);
   std::stringstream ss;
-  ss << basename << "_" << count << "." <<  width << "_" << height << "." <<
-      extension;
+  ss << basename << "_" << count << "." << width << "_" << height << "."
+     << extension;
   return ss.str();
 }
 
 }  // namespace
 
 FileOutputFrameReceiver::FileOutputFrameReceiver(
-    const std::string& base_out_filename, uint32_t ssrc)
+    const std::string& base_out_filename,
+    uint32_t ssrc)
     : out_filename_(),
       out_file_(NULL),
       timing_file_(NULL),
@@ -80,8 +83,8 @@ FileOutputFrameReceiver::FileOutputFrameReceiver(
     SplitFilename(base_out_filename, &basename, &extension);
   }
   std::stringstream ss;
-  ss << basename << "_" << std::hex << std::setw(8) << std::setfill('0') <<
-      ssrc << "." << extension;
+  ss << basename << "_" << std::hex << std::setw(8) << std::setfill('0') << ssrc
+     << "." << extension;
   out_filename_ = ss.str();
 }
 
@@ -113,8 +116,8 @@ int32_t FileOutputFrameReceiver::FrameToRender(
     printf("New size: %dx%d\n", video_frame.width(), video_frame.height());
     width_ = video_frame.width();
     height_ = video_frame.height();
-    std::string filename_with_width_height = AppendWidthHeightCount(
-        out_filename_, width_, height_, count_);
+    std::string filename_with_width_height =
+        AppendWidthHeightCount(out_filename_, width_, height_, count_);
     ++count_;
     out_file_ = fopen(filename_with_width_height.c_str(), "wb");
     if (out_file_ == NULL) {
@@ -122,7 +125,7 @@ int32_t FileOutputFrameReceiver::FrameToRender(
     }
   }
   fprintf(timing_file_, "%u, %u\n", video_frame.timestamp(),
-      webrtc::MaskWord64ToUWord32(video_frame.render_time_ms()));
+          webrtc::MaskWord64ToUWord32(video_frame.render_time_ms()));
   if (PrintVideoFrame(video_frame, out_file_) < 0) {
     return -1;
   }
@@ -130,7 +133,7 @@ int32_t FileOutputFrameReceiver::FrameToRender(
 }
 
 webrtc::RtpVideoCodecTypes ConvertCodecType(const char* plname) {
-  if (strncmp(plname,"VP8" , 3) == 0) {
+  if (strncmp(plname, "VP8", 3) == 0) {
     return webrtc::kRtpVideoVp8;
   } else {
     // Default value.
diff --git a/webrtc/modules/video_coding/main/test/test_util.h b/webrtc/modules/video_coding/test/test_util.h
index 27f66fe011..45b88b9b50 100644
--- a/webrtc/modules/video_coding/main/test/test_util.h
+++ b/webrtc/modules/video_coding/test/test_util.h
@@ -18,8 +18,8 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 
 enum { kMaxNackListSize = 250 };
@@ -33,11 +33,13 @@ class NullEvent : public webrtc::EventWrapper {
 
   virtual bool Reset() { return true; }
 
-  virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {
+  virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {  // NOLINT
     return webrtc::kEventTimeout;
   }
 
-  virtual bool StartTimer(bool periodic, unsigned long time) { return true; }
+  virtual bool StartTimer(bool periodic, unsigned long time) {  // NOLINT
+    return true;
+  }
 
   virtual bool StopTimer() { return true; }
 };
@@ -46,9 +48,7 @@ class NullEventFactory : public webrtc::EventFactory {
  public:
   virtual ~NullEventFactory() {}
 
-  virtual webrtc::EventWrapper* CreateEvent() {
-    return new NullEvent;
-  }
+  virtual webrtc::EventWrapper* CreateEvent() { return new NullEvent; }
 };
 
 class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
@@ -57,7 +57,7 @@ class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
   virtual ~FileOutputFrameReceiver();
 
   // VCMReceiveCallback
-  virtual int32_t FrameToRender(webrtc::VideoFrame& video_frame);
+  virtual int32_t FrameToRender(webrtc::VideoFrame& video_frame);  // NOLINT
 
  private:
   std::string out_filename_;
@@ -83,4 +83,4 @@ class CmdArgs {
   std::string outputFile;
 };
 
-#endif
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
diff --git a/webrtc/modules/video_coding/main/test/tester_main.cc b/webrtc/modules/video_coding/test/tester_main.cc
index 2885f00bd5..33ca82007d 100644
--- a/webrtc/modules/video_coding/main/test/tester_main.cc
+++ b/webrtc/modules/video_coding/test/tester_main.cc
@@ -8,25 +8,27 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
 #include <stdlib.h>
 #include <string.h>
 
 #include "gflags/gflags.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/test/receiver_tests.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/test/receiver_tests.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 DEFINE_string(codec, "VP8", "Codec to use (VP8 or I420).");
 DEFINE_int32(width, 352, "Width in pixels of the frames in the input file.");
 DEFINE_int32(height, 288, "Height in pixels of the frames in the input file.");
 DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds.");
-DEFINE_string(input_filename, webrtc::test::ProjectRootPath() +
-              "/resources/foreman_cif.yuv", "Input file.");
-DEFINE_string(output_filename, webrtc::test::OutputPath() +
-              "video_coding_test_output_352x288.yuv", "Output file.");
+DEFINE_string(input_filename,
+              webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv",
+              "Input file.");
+DEFINE_string(output_filename,
+              webrtc::test::OutputPath() +
+                  "video_coding_test_output_352x288.yuv",
+              "Output file.");
 
-using namespace webrtc;
+namespace webrtc {
 
 /*
  * Build with EVENT_DEBUG defined
@@ -36,36 +38,37 @@ using namespace webrtc;
 int vcmMacrosTests = 0;
 int vcmMacrosErrors = 0;
 
-int ParseArguments(CmdArgs& args) {
-  args.width = FLAGS_width;
-  args.height = FLAGS_height;
-  if (args.width < 1 || args.height < 1) {
+int ParseArguments(CmdArgs* args) {
+  args->width = FLAGS_width;
+  args->height = FLAGS_height;
+  if (args->width < 1 || args->height < 1) {
     return -1;
   }
-  args.codecName = FLAGS_codec;
-  if (args.codecName == "VP8") {
-    args.codecType = kVideoCodecVP8;
-  } else if (args.codecName == "VP9") {
-    args.codecType = kVideoCodecVP9;
-  } else if (args.codecName == "I420") {
-    args.codecType = kVideoCodecI420;
+  args->codecName = FLAGS_codec;
+  if (args->codecName == "VP8") {
+    args->codecType = kVideoCodecVP8;
+  } else if (args->codecName == "VP9") {
+    args->codecType = kVideoCodecVP9;
+  } else if (args->codecName == "I420") {
+    args->codecType = kVideoCodecI420;
   } else {
-    printf("Invalid codec: %s\n", args.codecName.c_str());
+    printf("Invalid codec: %s\n", args->codecName.c_str());
     return -1;
   }
-  args.inputFile = FLAGS_input_filename;
-  args.outputFile = FLAGS_output_filename;
-  args.rtt = FLAGS_rtt;
+  args->inputFile = FLAGS_input_filename;
+  args->outputFile = FLAGS_output_filename;
+  args->rtt = FLAGS_rtt;
   return 0;
 }
+}  // namespace webrtc
 
-int main(int argc, char **argv) {
+int main(int argc, char** argv) {
   // Initialize WebRTC fileutils.h so paths to resources can be resolved.
   webrtc::test::SetExecutablePath(argv[0]);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   CmdArgs args;
-  if (ParseArguments(args) != 0) {
+  if (webrtc::ParseArguments(&args) != 0) {
     printf("Unable to parse input arguments\n");
     return -1;
   }
diff --git a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
index 2d874cd1bd..c9ec372f41 100644
--- a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
+++ b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
@@ -8,23 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h"
+#include "webrtc/modules/video_coding/test/vcm_payload_sink_factory.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 namespace rtpplayer {
 
-class VcmPayloadSinkFactory::VcmPayloadSink
-    : public PayloadSinkInterface,
-      public VCMPacketRequestCallback {
+class VcmPayloadSinkFactory::VcmPayloadSink : public PayloadSinkInterface,
+                                              public VCMPacketRequestCallback {
  public:
   VcmPayloadSink(VcmPayloadSinkFactory* factory,
                  RtpStreamInterface* stream,
@@ -43,9 +42,7 @@ class VcmPayloadSinkFactory::VcmPayloadSink
     vcm_->RegisterReceiveCallback(frame_receiver_.get());
   }
 
-  virtual ~VcmPayloadSink() {
-    factory_->Remove(this);
-  }
+  virtual ~VcmPayloadSink() { factory_->Remove(this); }
 
   // PayloadSinkInterface
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
@@ -136,14 +133,11 @@ PayloadSinkInterface* VcmPayloadSinkFactory::Create(
   }
 
   const PayloadTypes& plt = stream->payload_types();
-  for (PayloadTypesIterator it = plt.begin(); it != plt.end();
-      ++it) {
+  for (PayloadTypesIterator it = plt.begin(); it != plt.end(); ++it) {
     if (it->codec_type() != kVideoCodecULPFEC &&
         it->codec_type() != kVideoCodecRED) {
       VideoCodec codec;
-      if (VideoCodingModule::Codec(it->codec_type(), &codec) < 0) {
-        return NULL;
-      }
+      VideoCodingModule::Codec(it->codec_type(), &codec);
       codec.plType = it->payload_type();
       if (vcm->RegisterReceiveCodec(&codec, 1) < 0) {
         return NULL;
diff --git a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
index ec94bdc382..dae53b0c08 100644
--- a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h
+++ b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
@@ -8,13 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VCM_PAYLOAD_SINK_FACTORY_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_VCM_PAYLOAD_SINK_FACTORY_H_
+
 #include <string>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/test/rtp_player.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/test/rtp_player.h"
 
 class NullEventFactory;
 
@@ -26,9 +29,11 @@ namespace rtpplayer {
 class VcmPayloadSinkFactory : public PayloadSinkFactoryInterface {
  public:
   VcmPayloadSinkFactory(const std::string& base_out_filename,
-                        Clock* clock, bool protection_enabled,
+                        Clock* clock,
+                        bool protection_enabled,
                         VCMVideoProtection protection_method,
-                        int64_t rtt_ms, uint32_t render_delay_ms,
+                        int64_t rtt_ms,
+                        uint32_t render_delay_ms,
                         uint32_t min_playout_delay_ms);
   virtual ~VcmPayloadSinkFactory();
 
@@ -61,3 +66,5 @@ class VcmPayloadSinkFactory : public PayloadSinkFactoryInterface {
 };
 }  // namespace rtpplayer
 }  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_VCM_PAYLOAD_SINK_FACTORY_H_
diff --git a/webrtc/modules/video_coding/main/test/video_rtp_play.cc b/webrtc/modules/video_coding/test/video_rtp_play.cc
index 8460601bf5..cb092e381e 100644
--- a/webrtc/modules/video_coding/main/test/video_rtp_play.cc
+++ b/webrtc/modules/video_coding/test/video_rtp_play.cc
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/receiver_tests.h"
-#include "webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h"
+#include "webrtc/modules/video_coding/test/receiver_tests.h"
+#include "webrtc/modules/video_coding/test/vcm_payload_sink_factory.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -48,9 +48,9 @@ int RtpPlay(const CmdArgs& args) {
     output_file = webrtc::test::OutputPath() + "RtpPlay_decoded.yuv";
 
   webrtc::SimulatedClock clock(0);
-  webrtc::rtpplayer::VcmPayloadSinkFactory factory(output_file, &clock,
-      kConfigProtectionEnabled, kConfigProtectionMethod, kConfigRttMs,
-      kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
+  webrtc::rtpplayer::VcmPayloadSinkFactory factory(
+      output_file, &clock, kConfigProtectionEnabled, kConfigProtectionMethod,
+      kConfigRttMs, kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
   rtc::scoped_ptr<webrtc::rtpplayer::RtpPlayerInterface> rtp_player(
       webrtc::rtpplayer::Create(args.inputFile, &factory, &clock, payload_types,
                                 kConfigLossRate, kConfigRttMs,
@@ -63,7 +63,7 @@ int RtpPlay(const CmdArgs& args) {
   while ((ret = rtp_player->NextPacket(clock.TimeInMilliseconds())) == 0) {
     ret = factory.DecodeAndProcessAll(true);
     if (ret < 0 || (kConfigMaxRuntimeMs > -1 &&
-        clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
+                    clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
       break;
     }
     clock.AdvanceTimeMilliseconds(1);
diff --git a/webrtc/modules/video_coding/test/video_source.h b/webrtc/modules/video_coding/test/video_source.h
new file mode 100644
index 0000000000..19d7f50b26
--- /dev/null
+++ b/webrtc/modules/video_coding/test/video_source.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+
+#include <string>
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/typedefs.h"
+
+enum VideoSize {
+  kUndefined,
+  kSQCIF,    // 128*96       = 12 288
+  kQQVGA,    // 160*120      = 19 200
+  kQCIF,     // 176*144      = 25 344
+  kCGA,      // 320*200      = 64 000
+  kQVGA,     // 320*240      = 76 800
+  kSIF,      // 352*240      = 84 480
+  kWQVGA,    // 400*240      = 96 000
+  kCIF,      // 352*288      = 101 376
+  kW288p,    // 512*288      = 147 456 (WCIF)
+  k448p,     // 576*448      = 281 088
+  kVGA,      // 640*480      = 307 200
+  k432p,     // 720*432      = 311 040
+  kW432p,    // 768*432      = 331 776
+  k4SIF,     // 704*480      = 337 920
+  kW448p,    // 768*448      = 344 064
+  kNTSC,     // 720*480      = 345 600
+  kFW448p,   // 800*448      = 358 400
+  kWVGA,     // 800*480      = 384 000
+  k4CIF,     // 704*576      = 405 504
+  kSVGA,     // 800*600      = 480 000
+  kW544p,    // 960*544      = 522 240
+  kW576p,    // 1024*576     = 589 824 (W4CIF)
+  kHD,       // 960*720      = 691 200
+  kXGA,      // 1024*768     = 786 432
+  kWHD,      // 1280*720     = 921 600
+  kFullHD,   // 1440*1080    = 1 555 200
+  kWFullHD,  // 1920*1080    = 2 073 600
+
+  kNumberOfVideoSizes
+};
+
+class VideoSource {
+ public:
+  VideoSource();
+  VideoSource(std::string fileName,
+              VideoSize size,
+              float frameRate,
+              webrtc::VideoType type = webrtc::kI420);
+  VideoSource(std::string fileName,
+              uint16_t width,
+              uint16_t height,
+              float frameRate = 30,
+              webrtc::VideoType type = webrtc::kI420);
+
+  std::string GetFileName() const { return _fileName; }
+  uint16_t GetWidth() const { return _width; }
+  uint16_t GetHeight() const { return _height; }
+  webrtc::VideoType GetType() const { return _type; }
+  float GetFrameRate() const { return _frameRate; }
+  int GetWidthHeight(VideoSize size);
+
+  // Returns the filename with the path (including the leading slash) removed.
+  std::string GetName() const;
+
+  size_t GetFrameLength() const;
+
+ private:
+  std::string _fileName;
+  uint16_t _width;
+  uint16_t _height;
+  webrtc::VideoType _type;
+  float _frameRate;
+};
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
diff --git a/webrtc/modules/video_coding/main/source/timestamp_map.cc b/webrtc/modules/video_coding/timestamp_map.cc
index c68a5af7ba..97d2777658 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_map.cc
+++ b/webrtc/modules/video_coding/timestamp_map.cc
@@ -11,8 +11,8 @@
 #include <assert.h>
 #include <stdlib.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/timestamp_map.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/timestamp_map.h"
 
 namespace webrtc {
 
@@ -20,11 +20,9 @@ VCMTimestampMap::VCMTimestampMap(size_t capacity)
     : ring_buffer_(new TimestampDataTuple[capacity]),
       capacity_(capacity),
       next_add_idx_(0),
-      next_pop_idx_(0) {
-}
+      next_pop_idx_(0) {}
 
-VCMTimestampMap::~VCMTimestampMap() {
-}
+VCMTimestampMap::~VCMTimestampMap() {}
 
 void VCMTimestampMap::Add(uint32_t timestamp, VCMFrameInformation* data) {
   ring_buffer_[next_add_idx_].timestamp = timestamp;
@@ -62,4 +60,4 @@ VCMFrameInformation* VCMTimestampMap::Pop(uint32_t timestamp) {
 bool VCMTimestampMap::IsEmpty() const {
   return (next_add_idx_ == next_pop_idx_);
 }
-}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/timestamp_map.h b/webrtc/modules/video_coding/timestamp_map.h
index 3d6f1bca0f..435d05895c 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_map.h
+++ b/webrtc/modules/video_coding/timestamp_map.h
@@ -44,4 +44,4 @@ class VCMTimestampMap {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
diff --git a/webrtc/modules/video_coding/main/source/timing.cc b/webrtc/modules/video_coding/timing.cc
index 8d59135876..08dc307524 100644
--- a/webrtc/modules/video_coding/main/source/timing.cc
+++ b/webrtc/modules/video_coding/timing.cc
@@ -8,19 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/timing.h"
+#include "webrtc/modules/video_coding/timing.h"
 
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
+#include <algorithm>
+
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
 
-
 namespace webrtc {
 
-VCMTiming::VCMTiming(Clock* clock,
-                     VCMTiming* master_timing)
+VCMTiming::VCMTiming(Clock* clock, VCMTiming* master_timing)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       clock_(clock),
       master_(false),
@@ -62,14 +62,16 @@ void VCMTiming::UpdateHistograms() const {
   if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
     return;
   }
-  RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.DecodedFramesPerSecond",
+  RTC_HISTOGRAM_COUNTS_SPARSE_100(
+      "WebRTC.Video.DecodedFramesPerSecond",
       static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
-  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer",
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+      "WebRTC.Video.DelayedFramesToRenderer",
       num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
   if (num_delayed_decoded_frames_ > 0) {
-    RTC_HISTOGRAM_COUNTS_1000(
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(
         "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
-            sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
+        sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
   }
 }
 
@@ -118,8 +120,8 @@ void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
     // Not initialized, set current delay to target.
     current_delay_ms_ = target_delay_ms;
   } else if (target_delay_ms != current_delay_ms_) {
-    int64_t delay_diff_ms = static_cast<int64_t>(target_delay_ms) -
-        current_delay_ms_;
+    int64_t delay_diff_ms =
+        static_cast<int64_t>(target_delay_ms) - current_delay_ms_;
     // Never change the delay with more than 100 ms every second. If we're
     // changing the delay in too large steps we will get noticeable freezes. By
     // limiting the change we can increase the delay in smaller steps, which
@@ -128,11 +130,13 @@ void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
     int64_t max_change_ms = 0;
     if (frame_timestamp < 0x0000ffff && prev_frame_timestamp_ > 0xffff0000) {
       // wrap
-      max_change_ms = kDelayMaxChangeMsPerS * (frame_timestamp +
-          (static_cast<int64_t>(1) << 32) - prev_frame_timestamp_) / 90000;
+      max_change_ms = kDelayMaxChangeMsPerS *
+                      (frame_timestamp + (static_cast<int64_t>(1) << 32) -
+                       prev_frame_timestamp_) /
+                      90000;
     } else {
       max_change_ms = kDelayMaxChangeMsPerS *
-          (frame_timestamp - prev_frame_timestamp_) / 90000;
+                      (frame_timestamp - prev_frame_timestamp_) / 90000;
     }
     if (max_change_ms <= 0) {
       // Any changes less than 1 ms are truncated and
@@ -153,7 +157,7 @@ void VCMTiming::UpdateCurrentDelay(int64_t render_time_ms,
   CriticalSectionScoped cs(crit_sect_);
   uint32_t target_delay_ms = TargetDelayInternal();
   int64_t delayed_ms = actual_decode_time_ms -
-      (render_time_ms - MaxDecodeTimeMs() - render_delay_ms_);
+                       (render_time_ms - MaxDecodeTimeMs() - render_delay_ms_);
   if (delayed_ms < 0) {
     return;
   }
@@ -165,13 +169,13 @@ void VCMTiming::UpdateCurrentDelay(int64_t render_time_ms,
 }
 
 int32_t VCMTiming::StopDecodeTimer(uint32_t time_stamp,
-                                   int64_t start_time_ms,
+                                   int32_t decode_time_ms,
                                    int64_t now_ms,
                                    int64_t render_time_ms) {
   CriticalSectionScoped cs(crit_sect_);
-  int32_t time_diff_ms = codec_timer_.StopTimer(start_time_ms, now_ms);
-  assert(time_diff_ms >= 0);
-  last_decode_ms_ = time_diff_ms;
+  codec_timer_.MaxFilter(decode_time_ms, now_ms);
+  assert(decode_time_ms >= 0);
+  last_decode_ms_ = decode_time_ms;
 
   // Update stats.
   ++num_decoded_frames_;
@@ -191,8 +195,8 @@ void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) {
   ts_extrapolator_->Update(now_ms, time_stamp);
 }
 
-int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms)
-    const {
+int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp,
+                                int64_t now_ms) const {
   CriticalSectionScoped cs(crit_sect_);
   const int64_t render_time_ms = RenderTimeMsInternal(frame_timestamp, now_ms);
   return render_time_ms;
@@ -201,7 +205,7 @@ int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms)
 int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
                                         int64_t now_ms) const {
   int64_t estimated_complete_time_ms =
-    ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
+      ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
   if (estimated_complete_time_ms == -1) {
     estimated_complete_time_ms = now_ms;
   }
@@ -212,19 +216,19 @@ int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
 }
 
 // Must be called from inside a critical section.
-int32_t VCMTiming::MaxDecodeTimeMs(FrameType frame_type /*= kVideoFrameDelta*/)
-    const {
+int32_t VCMTiming::MaxDecodeTimeMs(
+    FrameType frame_type /*= kVideoFrameDelta*/) const {
   const int32_t decode_time_ms = codec_timer_.RequiredDecodeTimeMs(frame_type);
   assert(decode_time_ms >= 0);
   return decode_time_ms;
 }
 
-uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms, int64_t now_ms)
-    const {
+uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms,
+                                   int64_t now_ms) const {
   CriticalSectionScoped cs(crit_sect_);
 
-  const int64_t max_wait_time_ms = render_time_ms - now_ms -
-      MaxDecodeTimeMs() - render_delay_ms_;
+  const int64_t max_wait_time_ms =
+      render_time_ms - now_ms - MaxDecodeTimeMs() - render_delay_ms_;
 
   if (max_wait_time_ms < 0) {
     return 0;
@@ -232,8 +236,8 @@ uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms, int64_t now_ms)
   return static_cast<uint32_t>(max_wait_time_ms);
 }
 
-bool VCMTiming::EnoughTimeToDecode(uint32_t available_processing_time_ms)
-    const {
+bool VCMTiming::EnoughTimeToDecode(
+    uint32_t available_processing_time_ms) const {
   CriticalSectionScoped cs(crit_sect_);
   int32_t max_decode_time_ms = MaxDecodeTimeMs();
   if (max_decode_time_ms < 0) {
@@ -246,7 +250,8 @@ bool VCMTiming::EnoughTimeToDecode(uint32_t available_processing_time_ms)
     max_decode_time_ms = 1;
   }
   return static_cast<int32_t>(available_processing_time_ms) -
-      max_decode_time_ms > 0;
+             max_decode_time_ms >
+         0;
 }
 
 uint32_t VCMTiming::TargetVideoDelay() const {
@@ -256,7 +261,7 @@ uint32_t VCMTiming::TargetVideoDelay() const {
 
 uint32_t VCMTiming::TargetDelayInternal() const {
   return std::max(min_playout_delay_ms_,
-      jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_);
+                  jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_);
 }
 
 void VCMTiming::GetTimings(int* decode_ms,
diff --git a/webrtc/modules/video_coding/main/source/timing.h b/webrtc/modules/video_coding/timing.h
index d3b8fa673f..a4d0cf4543 100644
--- a/webrtc/modules/video_coding/main/source/timing.h
+++ b/webrtc/modules/video_coding/timing.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/video_coding/main/source/codec_timer.h"
+#include "webrtc/modules/video_coding/codec_timer.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
@@ -25,8 +25,7 @@ class VCMTiming {
  public:
   // The primary timing component should be passed
   // if this is the dual timing component.
-  VCMTiming(Clock* clock,
-            VCMTiming* master_timing = NULL);
+  explicit VCMTiming(Clock* clock, VCMTiming* master_timing = NULL);
   ~VCMTiming();
 
   // Resets the timing to the initial state.
@@ -58,7 +57,7 @@ class VCMTiming {
   // Stops the decoder timer, should be called when the decoder returns a frame
   // or when the decoded frame callback is called.
   int32_t StopDecodeTimer(uint32_t time_stamp,
-                          int64_t start_time_ms,
+                          int32_t decode_time_ms,
                           int64_t now_ms,
                           int64_t render_time_ms);
 
@@ -124,4 +123,4 @@ class VCMTiming {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
diff --git a/webrtc/modules/video_coding/main/source/timing_unittest.cc b/webrtc/modules/video_coding/timing_unittest.cc
index 694a600c2a..2e8df83683 100644
--- a/webrtc/modules/video_coding/main/source/timing_unittest.cc
+++ b/webrtc/modules/video_coding/timing_unittest.cc
@@ -14,10 +14,10 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -55,8 +55,9 @@ TEST(ReceiverTiming, Tests) {
   clock.AdvanceTimeMilliseconds(1000);
   timing.SetJitterDelay(jitterDelayMs);
   timing.UpdateCurrentDelay(timeStamp);
-  waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(
-      timeStamp, clock.TimeInMilliseconds()), clock.TimeInMilliseconds());
+  waitTime = timing.MaxWaitingTime(
+      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
+      clock.TimeInMilliseconds());
   // Since we gradually increase the delay we only get 100 ms every second.
   EXPECT_EQ(jitterDelayMs - 10, waitTime);
 
@@ -85,9 +86,10 @@ TEST(ReceiverTiming, Tests) {
   for (int i = 0; i < 10; i++) {
     int64_t startTimeMs = clock.TimeInMilliseconds();
     clock.AdvanceTimeMilliseconds(10);
-    timing.StopDecodeTimer(timeStamp, startTimeMs,
-                           clock.TimeInMilliseconds(), timing.RenderTimeMs(
-                               timeStamp, clock.TimeInMilliseconds()));
+    timing.StopDecodeTimer(
+        timeStamp, clock.TimeInMilliseconds() - startTimeMs,
+        clock.TimeInMilliseconds(),
+        timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()));
     timeStamp += 90000 / 25;
     clock.AdvanceTimeMilliseconds(1000 / 25 - 10);
     timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
@@ -105,7 +107,7 @@ TEST(ReceiverTiming, Tests) {
   uint32_t minTotalDelayMs = 200;
   timing.set_min_playout_delay(minTotalDelayMs);
   clock.AdvanceTimeMilliseconds(5000);
-  timeStamp += 5*90000;
+  timeStamp += 5 * 90000;
   timing.UpdateCurrentDelay(timeStamp);
   const int kRenderDelayMs = 10;
   timing.set_render_delay(kRenderDelayMs);
@@ -121,7 +123,7 @@ TEST(ReceiverTiming, Tests) {
   // Reset playout delay.
   timing.set_min_playout_delay(0);
   clock.AdvanceTimeMilliseconds(5000);
-  timeStamp += 5*90000;
+  timeStamp += 5 * 90000;
   timing.UpdateCurrentDelay(timeStamp);
 }
 
@@ -135,8 +137,8 @@ TEST(ReceiverTiming, WrapAround) {
     timing.IncomingTimestamp(timestamp, clock.TimeInMilliseconds());
     clock.AdvanceTimeMilliseconds(1000 / kFramerate);
     timestamp += 90000 / kFramerate;
-    int64_t render_time = timing.RenderTimeMs(0xFFFFFFFFu,
-                                              clock.TimeInMilliseconds());
+    int64_t render_time =
+        timing.RenderTimeMs(0xFFFFFFFFu, clock.TimeInMilliseconds());
     EXPECT_EQ(3 * 1000 / kFramerate, render_time);
     render_time = timing.RenderTimeMs(89u,  // One second later in 90 kHz.
                                       clock.TimeInMilliseconds());
diff --git a/webrtc/modules/video_coding/utility/frame_dropper.cc b/webrtc/modules/video_coding/utility/frame_dropper.cc
index 5262c5b88a..a0aa67be4e 100644
--- a/webrtc/modules/video_coding/utility/frame_dropper.cc
+++ b/webrtc/modules/video_coding/utility/frame_dropper.cc
@@ -8,12 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
 
 #include "webrtc/system_wrappers/include/trace.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
 const float kDefaultKeyFrameSizeAvgKBits = 0.9f;
 const float kDefaultKeyFrameRatio = 0.99f;
@@ -22,339 +21,266 @@ const float kDefaultDropRatioMax = 0.96f;
 const float kDefaultMaxTimeToDropFrames = 4.0f;  // In seconds.
 
 FrameDropper::FrameDropper()
-:
-_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
-_keyFrameRatio(kDefaultKeyFrameRatio),
-_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
-_enabled(true),
-_max_time_drops(kDefaultMaxTimeToDropFrames)
-{
-    Reset();
+    : _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
+      _keyFrameRatio(kDefaultKeyFrameRatio),
+      _dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
+      _enabled(true),
+      _max_time_drops(kDefaultMaxTimeToDropFrames) {
+  Reset();
 }
 
 FrameDropper::FrameDropper(float max_time_drops)
-:
-_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
-_keyFrameRatio(kDefaultKeyFrameRatio),
-_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
-_enabled(true),
-_max_time_drops(max_time_drops)
-{
-    Reset();
+    : _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
+      _keyFrameRatio(kDefaultKeyFrameRatio),
+      _dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
+      _enabled(true),
+      _max_time_drops(max_time_drops) {
+  Reset();
 }
 
-void
-FrameDropper::Reset()
-{
-    _keyFrameRatio.Reset(0.99f);
-    _keyFrameRatio.Apply(1.0f, 1.0f/300.0f); // 1 key frame every 10th second in 30 fps
-    _keyFrameSizeAvgKbits.Reset(0.9f);
-    _keyFrameCount = 0;
-    _accumulator = 0.0f;
-    _accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
-    _targetBitRate = 300.0f;
-    _incoming_frame_rate = 30;
-    _keyFrameSpreadFrames = 0.5f * _incoming_frame_rate;
-    _dropNext = false;
-    _dropRatio.Reset(0.9f);
-    _dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
-    _dropCount = 0;
-    _windowSize = 0.5f;
-    _wasBelowMax = true;
-    _fastMode = false; // start with normal (non-aggressive) mode
-    // Cap for the encoder buffer level/accumulator, in secs.
-    _cap_buffer_size = 3.0f;
-    // Cap on maximum amount of dropped frames between kept frames, in secs.
-    _max_time_drops = 4.0f;
+void FrameDropper::Reset() {
+  _keyFrameRatio.Reset(0.99f);
+  _keyFrameRatio.Apply(
+      1.0f, 1.0f / 300.0f);  // 1 key frame every 10th second in 30 fps
+  _keyFrameSizeAvgKbits.Reset(0.9f);
+  _keyFrameCount = 0;
+  _accumulator = 0.0f;
+  _accumulatorMax = 150.0f;  // assume 300 kb/s and 0.5 s window
+  _targetBitRate = 300.0f;
+  _incoming_frame_rate = 30;
+  _keyFrameSpreadFrames = 0.5f * _incoming_frame_rate;
+  _dropNext = false;
+  _dropRatio.Reset(0.9f);
+  _dropRatio.Apply(0.0f, 0.0f);  // Initialize to 0
+  _dropCount = 0;
+  _windowSize = 0.5f;
+  _wasBelowMax = true;
+  _fastMode = false;  // start with normal (non-aggressive) mode
+  // Cap for the encoder buffer level/accumulator, in secs.
+  _cap_buffer_size = 3.0f;
+  // Cap on maximum amount of dropped frames between kept frames, in secs.
+  _max_time_drops = 4.0f;
 }
 
-void
-FrameDropper::Enable(bool enable)
-{
-    _enabled = enable;
+void FrameDropper::Enable(bool enable) {
+  _enabled = enable;
 }
 
-void
-FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame)
-{
-    if (!_enabled)
-    {
-        return;
-    }
-    float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
-    if (!deltaFrame && !_fastMode) // fast mode does not treat key-frames any different
-    {
-        _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
-        _keyFrameRatio.Apply(1.0, 1.0);
-        if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered())
-        {
-            // Remove the average key frame size since we
-            // compensate for key frames when adding delta
-            // frames.
-            frameSizeKbits -= _keyFrameSizeAvgKbits.filtered();
-        }
-        else
-        {
-            // Shouldn't be negative, so zero is the lower bound.
-            frameSizeKbits = 0;
-        }
-        if (_keyFrameRatio.filtered() > 1e-5 &&
-            1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
-        {
-            // We are sending key frames more often than our upper bound for
-            // how much we allow the key frame compensation to be spread
-            // out in time. Therefor we must use the key frame ratio rather
-            // than keyFrameSpreadFrames.
-            _keyFrameCount =
-                static_cast<int32_t>(1 / _keyFrameRatio.filtered() + 0.5);
-        }
-        else
-        {
-            // Compensate for the key frame the following frames
-            _keyFrameCount = static_cast<int32_t>(_keyFrameSpreadFrames + 0.5);
-        }
+void FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame) {
+  if (!_enabled) {
+    return;
+  }
+  float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
+  if (!deltaFrame &&
+      !_fastMode) {  // fast mode does not treat key-frames any different
+    _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
+    _keyFrameRatio.Apply(1.0, 1.0);
+    if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered()) {
+      // Remove the average key frame size since we
+      // compensate for key frames when adding delta
+      // frames.
+      frameSizeKbits -= _keyFrameSizeAvgKbits.filtered();
+    } else {
+      // Shouldn't be negative, so zero is the lower bound.
+      frameSizeKbits = 0;
     }
-    else
-    {
-        // Decrease the keyFrameRatio
-        _keyFrameRatio.Apply(1.0, 0.0);
+    if (_keyFrameRatio.filtered() > 1e-5 &&
+        1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) {
+      // We are sending key frames more often than our upper bound for
+      // how much we allow the key frame compensation to be spread
+      // out in time. Therefor we must use the key frame ratio rather
+      // than keyFrameSpreadFrames.
+      _keyFrameCount =
+          static_cast<int32_t>(1 / _keyFrameRatio.filtered() + 0.5);
+    } else {
+      // Compensate for the key frame the following frames
+      _keyFrameCount = static_cast<int32_t>(_keyFrameSpreadFrames + 0.5);
     }
-    // Change the level of the accumulator (bucket)
-    _accumulator += frameSizeKbits;
-    CapAccumulator();
+  } else {
+    // Decrease the keyFrameRatio
+    _keyFrameRatio.Apply(1.0, 0.0);
+  }
+  // Change the level of the accumulator (bucket)
+  _accumulator += frameSizeKbits;
+  CapAccumulator();
 }
 
-void
-FrameDropper::Leak(uint32_t inputFrameRate)
-{
-    if (!_enabled)
-    {
-        return;
-    }
-    if (inputFrameRate < 1)
-    {
-        return;
-    }
-    if (_targetBitRate < 0.0f)
-    {
-        return;
-    }
-    _keyFrameSpreadFrames = 0.5f * inputFrameRate;
-    // T is the expected bits per frame (target). If all frames were the same size,
-    // we would get T bits per frame. Notice that T is also weighted to be able to
-    // force a lower frame rate if wanted.
-    float T = _targetBitRate / inputFrameRate;
-    if (_keyFrameCount > 0)
-    {
-        // Perform the key frame compensation
-        if (_keyFrameRatio.filtered() > 0 &&
-            1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
-        {
-            T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered();
-        }
-        else
-        {
-            T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames;
-        }
-        _keyFrameCount--;
-    }
-    _accumulator -= T;
-    if (_accumulator < 0.0f)
-    {
-        _accumulator = 0.0f;
+void FrameDropper::Leak(uint32_t inputFrameRate) {
+  if (!_enabled) {
+    return;
+  }
+  if (inputFrameRate < 1) {
+    return;
+  }
+  if (_targetBitRate < 0.0f) {
+    return;
+  }
+  _keyFrameSpreadFrames = 0.5f * inputFrameRate;
+  // T is the expected bits per frame (target). If all frames were the same
+  // size,
+  // we would get T bits per frame. Notice that T is also weighted to be able to
+  // force a lower frame rate if wanted.
+  float T = _targetBitRate / inputFrameRate;
+  if (_keyFrameCount > 0) {
+    // Perform the key frame compensation
+    if (_keyFrameRatio.filtered() > 0 &&
+        1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) {
+      T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered();
+    } else {
+      T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames;
     }
-    UpdateRatio();
+    _keyFrameCount--;
+  }
+  _accumulator -= T;
+  if (_accumulator < 0.0f) {
+    _accumulator = 0.0f;
+  }
+  UpdateRatio();
 }
 
-void
-FrameDropper::UpdateNack(uint32_t nackBytes)
-{
-    if (!_enabled)
-    {
-        return;
-    }
-    _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
+void FrameDropper::UpdateNack(uint32_t nackBytes) {
+  if (!_enabled) {
+    return;
+  }
+  _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
 }
 
-void
-FrameDropper::FillBucket(float inKbits, float outKbits)
-{
-    _accumulator += (inKbits - outKbits);
+void FrameDropper::FillBucket(float inKbits, float outKbits) {
+  _accumulator += (inKbits - outKbits);
 }
 
-void
-FrameDropper::UpdateRatio()
-{
-    if (_accumulator > 1.3f * _accumulatorMax)
-    {
-        // Too far above accumulator max, react faster
-        _dropRatio.UpdateBase(0.8f);
+void FrameDropper::UpdateRatio() {
+  if (_accumulator > 1.3f * _accumulatorMax) {
+    // Too far above accumulator max, react faster
+    _dropRatio.UpdateBase(0.8f);
+  } else {
+    // Go back to normal reaction
+    _dropRatio.UpdateBase(0.9f);
+  }
+  if (_accumulator > _accumulatorMax) {
+    // We are above accumulator max, and should ideally
+    // drop a frame. Increase the dropRatio and drop
+    // the frame later.
+    if (_wasBelowMax) {
+      _dropNext = true;
     }
-    else
-    {
-        // Go back to normal reaction
-        _dropRatio.UpdateBase(0.9f);
+    if (_fastMode) {
+      // always drop in aggressive mode
+      _dropNext = true;
     }
-    if (_accumulator > _accumulatorMax)
-    {
-        // We are above accumulator max, and should ideally
-        // drop a frame. Increase the dropRatio and drop
-        // the frame later.
-        if (_wasBelowMax)
-        {
-            _dropNext = true;
-        }
-        if (_fastMode)
-        {
-            // always drop in aggressive mode
-            _dropNext = true;
-        }
 
-        _dropRatio.Apply(1.0f, 1.0f);
-        _dropRatio.UpdateBase(0.9f);
-    }
-    else
-    {
-        _dropRatio.Apply(1.0f, 0.0f);
-    }
-    _wasBelowMax = _accumulator < _accumulatorMax;
+    _dropRatio.Apply(1.0f, 1.0f);
+    _dropRatio.UpdateBase(0.9f);
+  } else {
+    _dropRatio.Apply(1.0f, 0.0f);
+  }
+  _wasBelowMax = _accumulator < _accumulatorMax;
 }
 
-// This function signals when to drop frames to the caller. It makes use of the dropRatio
+// This function signals when to drop frames to the caller. It makes use of the
+// dropRatio
 // to smooth out the drops over time.
-bool
-FrameDropper::DropFrame()
-{
-    if (!_enabled)
-    {
-        return false;
+bool FrameDropper::DropFrame() {
+  if (!_enabled) {
+    return false;
+  }
+  if (_dropNext) {
+    _dropNext = false;
+    _dropCount = 0;
+  }
+
+  if (_dropRatio.filtered() >= 0.5f) {  // Drops per keep
+    // limit is the number of frames we should drop between each kept frame
+    // to keep our drop ratio. limit is positive in this case.
+    float denom = 1.0f - _dropRatio.filtered();
+    if (denom < 1e-5) {
+      denom = 1e-5f;
+    }
+    int32_t limit = static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+    // Put a bound on the max amount of dropped frames between each kept
+    // frame, in terms of frame rate and window size (secs).
+    int max_limit = static_cast<int>(_incoming_frame_rate * _max_time_drops);
+    if (limit > max_limit) {
+      limit = max_limit;
     }
-    if (_dropNext)
-    {
-        _dropNext = false;
+    if (_dropCount < 0) {
+      // Reset the _dropCount since it was negative and should be positive.
+      if (_dropRatio.filtered() > 0.4f) {
+        _dropCount = -_dropCount;
+      } else {
         _dropCount = 0;
+      }
     }
-
-    if (_dropRatio.filtered() >= 0.5f) // Drops per keep
-    {
-        // limit is the number of frames we should drop between each kept frame
-        // to keep our drop ratio. limit is positive in this case.
-        float denom = 1.0f - _dropRatio.filtered();
-        if (denom < 1e-5)
-        {
-            denom = (float)1e-5;
-        }
-        int32_t limit = static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
-        // Put a bound on the max amount of dropped frames between each kept
-        // frame, in terms of frame rate and window size (secs).
-        int max_limit = static_cast<int>(_incoming_frame_rate *
-                                         _max_time_drops);
-        if (limit > max_limit) {
-          limit = max_limit;
-        }
-        if (_dropCount < 0)
-        {
-            // Reset the _dropCount since it was negative and should be positive.
-            if (_dropRatio.filtered() > 0.4f)
-            {
-                _dropCount = -_dropCount;
-            }
-            else
-            {
-                _dropCount = 0;
-            }
-        }
-        if (_dropCount < limit)
-        {
-            // As long we are below the limit we should drop frames.
-            _dropCount++;
-            return true;
-        }
-        else
-        {
-            // Only when we reset _dropCount a frame should be kept.
-            _dropCount = 0;
-            return false;
-        }
+    if (_dropCount < limit) {
+      // As long we are below the limit we should drop frames.
+      _dropCount++;
+      return true;
+    } else {
+      // Only when we reset _dropCount a frame should be kept.
+      _dropCount = 0;
+      return false;
     }
-    else if (_dropRatio.filtered() > 0.0f &&
-        _dropRatio.filtered() < 0.5f) // Keeps per drop
-    {
-        // limit is the number of frames we should keep between each drop
-        // in order to keep the drop ratio. limit is negative in this case,
-        // and the _dropCount is also negative.
-        float denom = _dropRatio.filtered();
-        if (denom < 1e-5)
-        {
-            denom = (float)1e-5;
-        }
-        int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
-        if (_dropCount > 0)
-        {
-            // Reset the _dropCount since we have a positive
-            // _dropCount, and it should be negative.
-            if (_dropRatio.filtered() < 0.6f)
-            {
-                _dropCount = -_dropCount;
-            }
-            else
-            {
-                _dropCount = 0;
-            }
-        }
-        if (_dropCount > limit)
-        {
-            if (_dropCount == 0)
-            {
-                // Drop frames when we reset _dropCount.
-                _dropCount--;
-                return true;
-            }
-            else
-            {
-                // Keep frames as long as we haven't reached limit.
-                _dropCount--;
-                return false;
-            }
-        }
-        else
-        {
-            _dropCount = 0;
-            return false;
-        }
+  } else if (_dropRatio.filtered() > 0.0f &&
+             _dropRatio.filtered() < 0.5f) {  // Keeps per drop
+    // limit is the number of frames we should keep between each drop
+    // in order to keep the drop ratio. limit is negative in this case,
+    // and the _dropCount is also negative.
+    float denom = _dropRatio.filtered();
+    if (denom < 1e-5) {
+      denom = 1e-5f;
     }
-    _dropCount = 0;
-    return false;
+    int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+    if (_dropCount > 0) {
+      // Reset the _dropCount since we have a positive
+      // _dropCount, and it should be negative.
+      if (_dropRatio.filtered() < 0.6f) {
+        _dropCount = -_dropCount;
+      } else {
+        _dropCount = 0;
+      }
+    }
+    if (_dropCount > limit) {
+      if (_dropCount == 0) {
+        // Drop frames when we reset _dropCount.
+        _dropCount--;
+        return true;
+      } else {
+        // Keep frames as long as we haven't reached limit.
+        _dropCount--;
+        return false;
+      }
+    } else {
+      _dropCount = 0;
+      return false;
+    }
+  }
+  _dropCount = 0;
+  return false;
 
-    // A simpler version, unfiltered and quicker
-    //bool dropNext = _dropNext;
-    //_dropNext = false;
-    //return dropNext;
+  // A simpler version, unfiltered and quicker
+  // bool dropNext = _dropNext;
+  // _dropNext = false;
+  // return dropNext;
 }
 
-void
-FrameDropper::SetRates(float bitRate, float incoming_frame_rate)
-{
-    // Bit rate of -1 means infinite bandwidth.
-    _accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
-    if (_targetBitRate > 0.0f && bitRate < _targetBitRate && _accumulator > _accumulatorMax)
-    {
-        // Rescale the accumulator level if the accumulator max decreases
-        _accumulator = bitRate / _targetBitRate * _accumulator;
-    }
-    _targetBitRate = bitRate;
-    CapAccumulator();
-    _incoming_frame_rate = incoming_frame_rate;
+void FrameDropper::SetRates(float bitRate, float incoming_frame_rate) {
+  // Bit rate of -1 means infinite bandwidth.
+  _accumulatorMax = bitRate * _windowSize;  // bitRate * windowSize (in seconds)
+  if (_targetBitRate > 0.0f && bitRate < _targetBitRate &&
+      _accumulator > _accumulatorMax) {
+    // Rescale the accumulator level if the accumulator max decreases
+    _accumulator = bitRate / _targetBitRate * _accumulator;
+  }
+  _targetBitRate = bitRate;
+  CapAccumulator();
+  _incoming_frame_rate = incoming_frame_rate;
 }
 
-float
-FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const
-{
-    if (!_enabled)
-    {
-        return static_cast<float>(inputFrameRate);
-    }
-    return inputFrameRate * (1.0f - _dropRatio.filtered());
+float FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const {
+  if (!_enabled) {
+    return static_cast<float>(inputFrameRate);
+  }
+  return inputFrameRate * (1.0f - _dropRatio.filtered());
 }
 
 // Put a cap on the accumulator, i.e., don't let it grow beyond some level.
@@ -366,5 +292,4 @@ void FrameDropper::CapAccumulator() {
     _accumulator = max_accumulator;
   }
 }
-
-}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/utility/frame_dropper.h b/webrtc/modules/video_coding/utility/frame_dropper.h
new file mode 100644
index 0000000000..7ec85ea880
--- /dev/null
+++ b/webrtc/modules/video_coding/utility/frame_dropper.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
+
+#include <cstddef>
+
+#include "webrtc/base/exp_filter.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// The Frame Dropper implements a variant of the leaky bucket algorithm
+// for keeping track of when to drop frames to avoid bit rate
+// over use when the encoder can't keep its bit rate.
+class FrameDropper {
+ public:
+  FrameDropper();
+  explicit FrameDropper(float max_time_drops);
+  virtual ~FrameDropper() {}
+
+  // Resets the FrameDropper to its initial state.
+  // This means that the frameRateWeight is set to its
+  // default value as well.
+  virtual void Reset();
+
+  virtual void Enable(bool enable);
+  // Answers the question if it's time to drop a frame
+  // if we want to reach a given frame rate. Must be
+  // called for every frame.
+  //
+  // Return value     : True if we should drop the current frame
+  virtual bool DropFrame();
+  // Updates the FrameDropper with the size of the latest encoded
+  // frame. The FrameDropper calculates a new drop ratio (can be
+  // seen as the probability to drop a frame) and updates its
+  // internal statistics.
+  //
+  // Input:
+  //          - frameSizeBytes    : The size of the latest frame
+  //                                returned from the encoder.
+  //          - deltaFrame        : True if the encoder returned
+  //                                a key frame.
+  virtual void Fill(size_t frameSizeBytes, bool deltaFrame);
+
+  virtual void Leak(uint32_t inputFrameRate);
+
+  void UpdateNack(uint32_t nackBytes);
+
+  // Sets the target bit rate and the frame rate produced by
+  // the camera.
+  //
+  // Input:
+  //          - bitRate       : The target bit rate
+  virtual void SetRates(float bitRate, float incoming_frame_rate);
+
+  // Return value     : The current average frame rate produced
+  //                    if the DropFrame() function is used as
+  //                    instruction of when to drop frames.
+  virtual float ActualFrameRate(uint32_t inputFrameRate) const;
+
+ private:
+  void FillBucket(float inKbits, float outKbits);
+  void UpdateRatio();
+  void CapAccumulator();
+
+  rtc::ExpFilter _keyFrameSizeAvgKbits;
+  rtc::ExpFilter _keyFrameRatio;
+  float _keyFrameSpreadFrames;
+  int32_t _keyFrameCount;
+  float _accumulator;
+  float _accumulatorMax;
+  float _targetBitRate;
+  bool _dropNext;
+  rtc::ExpFilter _dropRatio;
+  int32_t _dropCount;
+  float _windowSize;
+  float _incoming_frame_rate;
+  bool _wasBelowMax;
+  bool _enabled;
+  bool _fastMode;
+  float _cap_buffer_size;
+  float _max_time_drops;
+};  // end of VCMFrameDropper class
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
diff --git a/webrtc/modules/video_coding/utility/include/frame_dropper.h b/webrtc/modules/video_coding/utility/include/frame_dropper.h
deleted file mode 100644
index 2b78a7264f..0000000000
--- a/webrtc/modules/video_coding/utility/include/frame_dropper.h
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
-
-#include <cstddef>
-
-#include "webrtc/base/exp_filter.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-// The Frame Dropper implements a variant of the leaky bucket algorithm
-// for keeping track of when to drop frames to avoid bit rate
-// over use when the encoder can't keep its bit rate.
-class FrameDropper
-{
-public:
-    FrameDropper();
-    explicit FrameDropper(float max_time_drops);
-    virtual ~FrameDropper() {}
-
-    // Resets the FrameDropper to its initial state.
-    // This means that the frameRateWeight is set to its
-    // default value as well.
-    virtual void Reset();
-
-    virtual void Enable(bool enable);
-    // Answers the question if it's time to drop a frame
-    // if we want to reach a given frame rate. Must be
-    // called for every frame.
-    //
-    // Return value     : True if we should drop the current frame
-    virtual bool DropFrame();
-    // Updates the FrameDropper with the size of the latest encoded
-    // frame. The FrameDropper calculates a new drop ratio (can be
-    // seen as the probability to drop a frame) and updates its
-    // internal statistics.
-    //
-    // Input:
-    //          - frameSizeBytes    : The size of the latest frame
-    //                                returned from the encoder.
-    //          - deltaFrame        : True if the encoder returned
-    //                                a key frame.
-    virtual void Fill(size_t frameSizeBytes, bool deltaFrame);
-
-    virtual void Leak(uint32_t inputFrameRate);
-
-    void UpdateNack(uint32_t nackBytes);
-
-    // Sets the target bit rate and the frame rate produced by
-    // the camera.
-    //
-    // Input:
-    //          - bitRate       : The target bit rate
-    virtual void SetRates(float bitRate, float incoming_frame_rate);
-
-    // Return value     : The current average frame rate produced
-    //                    if the DropFrame() function is used as
-    //                    instruction of when to drop frames.
-    virtual float ActualFrameRate(uint32_t inputFrameRate) const;
-
-private:
-    void FillBucket(float inKbits, float outKbits);
-    void UpdateRatio();
-    void CapAccumulator();
-
-    rtc::ExpFilter _keyFrameSizeAvgKbits;
-    rtc::ExpFilter _keyFrameRatio;
-    float _keyFrameSpreadFrames;
-    int32_t _keyFrameCount;
-    float _accumulator;
-    float _accumulatorMax;
-    float _targetBitRate;
-    bool _dropNext;
-    rtc::ExpFilter _dropRatio;
-    int32_t _dropCount;
-    float _windowSize;
-    float _incoming_frame_rate;
-    bool _wasBelowMax;
-    bool _enabled;
-    bool _fastMode;
-    float _cap_buffer_size;
-    float _max_time_drops;
-}; // end of VCMFrameDropper class
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
diff --git a/webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h b/webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h
deleted file mode 100644
index 1e31e5442a..0000000000
--- a/webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_MOCK_MOCK_FRAME_DROPPER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_MOCK_MOCK_FRAME_DROPPER_H_
-
-#include <string>
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class MockFrameDropper : public FrameDropper {
- public:
-  MOCK_METHOD0(Reset,
-      void());
-  MOCK_METHOD1(Enable,
-      void(bool enable));
-  MOCK_METHOD0(DropFrame,
-      bool());
-  MOCK_METHOD2(Fill,
-      void(size_t frameSizeBytes, bool deltaFrame));
-  MOCK_METHOD1(Leak,
-      void(uint32_t inputFrameRate));
-  MOCK_METHOD2(SetRates,
-      void(float bitRate, float incoming_frame_rate));
-  MOCK_CONST_METHOD1(ActualFrameRate,
-      float(uint32_t inputFrameRate));
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_MOCK_MOCK_FRAME_DROPPER_H_
diff --git a/webrtc/modules/video_coding/utility/include/vp8_header_parser.h b/webrtc/modules/video_coding/utility/include/vp8_header_parser.h
deleted file mode 100644
index 88796ecd0e..0000000000
--- a/webrtc/modules/video_coding/utility/include/vp8_header_parser.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_PARSE_HEADER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_PARSE_HEADER_H_
-
-namespace webrtc {
-
-namespace vp8 {
-
-enum {
-  MB_FEATURE_TREE_PROBS = 3,
-  NUM_MB_SEGMENTS = 4,
-  NUM_REF_LF_DELTAS = 4,
-  NUM_MODE_LF_DELTAS = 4,
-};
-
-typedef struct VP8BitReader VP8BitReader;
-struct VP8BitReader {
-  // Boolean decoder.
-  uint32_t value_;            // Current value.
-  uint32_t range_;            // Current range minus 1. In [127, 254] interval.
-  int bits_;                  // Number of valid bits left.
-  // Read buffer.
-  const uint8_t* buf_;        // Next byte to be read.
-  const uint8_t* buf_end_;    // End of read buffer.
-  int eof_;                   // True if input is exhausted.
-};
-
-const uint8_t kVP8Log2Range[128] = {
-  7, 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4,
-  3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
-  2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
-  2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
-  1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
-  1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
-  1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
-  1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
-  0
-};
-
-// range = ((range - 1) << kVP8Log2Range[range]) + 1
-const uint8_t kVP8NewRange[128] = {
-  127, 127, 191, 127, 159, 191, 223, 127,
-  143, 159, 175, 191, 207, 223, 239, 127,
-  135, 143, 151, 159, 167, 175, 183, 191,
-  199, 207, 215, 223, 231, 239, 247, 127,
-  131, 135, 139, 143, 147, 151, 155, 159,
-  163, 167, 171, 175, 179, 183, 187, 191,
-  195, 199, 203, 207, 211, 215, 219, 223,
-  227, 231, 235, 239, 243, 247, 251, 127,
-  129, 131, 133, 135, 137, 139, 141, 143,
-  145, 147, 149, 151, 153, 155, 157, 159,
-  161, 163, 165, 167, 169, 171, 173, 175,
-  177, 179, 181, 183, 185, 187, 189, 191,
-  193, 195, 197, 199, 201, 203, 205, 207,
-  209, 211, 213, 215, 217, 219, 221, 223,
-  225, 227, 229, 231, 233, 235, 237, 239,
-  241, 243, 245, 247, 249, 251, 253, 127
-};
-
-// Gets the QP, QP range: [0, 127].
-// Returns true on success, false otherwise.
-bool GetQp(const uint8_t* buf, size_t length, int* qp);
-
-}  // namespace vp8
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_PARSE_HEADER_H_
diff --git a/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h b/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
new file mode 100644
index 0000000000..b68a4b8d5d
--- /dev/null
+++ b/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOCK_MOCK_FRAME_DROPPER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOCK_MOCK_FRAME_DROPPER_H_
+
+#include <string>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class MockFrameDropper : public FrameDropper {
+ public:
+  MOCK_METHOD0(Reset, void());
+  MOCK_METHOD1(Enable, void(bool enable));
+  MOCK_METHOD0(DropFrame, bool());
+  MOCK_METHOD2(Fill, void(size_t frameSizeBytes, bool deltaFrame));
+  MOCK_METHOD1(Leak, void(uint32_t inputFrameRate));
+  MOCK_METHOD2(SetRates, void(float bitRate, float incoming_frame_rate));
+  MOCK_CONST_METHOD1(ActualFrameRate, float(uint32_t inputFrameRate));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOCK_MOCK_FRAME_DROPPER_H_
diff --git a/webrtc/modules/video_coding/utility/include/moving_average.h b/webrtc/modules/video_coding/utility/moving_average.h
index 49c42c4ed4..494bfd51fb 100644
--- a/webrtc/modules/video_coding/utility/include/moving_average.h
+++ b/webrtc/modules/video_coding/utility/moving_average.h
@@ -16,7 +16,7 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-template<class T>
+template <class T>
 class MovingAverage {
  public:
   MovingAverage();
@@ -30,17 +30,17 @@ class MovingAverage {
   std::list<T> samples_;
 };
 
-template<class T>
-MovingAverage<T>::MovingAverage() : sum_(static_cast<T>(0)) {
-}
+template <class T>
+MovingAverage<T>::MovingAverage()
+    : sum_(static_cast<T>(0)) {}
 
-template<class T>
+template <class T>
 void MovingAverage<T>::AddSample(T sample) {
   samples_.push_back(sample);
   sum_ += sample;
 }
 
-template<class T>
+template <class T>
 bool MovingAverage<T>::GetAverage(size_t num_samples, T* avg) {
   if (num_samples > samples_.size())
     return false;
@@ -55,17 +55,17 @@ bool MovingAverage<T>::GetAverage(size_t num_samples, T* avg) {
   return true;
 }
 
-template<class T>
+template <class T>
 void MovingAverage<T>::Reset() {
   sum_ = static_cast<T>(0);
   samples_.clear();
 }
 
-template<class T>
+template <class T>
 int MovingAverage<T>::size() {
   return samples_.size();
 }
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MOVING_AVERAGE_SCALER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOVING_AVERAGE_H_
diff --git a/webrtc/modules/video_coding/utility/qp_parser.cc b/webrtc/modules/video_coding/utility/qp_parser.cc
index 62ce31351e..0916cb0094 100644
--- a/webrtc/modules/video_coding/utility/qp_parser.cc
+++ b/webrtc/modules/video_coding/utility/qp_parser.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/utility/include/qp_parser.h"
+#include "webrtc/modules/video_coding/utility/qp_parser.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/video_coding/utility/include/vp8_header_parser.h"
+#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_coding/utility/include/qp_parser.h b/webrtc/modules/video_coding/utility/qp_parser.h
index 805b37b45c..0b644ef61c 100644
--- a/webrtc/modules/video_coding/utility/include/qp_parser.h
+++ b/webrtc/modules/video_coding/utility/qp_parser.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
 
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_coding/utility/quality_scaler.cc b/webrtc/modules/video_coding/utility/quality_scaler.cc
index ec7715230e..76bf9f5b03 100644
--- a/webrtc/modules/video_coding/utility/quality_scaler.cc
+++ b/webrtc/modules/video_coding/utility/quality_scaler.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/video_coding/utility/include/quality_scaler.h"
+#include "webrtc/modules/video_coding/utility/quality_scaler.h"
 
 namespace webrtc {
 
@@ -26,8 +26,7 @@ QualityScaler::QualityScaler()
       downscale_shift_(0),
       framerate_down_(false),
       min_width_(kDefaultMinDownscaleDimension),
-      min_height_(kDefaultMinDownscaleDimension) {
-}
+      min_height_(kDefaultMinDownscaleDimension) {}
 
 void QualityScaler::Init(int low_qp_threshold,
                          int high_qp_threshold,
@@ -91,7 +90,7 @@ void QualityScaler::OnEncodeFrame(const VideoFrame& frame) {
       AdjustScale(false);
     }
   } else if (average_qp_.GetAverage(num_samples_, &avg_qp) &&
-      avg_qp <= low_qp_threshold_) {
+             avg_qp <= low_qp_threshold_) {
     if (use_framerate_reduction_ && framerate_down_) {
       target_framerate_ = -1;
       framerate_down_ = false;
@@ -104,7 +103,7 @@ void QualityScaler::OnEncodeFrame(const VideoFrame& frame) {
   assert(downscale_shift_ >= 0);
   for (int shift = downscale_shift_;
        shift > 0 && (res_.width / 2 >= min_width_) &&
-           (res_.height / 2 >= min_height_);
+       (res_.height / 2 >= min_height_);
        --shift) {
     res_.width /= 2;
     res_.height /= 2;
@@ -124,13 +123,8 @@ const VideoFrame& QualityScaler::GetScaledFrame(const VideoFrame& frame) {
   if (res.width == frame.width())
     return frame;
 
-  scaler_.Set(frame.width(),
-              frame.height(),
-              res.width,
-              res.height,
-              kI420,
-              kI420,
-              kScaleBox);
+  scaler_.Set(frame.width(), frame.height(), res.width, res.height, kI420,
+              kI420, kScaleBox);
   if (scaler_.Scale(frame, &scaled_frame_) != 0)
     return frame;
 
diff --git a/webrtc/modules/video_coding/utility/include/quality_scaler.h b/webrtc/modules/video_coding/utility/quality_scaler.h
index 29a1496c05..a1233cca51 100644
--- a/webrtc/modules/video_coding/utility/include/quality_scaler.h
+++ b/webrtc/modules/video_coding/utility/quality_scaler.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
 
 #include "webrtc/common_video/libyuv/include/scaler.h"
-#include "webrtc/modules/video_coding/utility/include/moving_average.h"
+#include "webrtc/modules/video_coding/utility/moving_average.h"
 
 namespace webrtc {
 class QualityScaler {
diff --git a/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc b/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
index 2ce1107472..bad73a748c 100644
--- a/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
+++ b/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/utility/include/quality_scaler.h"
+#include "webrtc/modules/video_coding/utility/quality_scaler.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -33,6 +33,7 @@ class QualityScalerTest : public ::testing::Test {
     int width;
     int height;
   };
+
  protected:
   enum ScaleDirection {
     kKeepScaleAtHighQp,
@@ -43,8 +44,8 @@ class QualityScalerTest : public ::testing::Test {
   enum BadQualityMetric { kDropFrame, kReportLowQP };
 
   QualityScalerTest() {
-    input_frame_.CreateEmptyFrame(
-        kWidth, kHeight, kWidth, kHalfWidth, kHalfWidth);
+    input_frame_.CreateEmptyFrame(kWidth, kHeight, kWidth, kHalfWidth,
+                                  kHalfWidth);
     qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator, kHighQp, false);
     qs_.ReportFramerate(kFramerate);
     qs_.OnEncodeFrame(input_frame_);
@@ -97,7 +98,8 @@ class QualityScalerTest : public ::testing::Test {
                                      int num_second,
                                      int initial_framerate);
 
-  void VerifyQualityAdaptation(int initial_framerate, int seconds,
+  void VerifyQualityAdaptation(int initial_framerate,
+                               int seconds,
                                bool expect_spatial_resize,
                                bool expect_framerate_reduction);
 
@@ -183,8 +185,8 @@ TEST_F(QualityScalerTest, DoesNotDownscaleAfterHalfFramedrop) {
 
 void QualityScalerTest::ContinuouslyDownscalesByHalfDimensionsAndBackUp() {
   const int initial_min_dimension = input_frame_.width() < input_frame_.height()
-                                  ? input_frame_.width()
-                                  : input_frame_.height();
+                                        ? input_frame_.width()
+                                        : input_frame_.height();
   int min_dimension = initial_min_dimension;
   int current_shift = 0;
   // Drop all frames to force-trigger downscaling.
@@ -229,14 +231,14 @@ TEST_F(QualityScalerTest,
   const int kOddWidth = 517;
   const int kHalfOddWidth = (kOddWidth + 1) / 2;
   const int kOddHeight = 1239;
-  input_frame_.CreateEmptyFrame(
-      kOddWidth, kOddHeight, kOddWidth, kHalfOddWidth, kHalfOddWidth);
+  input_frame_.CreateEmptyFrame(kOddWidth, kOddHeight, kOddWidth, kHalfOddWidth,
+                                kHalfOddWidth);
   ContinuouslyDownscalesByHalfDimensionsAndBackUp();
 }
 
 void QualityScalerTest::DoesNotDownscaleFrameDimensions(int width, int height) {
-  input_frame_.CreateEmptyFrame(
-      width, height, width, (width + 1) / 2, (width + 1) / 2);
+  input_frame_.CreateEmptyFrame(width, height, width, (width + 1) / 2,
+                                (width + 1) / 2);
 
   for (int i = 0; i < kFramerate * kNumSeconds; ++i) {
     qs_.ReportDroppedFrame();
@@ -259,7 +261,9 @@ TEST_F(QualityScalerTest, DoesNotDownscaleFrom1Px) {
 }
 
 QualityScalerTest::Resolution QualityScalerTest::TriggerResolutionChange(
-    BadQualityMetric dropframe_lowqp, int num_second, int initial_framerate) {
+    BadQualityMetric dropframe_lowqp,
+    int num_second,
+    int initial_framerate) {
   QualityScalerTest::Resolution res;
   res.framerate = initial_framerate;
   qs_.OnEncodeFrame(input_frame_);
@@ -288,7 +292,9 @@ QualityScalerTest::Resolution QualityScalerTest::TriggerResolutionChange(
 }
 
 void QualityScalerTest::VerifyQualityAdaptation(
-    int initial_framerate, int seconds, bool expect_spatial_resize,
+    int initial_framerate,
+    int seconds,
+    bool expect_spatial_resize,
     bool expect_framerate_reduction) {
   const int kDisabledBadQpThreshold = kMaxQp + 1;
   qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator,
@@ -298,8 +304,8 @@ void QualityScalerTest::VerifyQualityAdaptation(
   int init_height = qs_.GetScaledResolution().height;
 
   // Test reducing framerate by dropping frame continuously.
-  QualityScalerTest::Resolution res = TriggerResolutionChange(
-      kDropFrame, seconds, initial_framerate);
+  QualityScalerTest::Resolution res =
+      TriggerResolutionChange(kDropFrame, seconds, initial_framerate);
 
   if (expect_framerate_reduction) {
     EXPECT_LT(res.framerate, initial_framerate);
diff --git a/webrtc/modules/video_coding/utility/video_coding_utility.gyp b/webrtc/modules/video_coding/utility/video_coding_utility.gyp
index f0764bb7bf..42cbb3d4e0 100644
--- a/webrtc/modules/video_coding/utility/video_coding_utility.gyp
+++ b/webrtc/modules/video_coding/utility/video_coding_utility.gyp
@@ -19,14 +19,14 @@
       ],
       'sources': [
         'frame_dropper.cc',
-        'include/frame_dropper.h',
-        'include/moving_average.h',
-        'include/qp_parser.h',
-        'include/quality_scaler.h',
-        'include/vp8_header_parser.h',
+        'frame_dropper.h',
+        'moving_average.h',
         'qp_parser.cc',
+        'qp_parser.h',
         'quality_scaler.cc',
+        'quality_scaler.h',
         'vp8_header_parser.cc',
+        'vp8_header_parser.h',
       ],
     },
   ], # targets
diff --git a/webrtc/modules/video_coding/utility/vp8_header_parser.cc b/webrtc/modules/video_coding/utility/vp8_header_parser.cc
index dc5a0e5d15..631385d0f2 100644
--- a/webrtc/modules/video_coding/utility/vp8_header_parser.cc
+++ b/webrtc/modules/video_coding/utility/vp8_header_parser.cc
@@ -7,12 +7,9 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include <stdint.h>
-#include <stdio.h>
+#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
 
-#include "webrtc/modules/video_coding/utility/include/vp8_header_parser.h"
-
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/base/logging.h"
 
 namespace webrtc {
 
@@ -46,12 +43,12 @@ static void VP8LoadNewBytes(VP8BitReader* const br) {
     const uint32_t in_bits = *(const uint32_t*)(br->buf_);
     br->buf_ += BITS >> 3;
 #if defined(WEBRTC_ARCH_BIG_ENDIAN)
-      bits = static_cast<uint32_t>(in_bits);
-      if (BITS != 8 * sizeof(uint32_t))
-        bits >>= (8 * sizeof(uint32_t) - BITS);
+    bits = static_cast<uint32_t>(in_bits);
+    if (BITS != 8 * sizeof(uint32_t))
+      bits >>= (8 * sizeof(uint32_t) - BITS);
 #else
-      bits = BSwap32(in_bits);
-      bits >>= 32 - BITS;
+    bits = BSwap32(in_bits);
+    bits >>= 32 - BITS;
 #endif
     br->value_ = bits | (br->value_ << BITS);
     br->bits_ += BITS;
@@ -63,12 +60,12 @@ static void VP8LoadNewBytes(VP8BitReader* const br) {
 static void VP8InitBitReader(VP8BitReader* const br,
                              const uint8_t* const start,
                              const uint8_t* const end) {
-  br->range_   = 255 - 1;
-  br->buf_     = start;
+  br->range_ = 255 - 1;
+  br->buf_ = start;
   br->buf_end_ = end;
-  br->value_   = 0;
-  br->bits_    = -8;   // To load the very first 8bits.
-  br->eof_     = 0;
+  br->value_ = 0;
+  br->bits_ = -8;  // To load the very first 8bits.
+  br->eof_ = 0;
   VP8LoadNewBytes(br);
 }
 
@@ -125,7 +122,7 @@ static void ParseSegmentHeader(VP8BitReader* br) {
       int s;
       VP8Get(br);
       for (s = 0; s < NUM_MB_SEGMENTS; ++s) {
-       VP8Get(br) ? VP8GetSignedValue(br, 7) : 0;
+        VP8Get(br) ? VP8GetSignedValue(br, 7) : 0;
       }
       for (s = 0; s < NUM_MB_SEGMENTS; ++s) {
         VP8Get(br) ? VP8GetSignedValue(br, 6) : 0;
diff --git a/webrtc/modules/video_coding/utility/vp8_header_parser.h b/webrtc/modules/video_coding/utility/vp8_header_parser.h
new file mode 100644
index 0000000000..b0c684c578
--- /dev/null
+++ b/webrtc/modules/video_coding/utility/vp8_header_parser.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
+
+#include <stdint.h>
+#include <stdio.h>
+
+namespace webrtc {
+
+namespace vp8 {
+
+enum {
+  MB_FEATURE_TREE_PROBS = 3,
+  NUM_MB_SEGMENTS = 4,
+  NUM_REF_LF_DELTAS = 4,
+  NUM_MODE_LF_DELTAS = 4,
+};
+
+typedef struct VP8BitReader VP8BitReader;
+struct VP8BitReader {
+  // Boolean decoder.
+  uint32_t value_;  // Current value.
+  uint32_t range_;  // Current range minus 1. In [127, 254] interval.
+  int bits_;        // Number of valid bits left.
+  // Read buffer.
+  const uint8_t* buf_;      // Next byte to be read.
+  const uint8_t* buf_end_;  // End of read buffer.
+  int eof_;                 // True if input is exhausted.
+};
+
+const uint8_t kVP8Log2Range[128] = {
+    7, 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, 3, 3,
+    3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+    2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0};
+
+// range = ((range - 1) << kVP8Log2Range[range]) + 1
+const uint8_t kVP8NewRange[128] = {
+    127, 127, 191, 127, 159, 191, 223, 127, 143, 159, 175, 191, 207, 223, 239,
+    127, 135, 143, 151, 159, 167, 175, 183, 191, 199, 207, 215, 223, 231, 239,
+    247, 127, 131, 135, 139, 143, 147, 151, 155, 159, 163, 167, 171, 175, 179,
+    183, 187, 191, 195, 199, 203, 207, 211, 215, 219, 223, 227, 231, 235, 239,
+    243, 247, 251, 127, 129, 131, 133, 135, 137, 139, 141, 143, 145, 147, 149,
+    151, 153, 155, 157, 159, 161, 163, 165, 167, 169, 171, 173, 175, 177, 179,
+    181, 183, 185, 187, 189, 191, 193, 195, 197, 199, 201, 203, 205, 207, 209,
+    211, 213, 215, 217, 219, 221, 223, 225, 227, 229, 231, 233, 235, 237, 239,
+    241, 243, 245, 247, 249, 251, 253, 127};
+
+// Gets the QP, QP range: [0, 127].
+// Returns true on success, false otherwise.
+bool GetQp(const uint8_t* buf, size_t length, int* qp);
+
+}  // namespace vp8
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
diff --git a/webrtc/modules/video_coding/video_coding.gypi b/webrtc/modules/video_coding/video_coding.gypi
index b292e0ae31..438d8f1c1f 100644
--- a/webrtc/modules/video_coding/video_coding.gypi
+++ b/webrtc/modules/video_coding/video_coding.gypi
@@ -22,61 +22,61 @@
       ],
       'sources': [
         # interfaces
-        'main/interface/video_coding.h',
-        'main/interface/video_coding_defines.h',
+        'include/video_coding.h',
+        'include/video_coding_defines.h',
 
         # headers
-        'main/source/codec_database.h',
-        'main/source/codec_timer.h',
-        'main/source/content_metrics_processing.h',
-        'main/source/decoding_state.h',
-        'main/source/encoded_frame.h',
-        'main/source/fec_tables_xor.h',
-        'main/source/frame_buffer.h',
-        'main/source/generic_decoder.h',
-        'main/source/generic_encoder.h',
-        'main/source/inter_frame_delay.h',
-        'main/source/internal_defines.h',
-        'main/source/jitter_buffer.h',
-        'main/source/jitter_buffer_common.h',
-        'main/source/jitter_estimator.h',
-        'main/source/media_opt_util.h',
-        'main/source/media_optimization.h',
-        'main/source/nack_fec_tables.h',
-        'main/source/packet.h',
-        'main/source/qm_select_data.h',
-        'main/source/qm_select.h',
-        'main/source/receiver.h',
-        'main/source/rtt_filter.h',
-        'main/source/session_info.h',
-        'main/source/timestamp_map.h',
-        'main/source/timing.h',
-        'main/source/video_coding_impl.h',
+        'codec_database.h',
+        'codec_timer.h',
+        'content_metrics_processing.h',
+        'decoding_state.h',
+        'encoded_frame.h',
+        'fec_tables_xor.h',
+        'frame_buffer.h',
+        'generic_decoder.h',
+        'generic_encoder.h',
+        'inter_frame_delay.h',
+        'internal_defines.h',
+        'jitter_buffer.h',
+        'jitter_buffer_common.h',
+        'jitter_estimator.h',
+        'media_opt_util.h',
+        'media_optimization.h',
+        'nack_fec_tables.h',
+        'packet.h',
+        'qm_select_data.h',
+        'qm_select.h',
+        'receiver.h',
+        'rtt_filter.h',
+        'session_info.h',
+        'timestamp_map.h',
+        'timing.h',
+        'video_coding_impl.h',
 
         # sources
-        'main/source/codec_database.cc',
-        'main/source/codec_timer.cc',
-        'main/source/content_metrics_processing.cc',
-        'main/source/decoding_state.cc',
-        'main/source/encoded_frame.cc',
-        'main/source/frame_buffer.cc',
-        'main/source/generic_decoder.cc',
-        'main/source/generic_encoder.cc',
-        'main/source/inter_frame_delay.cc',
-        'main/source/jitter_buffer.cc',
-        'main/source/jitter_estimator.cc',
-        'main/source/media_opt_util.cc',
-        'main/source/media_optimization.cc',
-        'main/source/packet.cc',
-        'main/source/qm_select.cc',
-        'main/source/receiver.cc',
-        'main/source/rtt_filter.cc',
-        'main/source/session_info.cc',
-        'main/source/timestamp_map.cc',
-        'main/source/timing.cc',
-        'main/source/video_coding_impl.cc',
-        'main/source/video_sender.cc',
-        'main/source/video_receiver.cc',
+        'codec_database.cc',
+        'codec_timer.cc',
+        'content_metrics_processing.cc',
+        'decoding_state.cc',
+        'encoded_frame.cc',
+        'frame_buffer.cc',
+        'generic_decoder.cc',
+        'generic_encoder.cc',
+        'inter_frame_delay.cc',
+        'jitter_buffer.cc',
+        'jitter_estimator.cc',
+        'media_opt_util.cc',
+        'media_optimization.cc',
+        'packet.cc',
+        'qm_select.cc',
+        'receiver.cc',
+        'rtt_filter.cc',
+        'session_info.cc',
+        'timestamp_map.cc',
+        'timing.cc',
+        'video_coding_impl.cc',
+        'video_sender.cc',
+        'video_receiver.cc',
       ], # source
       # TODO(jschuh): Bug 1348: fix size_t to int truncations.
       'msvs_disabled_warnings': [ 4267, ],
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.cc b/webrtc/modules/video_coding/video_coding_impl.cc
index b0a6754cbd..1e26a7e243 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.cc
+++ b/webrtc/modules/video_coding/video_coding_impl.cc
@@ -8,33 +8,33 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+
+#include <algorithm>
+
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/packet.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace vcm {
 
-int64_t
-VCMProcessTimer::Period() const {
-    return _periodMs;
+int64_t VCMProcessTimer::Period() const {
+  return _periodMs;
 }
 
-int64_t
-VCMProcessTimer::TimeUntilProcess() const {
-    const int64_t time_since_process = _clock->TimeInMilliseconds() - _latestMs;
-    const int64_t time_until_process = _periodMs - time_since_process;
-    return std::max<int64_t>(time_until_process, 0);
+int64_t VCMProcessTimer::TimeUntilProcess() const {
+  const int64_t time_since_process = _clock->TimeInMilliseconds() - _latestMs;
+  const int64_t time_until_process = _periodMs - time_since_process;
+  return std::max<int64_t>(time_until_process, 0);
 }
 
-void
-VCMProcessTimer::Processed() {
-    _latestMs = _clock->TimeInMilliseconds();
+void VCMProcessTimer::Processed() {
+  _latestMs = _clock->TimeInMilliseconds();
 }
 }  // namespace vcm
 
@@ -59,8 +59,8 @@ class EncodedImageCallbackWrapper : public EncodedImageCallback {
                           const RTPFragmentationHeader* fragmentation) {
     CriticalSectionScoped cs(cs_.get());
     if (callback_)
-      return callback_->Encoded(
-          encoded_image, codec_specific_info, fragmentation);
+      return callback_->Encoded(encoded_image, codec_specific_info,
+                                fragmentation);
     return 0;
   }
 
@@ -77,30 +77,26 @@ class VideoCodingModuleImpl : public VideoCodingModule {
                         VideoEncoderRateObserver* encoder_rate_observer,
                         VCMQMSettingsCallback* qm_settings_callback)
       : VideoCodingModule(),
-        sender_(new vcm::VideoSender(clock,
-                                     &post_encode_callback_,
-                                     encoder_rate_observer,
-                                     qm_settings_callback)),
-        receiver_(new vcm::VideoReceiver(clock, event_factory)),
+        sender_(clock,
+                &post_encode_callback_,
+                encoder_rate_observer,
+                qm_settings_callback),
+        receiver_(clock, event_factory),
         own_event_factory_(owns_event_factory ? event_factory : NULL) {}
 
-  virtual ~VideoCodingModuleImpl() {
-    sender_.reset();
-    receiver_.reset();
-    own_event_factory_.reset();
-  }
+  virtual ~VideoCodingModuleImpl() { own_event_factory_.reset(); }
 
   int64_t TimeUntilNextProcess() override {
-    int64_t sender_time = sender_->TimeUntilNextProcess();
-    int64_t receiver_time = receiver_->TimeUntilNextProcess();
+    int64_t sender_time = sender_.TimeUntilNextProcess();
+    int64_t receiver_time = receiver_.TimeUntilNextProcess();
     assert(sender_time >= 0);
     assert(receiver_time >= 0);
     return VCM_MIN(sender_time, receiver_time);
   }
 
   int32_t Process() override {
-    int32_t sender_return = sender_->Process();
-    int32_t receiver_return = receiver_->Process();
+    int32_t sender_return = sender_.Process();
+    int32_t receiver_return = receiver_.Process();
     if (sender_return != VCM_OK)
       return sender_return;
     return receiver_return;
@@ -109,192 +105,176 @@ class VideoCodingModuleImpl : public VideoCodingModule {
   int32_t RegisterSendCodec(const VideoCodec* sendCodec,
                             uint32_t numberOfCores,
                             uint32_t maxPayloadSize) override {
-    return sender_->RegisterSendCodec(sendCodec, numberOfCores, maxPayloadSize);
-  }
-
-  const VideoCodec& GetSendCodec() const override {
-    return sender_->GetSendCodec();
-  }
-
-  // DEPRECATED.
-  int32_t SendCodec(VideoCodec* currentSendCodec) const override {
-    return sender_->SendCodecBlocking(currentSendCodec);
-  }
-
-  // DEPRECATED.
-  VideoCodecType SendCodec() const override {
-    return sender_->SendCodecBlocking();
+    return sender_.RegisterSendCodec(sendCodec, numberOfCores, maxPayloadSize);
   }
 
   int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
                                   uint8_t payloadType,
                                   bool internalSource) override {
-    return sender_->RegisterExternalEncoder(
-        externalEncoder, payloadType, internalSource);
+    sender_.RegisterExternalEncoder(externalEncoder, payloadType,
+                                    internalSource);
+    return 0;
   }
 
   int Bitrate(unsigned int* bitrate) const override {
-    return sender_->Bitrate(bitrate);
+    return sender_.Bitrate(bitrate);
   }
 
   int FrameRate(unsigned int* framerate) const override {
-    return sender_->FrameRate(framerate);
+    return sender_.FrameRate(framerate);
   }
 
   int32_t SetChannelParameters(uint32_t target_bitrate,  // bits/s.
                                uint8_t lossRate,
                                int64_t rtt) override {
-    return sender_->SetChannelParameters(target_bitrate, lossRate, rtt);
+    return sender_.SetChannelParameters(target_bitrate, lossRate, rtt);
   }
 
   int32_t RegisterTransportCallback(
       VCMPacketizationCallback* transport) override {
-    return sender_->RegisterTransportCallback(transport);
+    return sender_.RegisterTransportCallback(transport);
   }
 
   int32_t RegisterSendStatisticsCallback(
       VCMSendStatisticsCallback* sendStats) override {
-    return sender_->RegisterSendStatisticsCallback(sendStats);
+    return sender_.RegisterSendStatisticsCallback(sendStats);
   }
 
   int32_t RegisterProtectionCallback(
       VCMProtectionCallback* protection) override {
-    return sender_->RegisterProtectionCallback(protection);
+    return sender_.RegisterProtectionCallback(protection);
   }
 
   int32_t SetVideoProtection(VCMVideoProtection videoProtection,
                              bool enable) override {
     // TODO(pbos): Remove enable from receive-side protection modes as well.
     if (enable)
-      sender_->SetVideoProtection(videoProtection);
-    return receiver_->SetVideoProtection(videoProtection, enable);
+      sender_.SetVideoProtection(videoProtection);
+    return receiver_.SetVideoProtection(videoProtection, enable);
   }
 
   int32_t AddVideoFrame(const VideoFrame& videoFrame,
                         const VideoContentMetrics* contentMetrics,
                         const CodecSpecificInfo* codecSpecificInfo) override {
-    return sender_->AddVideoFrame(
-        videoFrame, contentMetrics, codecSpecificInfo);
+    return sender_.AddVideoFrame(videoFrame, contentMetrics, codecSpecificInfo);
   }
 
   int32_t IntraFrameRequest(int stream_index) override {
-    return sender_->IntraFrameRequest(stream_index);
+    return sender_.IntraFrameRequest(stream_index);
   }
 
   int32_t EnableFrameDropper(bool enable) override {
-    return sender_->EnableFrameDropper(enable);
+    return sender_.EnableFrameDropper(enable);
   }
 
   void SuspendBelowMinBitrate() override {
-    return sender_->SuspendBelowMinBitrate();
+    return sender_.SuspendBelowMinBitrate();
   }
 
-  bool VideoSuspended() const override { return sender_->VideoSuspended(); }
+  bool VideoSuspended() const override { return sender_.VideoSuspended(); }
 
   int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
                                int32_t numberOfCores,
                                bool requireKeyFrame) override {
-    return receiver_->RegisterReceiveCodec(
-        receiveCodec, numberOfCores, requireKeyFrame);
+    return receiver_.RegisterReceiveCodec(receiveCodec, numberOfCores,
+                                          requireKeyFrame);
   }
 
-  int32_t RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                  uint8_t payloadType,
-                                  bool internalRenderTiming) override {
-    return receiver_->RegisterExternalDecoder(
-        externalDecoder, payloadType, internalRenderTiming);
+  void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                               uint8_t payloadType) override {
+    receiver_.RegisterExternalDecoder(externalDecoder, payloadType);
   }
 
   int32_t RegisterReceiveCallback(
       VCMReceiveCallback* receiveCallback) override {
-    return receiver_->RegisterReceiveCallback(receiveCallback);
+    return receiver_.RegisterReceiveCallback(receiveCallback);
   }
 
   int32_t RegisterReceiveStatisticsCallback(
       VCMReceiveStatisticsCallback* receiveStats) override {
-    return receiver_->RegisterReceiveStatisticsCallback(receiveStats);
+    return receiver_.RegisterReceiveStatisticsCallback(receiveStats);
   }
 
   int32_t RegisterDecoderTimingCallback(
       VCMDecoderTimingCallback* decoderTiming) override {
-    return receiver_->RegisterDecoderTimingCallback(decoderTiming);
+    return receiver_.RegisterDecoderTimingCallback(decoderTiming);
   }
 
   int32_t RegisterFrameTypeCallback(
       VCMFrameTypeCallback* frameTypeCallback) override {
-    return receiver_->RegisterFrameTypeCallback(frameTypeCallback);
+    return receiver_.RegisterFrameTypeCallback(frameTypeCallback);
   }
 
   int32_t RegisterPacketRequestCallback(
       VCMPacketRequestCallback* callback) override {
-    return receiver_->RegisterPacketRequestCallback(callback);
+    return receiver_.RegisterPacketRequestCallback(callback);
   }
 
   int RegisterRenderBufferSizeCallback(
       VCMRenderBufferSizeCallback* callback) override {
-    return receiver_->RegisterRenderBufferSizeCallback(callback);
+    return receiver_.RegisterRenderBufferSizeCallback(callback);
   }
 
   int32_t Decode(uint16_t maxWaitTimeMs) override {
-    return receiver_->Decode(maxWaitTimeMs);
+    return receiver_.Decode(maxWaitTimeMs);
   }
 
-  int32_t ResetDecoder() override { return receiver_->ResetDecoder(); }
+  int32_t ResetDecoder() override { return receiver_.ResetDecoder(); }
 
   int32_t ReceiveCodec(VideoCodec* currentReceiveCodec) const override {
-    return receiver_->ReceiveCodec(currentReceiveCodec);
+    return receiver_.ReceiveCodec(currentReceiveCodec);
   }
 
   VideoCodecType ReceiveCodec() const override {
-    return receiver_->ReceiveCodec();
+    return receiver_.ReceiveCodec();
   }
 
   int32_t IncomingPacket(const uint8_t* incomingPayload,
                          size_t payloadLength,
                          const WebRtcRTPHeader& rtpInfo) override {
-    return receiver_->IncomingPacket(incomingPayload, payloadLength, rtpInfo);
+    return receiver_.IncomingPacket(incomingPayload, payloadLength, rtpInfo);
   }
 
   int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) override {
-    return receiver_->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+    return receiver_.SetMinimumPlayoutDelay(minPlayoutDelayMs);
   }
 
   int32_t SetRenderDelay(uint32_t timeMS) override {
-    return receiver_->SetRenderDelay(timeMS);
+    return receiver_.SetRenderDelay(timeMS);
   }
 
-  int32_t Delay() const override { return receiver_->Delay(); }
+  int32_t Delay() const override { return receiver_.Delay(); }
 
   uint32_t DiscardedPackets() const override {
-    return receiver_->DiscardedPackets();
+    return receiver_.DiscardedPackets();
   }
 
   int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
                                 VCMDecodeErrorMode errorMode) override {
-    return receiver_->SetReceiverRobustnessMode(robustnessMode, errorMode);
+    return receiver_.SetReceiverRobustnessMode(robustnessMode, errorMode);
   }
 
   void SetNackSettings(size_t max_nack_list_size,
                        int max_packet_age_to_nack,
                        int max_incomplete_time_ms) override {
-    return receiver_->SetNackSettings(
-        max_nack_list_size, max_packet_age_to_nack, max_incomplete_time_ms);
+    return receiver_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
+                                     max_incomplete_time_ms);
   }
 
   void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) override {
-    return receiver_->SetDecodeErrorMode(decode_error_mode);
+    return receiver_.SetDecodeErrorMode(decode_error_mode);
   }
 
   int SetMinReceiverDelay(int desired_delay_ms) override {
-    return receiver_->SetMinReceiverDelay(desired_delay_ms);
+    return receiver_.SetMinReceiverDelay(desired_delay_ms);
   }
 
   int32_t SetReceiveChannelParameters(int64_t rtt) override {
-    return receiver_->SetReceiveChannelParameters(rtt);
+    return receiver_.SetReceiveChannelParameters(rtt);
   }
 
   void RegisterPreDecodeImageCallback(EncodedImageCallback* observer) override {
-    receiver_->RegisterPreDecodeImageCallback(observer);
+    receiver_.RegisterPreDecodeImageCallback(observer);
   }
 
   void RegisterPostEncodeImageCallback(
@@ -302,36 +282,18 @@ class VideoCodingModuleImpl : public VideoCodingModule {
     post_encode_callback_.Register(observer);
   }
 
-  void TriggerDecoderShutdown() override {
-    receiver_->TriggerDecoderShutdown();
-  }
+  void TriggerDecoderShutdown() override { receiver_.TriggerDecoderShutdown(); }
 
  private:
   EncodedImageCallbackWrapper post_encode_callback_;
-  // TODO(tommi): Change sender_ and receiver_ to be non pointers
-  // (construction is 1 alloc instead of 3).
-  rtc::scoped_ptr<vcm::VideoSender> sender_;
-  rtc::scoped_ptr<vcm::VideoReceiver> receiver_;
+  vcm::VideoSender sender_;
+  vcm::VideoReceiver receiver_;
   rtc::scoped_ptr<EventFactory> own_event_factory_;
 };
 }  // namespace
 
-uint8_t VideoCodingModule::NumberOfCodecs() {
-  return VCMCodecDataBase::NumberOfCodecs();
-}
-
-int32_t VideoCodingModule::Codec(uint8_t listId, VideoCodec* codec) {
-  if (codec == NULL) {
-    return VCM_PARAMETER_ERROR;
-  }
-  return VCMCodecDataBase::Codec(listId, codec) ? 0 : -1;
-}
-
-int32_t VideoCodingModule::Codec(VideoCodecType codecType, VideoCodec* codec) {
-  if (codec == NULL) {
-    return VCM_PARAMETER_ERROR;
-  }
-  return VCMCodecDataBase::Codec(codecType, codec) ? 0 : -1;
+void VideoCodingModule::Codec(VideoCodecType codecType, VideoCodec* codec) {
+  VCMCodecDataBase::Codec(codecType, codec);
 }
 
 VideoCodingModule* VideoCodingModule::Create(
@@ -342,9 +304,8 @@ VideoCodingModule* VideoCodingModule::Create(
                                    encoder_rate_observer, qm_settings_callback);
 }
 
-VideoCodingModule* VideoCodingModule::Create(
-    Clock* clock,
-    EventFactory* event_factory) {
+VideoCodingModule* VideoCodingModule::Create(Clock* clock,
+                                             EventFactory* event_factory) {
   assert(clock);
   assert(event_factory);
   return new VideoCodingModuleImpl(clock, event_factory, false, nullptr,
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.h b/webrtc/modules/video_coding/video_coding_impl.h
index 57f38dad13..f105fa9c18 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.h
+++ b/webrtc/modules/video_coding/video_coding_impl.h
@@ -11,21 +11,21 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
 #define WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
 
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 
 #include <vector>
 
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/video_coding/main/source/codec_database.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/generic_decoder.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
-#include "webrtc/modules/video_coding/main/source/receiver.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-#include "webrtc/modules/video_coding/utility/include/qp_parser.h"
+#include "webrtc/modules/video_coding/codec_database.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/generic_decoder.h"
+#include "webrtc/modules/video_coding/generic_encoder.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
+#include "webrtc/modules/video_coding/receiver.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/modules/video_coding/utility/qp_parser.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
@@ -67,24 +67,10 @@ class VideoSender {
   int32_t RegisterSendCodec(const VideoCodec* sendCodec,
                             uint32_t numberOfCores,
                             uint32_t maxPayloadSize);
-  // Non-blocking access to the currently active send codec configuration.
-  // Must be called from the same thread as the VideoSender instance was
-  // created on.
-  const VideoCodec& GetSendCodec() const;
 
-  // Get a copy of the currently configured send codec.
-  // This method acquires a lock to copy the current configuration out,
-  // so it can block and the returned information is not guaranteed to be
-  // accurate upon return.  Consider using GetSendCodec() instead and make
-  // decisions on that thread with regards to the current codec.
-  int32_t SendCodecBlocking(VideoCodec* currentSendCodec) const;
-
-  // Same as SendCodecBlocking.  Try to use GetSendCodec() instead.
-  VideoCodecType SendCodecBlocking() const;
-
-  int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
-                                  uint8_t payloadType,
-                                  bool internalSource);
+  void RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                               uint8_t payloadType,
+                               bool internalSource);
 
   int Bitrate(unsigned int* bitrate) const;
   int FrameRate(unsigned int* framerate) const;
@@ -150,9 +136,8 @@ class VideoReceiver {
                                int32_t numberOfCores,
                                bool requireKeyFrame);
 
-  int32_t RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                  uint8_t payloadType,
-                                  bool internalRenderTiming);
+  void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                               uint8_t payloadType);
   int32_t RegisterReceiveCallback(VCMReceiveCallback* receiveCallback);
   int32_t RegisterReceiveStatisticsCallback(
       VCMReceiveStatisticsCallback* receiveStats);
diff --git a/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc b/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
index ac6e16bd80..dd6565d505 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
+++ b/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
@@ -10,10 +10,10 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -42,16 +42,12 @@ class VCMRobustnessTest : public ::testing::Test {
     vcm_->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack, 0);
     ASSERT_EQ(0, vcm_->RegisterFrameTypeCallback(&frame_type_callback_));
     ASSERT_EQ(0, vcm_->RegisterPacketRequestCallback(&request_callback_));
-    ASSERT_EQ(VCM_OK, vcm_->Codec(kVideoCodecVP8, &video_codec_));
+    VideoCodingModule::Codec(kVideoCodecVP8, &video_codec_);
     ASSERT_EQ(VCM_OK, vcm_->RegisterReceiveCodec(&video_codec_, 1));
-    ASSERT_EQ(VCM_OK, vcm_->RegisterExternalDecoder(&decoder_,
-                                                    video_codec_.plType,
-                                                    true));
+    vcm_->RegisterExternalDecoder(&decoder_, video_codec_.plType);
   }
 
-  virtual void TearDown() {
-    VideoCodingModule::Destroy(vcm_);
-  }
+  virtual void TearDown() { VideoCodingModule::Destroy(vcm_); }
 
   void InsertPacket(uint32_t timestamp,
                     uint16_t seq_no,
@@ -89,19 +85,17 @@ TEST_F(VCMRobustnessTest, TestHardNack) {
       .With(Args<0, 1>(ElementsAre(6, 7)))
       .Times(1);
   for (int ts = 0; ts <= 6000; ts += 3000) {
-    EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
-                                       Field(&EncodedImage::_length,
-                                             kPayloadLen * 3),
-                                       Field(&EncodedImage::_completeFrame,
-                                             true)),
-                                 false, _, _, _))
+    EXPECT_CALL(decoder_,
+                Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
+                             Field(&EncodedImage::_length, kPayloadLen * 3),
+                             Field(&EncodedImage::_completeFrame, true)),
+                       false, _, _, _))
         .Times(1)
         .InSequence(s);
   }
 
   ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
-      VideoCodingModule::kHardNack,
-      kNoErrors));
+                        VideoCodingModule::kHardNack, kNoErrors));
 
   InsertPacket(0, 0, true, false, kVideoFrameKey);
   InsertPacket(0, 1, false, false, kVideoFrameKey);
@@ -138,14 +132,11 @@ TEST_F(VCMRobustnessTest, TestHardNack) {
 }
 
 TEST_F(VCMRobustnessTest, TestHardNackNoneDecoded) {
-  EXPECT_CALL(request_callback_, ResendPackets(_, _))
-      .Times(0);
-  EXPECT_CALL(frame_type_callback_, RequestKeyFrame())
-        .Times(1);
+  EXPECT_CALL(request_callback_, ResendPackets(_, _)).Times(0);
+  EXPECT_CALL(frame_type_callback_, RequestKeyFrame()).Times(1);
 
   ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
-      VideoCodingModule::kHardNack,
-      kNoErrors));
+                        VideoCodingModule::kHardNack, kNoErrors));
 
   InsertPacket(3000, 3, true, false, kVideoFrameDelta);
   InsertPacket(3000, 4, false, false, kVideoFrameDelta);
@@ -168,46 +159,43 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
       .With(Args<0, 1>(ElementsAre(4)))
       .Times(0);
 
-  EXPECT_CALL(decoder_, Copy())
-      .Times(0);
-  EXPECT_CALL(decoderCopy_, Copy())
-      .Times(0);
+  EXPECT_CALL(decoder_, Copy()).Times(0);
+  EXPECT_CALL(decoderCopy_, Copy()).Times(0);
 
   // Decode operations
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
-                                     Field(&EncodedImage::_completeFrame,
-                                           true)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
-                                     Field(&EncodedImage::_completeFrame,
-                                           false)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
-                                     Field(&EncodedImage::_completeFrame,
-                                           true)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
-                                     Field(&EncodedImage::_completeFrame,
-                                           true)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
-
-  ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
-      VideoCodingModule::kNone,
-      kWithErrors));
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
+                           Field(&EncodedImage::_completeFrame, true)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
+                           Field(&EncodedImage::_completeFrame, false)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
+                           Field(&EncodedImage::_completeFrame, true)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
+                           Field(&EncodedImage::_completeFrame, true)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
+
+  ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(VideoCodingModule::kNone,
+                                                    kWithErrors));
 
   InsertPacket(0, 0, true, false, kVideoFrameKey);
   InsertPacket(0, 1, false, false, kVideoFrameKey);
   InsertPacket(0, 2, false, true, kVideoFrameKey);
-  EXPECT_EQ(VCM_OK, vcm_->Decode(0));  // Decode timestamp 0.
-  EXPECT_EQ(VCM_OK, vcm_->Process());  // Expect no NACK list.
+  EXPECT_EQ(VCM_OK, vcm_->Decode(33));  // Decode timestamp 0.
+  EXPECT_EQ(VCM_OK, vcm_->Process());   // Expect no NACK list.
 
   clock_->AdvanceTimeMilliseconds(33);
   InsertPacket(3000, 3, true, false, kVideoFrameDelta);
@@ -224,8 +212,8 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
   EXPECT_EQ(VCM_OK, vcm_->Process());  // Expect no NACK list.
 
   clock_->AdvanceTimeMilliseconds(10);
-  EXPECT_EQ(VCM_OK, vcm_->Decode(0));  // Decode timestamp 6000 complete.
-  EXPECT_EQ(VCM_OK, vcm_->Process());  // Expect no NACK list.
+  EXPECT_EQ(VCM_OK, vcm_->Decode(23));  // Decode timestamp 6000 complete.
+  EXPECT_EQ(VCM_OK, vcm_->Process());   // Expect no NACK list.
 
   clock_->AdvanceTimeMilliseconds(23);
   InsertPacket(3000, 4, false, false, kVideoFrameDelta);
@@ -233,6 +221,6 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
   InsertPacket(9000, 9, true, false, kVideoFrameDelta);
   InsertPacket(9000, 10, false, false, kVideoFrameDelta);
   InsertPacket(9000, 11, false, true, kVideoFrameDelta);
-  EXPECT_EQ(VCM_OK, vcm_->Decode(0));  // Decode timestamp 9000 complete.
+  EXPECT_EQ(VCM_OK, vcm_->Decode(33));  // Decode timestamp 9000 complete.
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/video_coding/video_coding_test.gypi b/webrtc/modules/video_coding/video_coding_test.gypi
index 5d720ebb63..fc2fec6c98 100644
--- a/webrtc/modules/video_coding/video_coding_test.gypi
+++ b/webrtc/modules/video_coding/video_coding_test.gypi
@@ -19,16 +19,16 @@
       ],
       'sources': [
         # headers
-        'main/test/receiver_tests.h',
-        'main/test/rtp_player.h',
-        'main/test/vcm_payload_sink_factory.h',
+        'test/receiver_tests.h',
+        'test/rtp_player.h',
+        'test/vcm_payload_sink_factory.h',
 
         # sources
-        'main/test/rtp_player.cc',
-        'main/test/test_util.cc',
-        'main/test/tester_main.cc',
-        'main/test/vcm_payload_sink_factory.cc',
-        'main/test/video_rtp_play.cc',
+        'test/rtp_player.cc',
+        'test/test_util.cc',
+        'test/tester_main.cc',
+        'test/vcm_payload_sink_factory.cc',
+        'test/video_rtp_play.cc',
       ], # sources
     },
   ],
diff --git a/webrtc/modules/video_coding/main/source/video_receiver.cc b/webrtc/modules/video_coding/video_receiver.cc
index 77c069cf2d..02c0da8f48 100644
--- a/webrtc/modules/video_coding/main/source/video_receiver.cc
+++ b/webrtc/modules/video_coding/video_receiver.cc
@@ -9,16 +9,16 @@
  */
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 // #define DEBUG_DECODER_BIT_STREAM
 
@@ -31,7 +31,7 @@ VideoReceiver::VideoReceiver(Clock* clock, EventFactory* event_factory)
       _receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()),
       _timing(clock_),
       _receiver(&_timing, clock_, event_factory),
-      _decodedFrameCallback(_timing, clock_),
+      _decodedFrameCallback(&_timing, clock_),
       _frameTypeCallback(NULL),
       _receiveStatsCallback(NULL),
       _decoderTimingCallback(NULL),
@@ -84,20 +84,12 @@ int32_t VideoReceiver::Process() {
       int jitter_buffer_ms;
       int min_playout_delay_ms;
       int render_delay_ms;
-      _timing.GetTimings(&decode_ms,
-                         &max_decode_ms,
-                         &current_delay_ms,
-                         &target_delay_ms,
-                         &jitter_buffer_ms,
-                         &min_playout_delay_ms,
-                         &render_delay_ms);
-      _decoderTimingCallback->OnDecoderTiming(decode_ms,
-                                              max_decode_ms,
-                                              current_delay_ms,
-                                              target_delay_ms,
-                                              jitter_buffer_ms,
-                                              min_playout_delay_ms,
-                                              render_delay_ms);
+      _timing.GetTimings(&decode_ms, &max_decode_ms, &current_delay_ms,
+                         &target_delay_ms, &jitter_buffer_ms,
+                         &min_playout_delay_ms, &render_delay_ms);
+      _decoderTimingCallback->OnDecoderTiming(
+          decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
+          jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
     }
 
     // Size of render buffer.
@@ -235,21 +227,17 @@ int32_t VideoReceiver::RegisterDecoderTimingCallback(
   return VCM_OK;
 }
 
-// Register an externally defined decoder/render object.
-// Can be a decoder only or a decoder coupled with a renderer.
-int32_t VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                               uint8_t payloadType,
-                                               bool internalRenderTiming) {
+// Register an externally defined decoder object.
+void VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                            uint8_t payloadType) {
   CriticalSectionScoped cs(_receiveCritSect);
   if (externalDecoder == NULL) {
     // Make sure the VCM updates the decoder next time it decodes.
     _decoder = NULL;
-    return _codecDataBase.DeregisterExternalDecoder(payloadType) ? 0 : -1;
+    RTC_CHECK(_codecDataBase.DeregisterExternalDecoder(payloadType));
+    return;
   }
-  return _codecDataBase.RegisterExternalDecoder(
-             externalDecoder, payloadType, internalRenderTiming)
-             ? 0
-             : -1;
+  _codecDataBase.RegisterExternalDecoder(externalDecoder, payloadType);
 }
 
 // Register a frame type request callback.
@@ -282,52 +270,46 @@ void VideoReceiver::TriggerDecoderShutdown() {
 // Should be called as often as possible to get the most out of the decoder.
 int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
   int64_t nextRenderTimeMs;
-  bool supports_render_scheduling;
+  bool prefer_late_decoding = false;
   {
     CriticalSectionScoped cs(_receiveCritSect);
-    supports_render_scheduling = _codecDataBase.SupportsRenderScheduling();
+    prefer_late_decoding = _codecDataBase.PrefersLateDecoding();
   }
 
   VCMEncodedFrame* frame = _receiver.FrameForDecoding(
-      maxWaitTimeMs, nextRenderTimeMs, supports_render_scheduling);
+      maxWaitTimeMs, &nextRenderTimeMs, prefer_late_decoding);
 
-  if (frame == NULL) {
+  if (!frame)
     return VCM_FRAME_NOT_READY;
-  } else {
-    CriticalSectionScoped cs(_receiveCritSect);
 
-    // If this frame was too late, we should adjust the delay accordingly
-    _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
-                               clock_->TimeInMilliseconds());
+  CriticalSectionScoped cs(_receiveCritSect);
 
-    if (pre_decode_image_callback_) {
-      EncodedImage encoded_image(frame->EncodedImage());
-      int qp = -1;
-      if (qp_parser_.GetQp(*frame, &qp)) {
-        encoded_image.qp_ = qp;
-      }
-      pre_decode_image_callback_->Encoded(
-          encoded_image, frame->CodecSpecific(), NULL);
+  // If this frame was too late, we should adjust the delay accordingly
+  _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
+                             clock_->TimeInMilliseconds());
+
+  if (pre_decode_image_callback_) {
+    EncodedImage encoded_image(frame->EncodedImage());
+    int qp = -1;
+    if (qp_parser_.GetQp(*frame, &qp)) {
+      encoded_image.qp_ = qp;
     }
+    pre_decode_image_callback_->Encoded(encoded_image, frame->CodecSpecific(),
+                                        NULL);
+  }
 
 #ifdef DEBUG_DECODER_BIT_STREAM
-    if (_bitStreamBeforeDecoder != NULL) {
-      // Write bit stream to file for debugging purposes
-      if (fwrite(
-              frame->Buffer(), 1, frame->Length(), _bitStreamBeforeDecoder) !=
-          frame->Length()) {
-        return -1;
-      }
-    }
-#endif
-    const int32_t ret = Decode(*frame);
-    _receiver.ReleaseFrame(frame);
-    frame = NULL;
-    if (ret != VCM_OK) {
-      return ret;
+  if (_bitStreamBeforeDecoder != NULL) {
+    // Write bit stream to file for debugging purposes
+    if (fwrite(frame->Buffer(), 1, frame->Length(), _bitStreamBeforeDecoder) !=
+        frame->Length()) {
+      return -1;
     }
   }
-  return VCM_OK;
+#endif
+  const int32_t ret = Decode(*frame);
+  _receiver.ReleaseFrame(frame);
+  return ret;
 }
 
 int32_t VideoReceiver::RequestSliceLossIndication(
@@ -363,21 +345,10 @@ int32_t VideoReceiver::RequestKeyFrame() {
 
 // Must be called from inside the receive side critical section.
 int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
-  TRACE_EVENT_ASYNC_STEP1("webrtc",
-                          "Video",
-                          frame.TimeStamp(),
-                          "Decode",
-                          "type",
-                          frame.FrameType());
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame.TimeStamp(), "Decode",
+                          "type", frame.FrameType());
   // Change decoder if payload type has changed
-  const bool renderTimingBefore = _codecDataBase.SupportsRenderScheduling();
-  _decoder =
-      _codecDataBase.GetDecoder(frame.PayloadType(), &_decodedFrameCallback);
-  if (renderTimingBefore != _codecDataBase.SupportsRenderScheduling()) {
-    // Make sure we reset the decode time estimate since it will
-    // be zero for codecs without render timing.
-    _timing.ResetDecodeTime();
-  }
+  _decoder = _codecDataBase.GetDecoder(frame, &_decodedFrameCallback);
   if (_decoder == NULL) {
     return VCM_NO_CODEC_REGISTERED;
   }
@@ -436,8 +407,8 @@ int32_t VideoReceiver::RegisterReceiveCodec(const VideoCodec* receiveCodec,
   if (receiveCodec == NULL) {
     return VCM_PARAMETER_ERROR;
   }
-  if (!_codecDataBase.RegisterReceiveCodec(
-          receiveCodec, numberOfCores, requireKeyFrame)) {
+  if (!_codecDataBase.RegisterReceiveCodec(receiveCodec, numberOfCores,
+                                           requireKeyFrame)) {
     return -1;
   }
   return 0;
@@ -463,9 +434,7 @@ int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload,
                                       size_t payloadLength,
                                       const WebRtcRTPHeader& rtpInfo) {
   if (rtpInfo.frameType == kVideoFrameKey) {
-    TRACE_EVENT1("webrtc",
-                 "VCM::PacketKeyFrame",
-                 "seqnum",
+    TRACE_EVENT1("webrtc", "VCM::PacketKeyFrame", "seqnum",
                  rtpInfo.header.sequenceNumber);
   }
   if (incomingPayload == NULL) {
@@ -504,7 +473,9 @@ int32_t VideoReceiver::SetRenderDelay(uint32_t timeMS) {
 }
 
 // Current video delay
-int32_t VideoReceiver::Delay() const { return _timing.TargetVideoDelay(); }
+int32_t VideoReceiver::Delay() const {
+  return _timing.TargetVideoDelay();
+}
 
 uint32_t VideoReceiver::DiscardedPackets() const {
   return _receiver.DiscardedPackets();
@@ -560,8 +531,8 @@ void VideoReceiver::SetNackSettings(size_t max_nack_list_size,
     CriticalSectionScoped process_cs(process_crit_sect_.get());
     max_nack_list_size_ = max_nack_list_size;
   }
-  _receiver.SetNackSettings(
-      max_nack_list_size, max_packet_age_to_nack, max_incomplete_time_ms);
+  _receiver.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
+                            max_incomplete_time_ms);
 }
 
 int VideoReceiver::SetMinReceiverDelay(int desired_delay_ms) {
diff --git a/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc b/webrtc/modules/video_coding/video_receiver_unittest.cc
index 75ea29a1ec..820ce9ae2d 100644
--- a/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc
+++ b/webrtc/modules/video_coding/video_receiver_unittest.cc
@@ -12,11 +12,11 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 using ::testing::_;
@@ -34,14 +34,12 @@ class TestVideoReceiver : public ::testing::Test {
 
   virtual void SetUp() {
     receiver_.reset(new VideoReceiver(&clock_, &event_factory_));
-    EXPECT_EQ(0, receiver_->RegisterExternalDecoder(&decoder_,
-                                                    kUnusedPayloadType, true));
+    receiver_->RegisterExternalDecoder(&decoder_, kUnusedPayloadType);
     const size_t kMaxNackListSize = 250;
     const int kMaxPacketAgeToNack = 450;
     receiver_->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack, 0);
 
-    memset(&settings_, 0, sizeof(settings_));
-    EXPECT_EQ(0, VideoCodingModule::Codec(kVideoCodecVP8, &settings_));
+    VideoCodingModule::Codec(kVideoCodecVP8, &settings_);
     settings_.plType = kUnusedPayloadType;  // Use the mocked encoder.
     EXPECT_EQ(0, receiver_->RegisterReceiveCodec(&settings_, 1, true));
   }
@@ -56,7 +54,7 @@ class TestVideoReceiver : public ::testing::Test {
     }
     EXPECT_EQ(0, receiver_->Process());
     EXPECT_CALL(decoder_, Decode(_, _, _, _, _)).Times(0);
-    EXPECT_EQ(VCM_FRAME_NOT_READY, receiver_->Decode(0));
+    EXPECT_EQ(VCM_FRAME_NOT_READY, receiver_->Decode(100));
   }
 
   void InsertAndVerifyDecodableFrame(const uint8_t* payload,
@@ -68,7 +66,7 @@ class TestVideoReceiver : public ::testing::Test {
     EXPECT_CALL(packet_request_callback_, ResendPackets(_, _)).Times(0);
     EXPECT_EQ(0, receiver_->Process());
     EXPECT_CALL(decoder_, Decode(_, _, _, _, _)).Times(1);
-    EXPECT_EQ(0, receiver_->Decode(0));
+    EXPECT_EQ(0, receiver_->Decode(100));
   }
 
   SimulatedClock clock_;
diff --git a/webrtc/modules/video_coding/main/source/video_sender.cc b/webrtc/modules/video_coding/video_sender.cc
index 98230b1e9e..ac901f95b9 100644
--- a/webrtc/modules/video_coding/main/source/video_sender.cc
+++ b/webrtc/modules/video_coding/video_sender.cc
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/common_types.h"
 
 #include <algorithm>  // std::max
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/modules/video_coding/utility/include/quality_scaler.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/utility/quality_scaler.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 namespace vcm {
@@ -126,57 +126,34 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
   _nextFrameTypes.resize(VCM_MAX(sendCodec->numberOfSimulcastStreams, 1),
                          kVideoFrameDelta);
 
-  _mediaOpt.SetEncodingData(sendCodec->codecType,
-                            sendCodec->maxBitrate * 1000,
-                            sendCodec->startBitrate * 1000,
-                            sendCodec->width,
-                            sendCodec->height,
-                            sendCodec->maxFramerate,
-                            numLayers,
-                            maxPayloadSize);
+  _mediaOpt.SetEncodingData(sendCodec->codecType, sendCodec->maxBitrate * 1000,
+                            sendCodec->startBitrate * 1000, sendCodec->width,
+                            sendCodec->height, sendCodec->maxFramerate,
+                            numLayers, maxPayloadSize);
   return VCM_OK;
 }
 
-const VideoCodec& VideoSender::GetSendCodec() const {
-  RTC_DCHECK(main_thread_.CalledOnValidThread());
-  return current_codec_;
-}
-
-int32_t VideoSender::SendCodecBlocking(VideoCodec* currentSendCodec) const {
-  rtc::CritScope lock(&send_crit_);
-  if (currentSendCodec == nullptr) {
-    return VCM_PARAMETER_ERROR;
-  }
-  return _codecDataBase.SendCodec(currentSendCodec) ? 0 : -1;
-}
-
-VideoCodecType VideoSender::SendCodecBlocking() const {
-  rtc::CritScope lock(&send_crit_);
-  return _codecDataBase.SendCodec();
-}
-
 // Register an external decoder object.
 // This can not be used together with external decoder callbacks.
-int32_t VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
-                                             uint8_t payloadType,
-                                             bool internalSource /*= false*/) {
+void VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                          uint8_t payloadType,
+                                          bool internalSource /*= false*/) {
   RTC_DCHECK(main_thread_.CalledOnValidThread());
 
   rtc::CritScope lock(&send_crit_);
 
   if (externalEncoder == nullptr) {
     bool wasSendCodec = false;
-    const bool ret =
-        _codecDataBase.DeregisterExternalEncoder(payloadType, &wasSendCodec);
+    RTC_CHECK(
+        _codecDataBase.DeregisterExternalEncoder(payloadType, &wasSendCodec));
     if (wasSendCodec) {
       // Make sure the VCM doesn't use the de-registered codec
       _encoder = nullptr;
     }
-    return ret ? 0 : -1;
+    return;
   }
-  _codecDataBase.RegisterExternalEncoder(
-      externalEncoder, payloadType, internalSource);
-  return 0;
+  _codecDataBase.RegisterExternalEncoder(externalEncoder, payloadType,
+                                         internalSource);
 }
 
 // Get encode bitrate
@@ -369,7 +346,6 @@ void VideoSender::SuspendBelowMinBitrate() {
 }
 
 bool VideoSender::VideoSuspended() const {
-  rtc::CritScope lock(&send_crit_);
   return _mediaOpt.IsVideoSuspended();
 }
 }  // namespace vcm
diff --git a/webrtc/modules/video_coding/main/source/video_sender_unittest.cc b/webrtc/modules/video_coding/video_sender_unittest.cc
index e9c8bd79b6..741c7b7a60 100644
--- a/webrtc/modules/video_coding/main/source/video_sender_unittest.cc
+++ b/webrtc/modules/video_coding/video_sender_unittest.cc
@@ -13,18 +13,17 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
-#include "webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using ::testing::_;
 using ::testing::AllOf;
@@ -41,9 +40,7 @@ using webrtc::test::FrameGenerator;
 namespace webrtc {
 namespace vcm {
 namespace {
-enum {
-  kMaxNumberOfTemporalLayers = 3
-};
+enum { kMaxNumberOfTemporalLayers = 3 };
 
 struct Vp8StreamInfo {
   float framerate_fps[kMaxNumberOfTemporalLayers];
@@ -87,7 +84,7 @@ class EmptyFrameGenerator : public FrameGenerator {
 
 class PacketizationCallback : public VCMPacketizationCallback {
  public:
-  PacketizationCallback(Clock* clock)
+  explicit PacketizationCallback(Clock* clock)
       : clock_(clock), start_time_ms_(clock_->TimeInMilliseconds()) {}
 
   virtual ~PacketizationCallback() {}
@@ -207,22 +204,15 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
 
   void SetUp() override {
     TestVideoSender::SetUp();
-    EXPECT_EQ(
-        0,
-        sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, false));
-    memset(&settings_, 0, sizeof(settings_));
-    EXPECT_EQ(0, VideoCodingModule::Codec(kVideoCodecVP8, &settings_));
+    sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, false);
+    VideoCodingModule::Codec(kVideoCodecVP8, &settings_);
     settings_.numberOfSimulcastStreams = kNumberOfStreams;
-    ConfigureStream(kDefaultWidth / 4,
-                    kDefaultHeight / 4,
-                    100,
+    ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, 100,
                     &settings_.simulcastStream[0]);
-    ConfigureStream(kDefaultWidth / 2,
-                    kDefaultHeight / 2,
-                    500,
+    ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, 500,
                     &settings_.simulcastStream[1]);
-    ConfigureStream(
-        kDefaultWidth, kDefaultHeight, 1200, &settings_.simulcastStream[2]);
+    ConfigureStream(kDefaultWidth, kDefaultHeight, 1200,
+                    &settings_.simulcastStream[2]);
     settings_.plType = kUnusedPayloadType;  // Use the mocked encoder.
     generator_.reset(
         new EmptyFrameGenerator(settings_.width, settings_.height));
@@ -246,12 +236,11 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
     assert(stream < kNumberOfStreams);
     std::vector<FrameType> frame_types(kNumberOfStreams, kVideoFrameDelta);
     frame_types[stream] = kVideoFrameKey;
-    EXPECT_CALL(
-        encoder_,
-        Encode(_,
-               _,
-               Pointee(ElementsAreArray(&frame_types[0], frame_types.size()))))
-        .Times(1).WillRepeatedly(Return(0));
+    EXPECT_CALL(encoder_,
+                Encode(_, _, Pointee(ElementsAreArray(&frame_types[0],
+                                                      frame_types.size()))))
+        .Times(1)
+        .WillRepeatedly(Return(0));
   }
 
   static void ConfigureStream(int width,
@@ -300,11 +289,9 @@ TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequests) {
 
 TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequestsInternalCapture) {
   // De-register current external encoder.
-  EXPECT_EQ(0,
-            sender_->RegisterExternalEncoder(NULL, kUnusedPayloadType, false));
+  sender_->RegisterExternalEncoder(nullptr, kUnusedPayloadType, false);
   // Register encoder with internal capture.
-  EXPECT_EQ(
-      0, sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, true));
+  sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, true);
   EXPECT_EQ(0, sender_->RegisterSendCodec(&settings_, 1, 1200));
   ExpectIntraRequest(0);
   EXPECT_EQ(0, sender_->IntraFrameRequest(0));
@@ -384,8 +371,7 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
     codec_.startBitrate = codec_bitrate_kbps_;
     codec_.maxBitrate = codec_bitrate_kbps_;
     encoder_.reset(VP8Encoder::Create());
-    ASSERT_EQ(0, sender_->RegisterExternalEncoder(encoder_.get(), codec_.plType,
-                                                  false));
+    sender_->RegisterExternalEncoder(encoder_.get(), codec_.plType, false);
     EXPECT_EQ(0, sender_->RegisterSendCodec(&codec_, 1, 1200));
   }
 
@@ -393,8 +379,7 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
                                       int height,
                                       int temporal_layers) {
     VideoCodec codec;
-    memset(&codec, 0, sizeof(codec));
-    EXPECT_EQ(0, VideoCodingModule::Codec(kVideoCodecVP8, &codec));
+    VideoCodingModule::Codec(kVideoCodecVP8, &codec);
     codec.width = width;
     codec.height = height;
     codec.codecSpecific.VP8.numberOfTemporalLayers = temporal_layers;
@@ -436,8 +421,12 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
   int available_bitrate_kbps_;
 };
 
-TEST_F(TestVideoSenderWithVp8,
-       DISABLED_ON_IOS(DISABLED_ON_ANDROID(FixedTemporalLayersStrategy))) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_FixedTemporalLayersStrategy DISABLED_FixedTemporalLayersStrategy
+#else
+#define MAYBE_FixedTemporalLayersStrategy FixedTemporalLayersStrategy
+#endif
+TEST_F(TestVideoSenderWithVp8, MAYBE_FixedTemporalLayersStrategy) {
   const int low_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][0];
   const int mid_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][1];
   const int high_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][2];
@@ -451,8 +440,13 @@ TEST_F(TestVideoSenderWithVp8,
   }
 }
 
-TEST_F(TestVideoSenderWithVp8,
-       DISABLED_ON_IOS(DISABLED_ON_ANDROID(RealTimeTemporalLayersStrategy))) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_RealTimeTemporalLayersStrategy \
+  DISABLED_RealTimeTemporalLayersStrategy
+#else
+#define MAYBE_RealTimeTemporalLayersStrategy RealTimeTemporalLayersStrategy
+#endif
+TEST_F(TestVideoSenderWithVp8, MAYBE_RealTimeTemporalLayersStrategy) {
   Config extra_options;
   extra_options.Set<TemporalLayers::Factory>(
       new RealTimeTemporalLayersFactory());
diff --git a/webrtc/modules/video_processing/BUILD.gn b/webrtc/modules/video_processing/BUILD.gn
index 00d2911eef..6d411edda1 100644
--- a/webrtc/modules/video_processing/BUILD.gn
+++ b/webrtc/modules/video_processing/BUILD.gn
@@ -6,30 +6,37 @@
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
+import("//build/config/arm.gni")
 import("../../build/webrtc.gni")
 
 build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64"
 
 source_set("video_processing") {
   sources = [
-    "main/interface/video_processing.h",
-    "main/interface/video_processing_defines.h",
-    "main/source/brighten.cc",
-    "main/source/brighten.h",
-    "main/source/brightness_detection.cc",
-    "main/source/brightness_detection.h",
-    "main/source/content_analysis.cc",
-    "main/source/content_analysis.h",
-    "main/source/deflickering.cc",
-    "main/source/deflickering.h",
-    "main/source/frame_preprocessor.cc",
-    "main/source/frame_preprocessor.h",
-    "main/source/spatial_resampler.cc",
-    "main/source/spatial_resampler.h",
-    "main/source/video_decimator.cc",
-    "main/source/video_decimator.h",
-    "main/source/video_processing_impl.cc",
-    "main/source/video_processing_impl.h",
+    "brightness_detection.cc",
+    "brightness_detection.h",
+    "content_analysis.cc",
+    "content_analysis.h",
+    "deflickering.cc",
+    "deflickering.h",
+    "frame_preprocessor.cc",
+    "frame_preprocessor.h",
+    "include/video_processing.h",
+    "include/video_processing_defines.h",
+    "spatial_resampler.cc",
+    "spatial_resampler.h",
+    "util/denoiser_filter.cc",
+    "util/denoiser_filter.h",
+    "util/denoiser_filter_c.cc",
+    "util/denoiser_filter_c.h",
+    "util/skin_detection.cc",
+    "util/skin_detection.h",
+    "video_decimator.cc",
+    "video_decimator.h",
+    "video_denoiser.cc",
+    "video_denoiser.h",
+    "video_processing_impl.cc",
+    "video_processing_impl.h",
   ]
 
   deps = [
@@ -41,6 +48,9 @@ source_set("video_processing") {
   if (build_video_processing_sse2) {
     deps += [ ":video_processing_sse2" ]
   }
+  if (rtc_build_with_neon) {
+    deps += [ ":video_processing_neon" ]
+  }
 
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
@@ -55,7 +65,9 @@ source_set("video_processing") {
 if (build_video_processing_sse2) {
   source_set("video_processing_sse2") {
     sources = [
-      "main/source/content_analysis_sse2.cc",
+      "content_analysis_sse2.cc",
+      "util/denoiser_filter_sse2.cc",
+      "util/denoiser_filter_sse2.h",
     ]
 
     configs += [ "../..:common_config" ]
@@ -72,3 +84,18 @@ if (build_video_processing_sse2) {
     }
   }
 }
+
+if (rtc_build_with_neon) {
+  source_set("video_processing_neon") {
+    sources = [
+      "util/denoiser_filter_neon.cc",
+      "util/denoiser_filter_neon.h",
+    ]
+    if (current_cpu != "arm64") {
+      configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags = [ "-mfpu=neon" ]
+    }
+    configs += [ "../..:common_config" ]
+    public_configs = [ "../..:common_inherited_config" ]
+  }
+}
diff --git a/webrtc/modules/video_processing/OWNERS b/webrtc/modules/video_processing/OWNERS
index f452c9ed83..389d632dfd 100644
--- a/webrtc/modules/video_processing/OWNERS
+++ b/webrtc/modules/video_processing/OWNERS
@@ -1,4 +1,9 @@
 stefan@webrtc.org
 marpan@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/webrtc/modules/video_processing/main/source/brightness_detection.cc b/webrtc/modules/video_processing/brightness_detection.cc
index bae225b3b0..7455cf9759 100644
--- a/webrtc/modules/video_processing/main/source/brightness_detection.cc
+++ b/webrtc/modules/video_processing/brightness_detection.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/brightness_detection.h"
+#include "webrtc/modules/video_processing/brightness_detection.h"
 
 #include <math.h>
 
+#include "webrtc/modules/video_processing/include/video_processing.h"
+
 namespace webrtc {
 
 VPMBrightnessDetection::VPMBrightnessDetection() {
@@ -28,14 +29,14 @@ void VPMBrightnessDetection::Reset() {
 
 int32_t VPMBrightnessDetection::ProcessFrame(
     const VideoFrame& frame,
-    const VideoProcessingModule::FrameStats& stats) {
+    const VideoProcessing::FrameStats& stats) {
   if (frame.IsZeroSize()) {
     return VPM_PARAMETER_ERROR;
   }
   int width = frame.width();
   int height = frame.height();
 
-  if (!VideoProcessingModule::ValidFrameStats(stats)) {
+  if (!VideoProcessing::ValidFrameStats(stats)) {
     return VPM_PARAMETER_ERROR;
   }
 
@@ -62,11 +63,11 @@ int32_t VPMBrightnessDetection::ProcessFrame(
       // Standard deviation of Y
       const uint8_t* buffer = frame.buffer(kYPlane);
       float std_y = 0;
-      for (int h = 0; h < height; h += (1 << stats.subSamplHeight)) {
-        int row = h*width;
-        for (int w = 0; w < width; w += (1 << stats.subSamplWidth)) {
-          std_y += (buffer[w + row] - stats.mean) * (buffer[w + row] -
-              stats.mean);
+      for (int h = 0; h < height; h += (1 << stats.sub_sampling_factor)) {
+        int row = h * width;
+        for (int w = 0; w < width; w += (1 << stats.sub_sampling_factor)) {
+          std_y +=
+              (buffer[w + row] - stats.mean) * (buffer[w + row] - stats.mean);
         }
       }
       std_y = sqrt(std_y / stats.num_pixels);
@@ -81,37 +82,39 @@ int32_t VPMBrightnessDetection::ProcessFrame(
       float posPerc95 = stats.num_pixels * 0.95f;
       for (uint32_t i = 0; i < 256; i++) {
         sum += stats.hist[i];
-        if (sum < pos_perc05) perc05 = i;     // 5th perc.
-        if (sum < pos_median) median_y = i;    // 50th perc.
+        if (sum < pos_perc05)
+          perc05 = i;  // 5th perc.
+        if (sum < pos_median)
+          median_y = i;  // 50th perc.
         if (sum < posPerc95)
-          perc95 = i;     // 95th perc.
+          perc95 = i;  // 95th perc.
         else
           break;
       }
 
-        // Check if image is too dark
-        if ((std_y < 55) && (perc05 < 50))  {
-          if (median_y < 60 || stats.mean < 80 ||  perc95 < 130 ||
-              prop_low > 0.20) {
-            frame_cnt_dark_++;
-          } else {
-            frame_cnt_dark_ = 0;
-          }
+      // Check if image is too dark
+      if ((std_y < 55) && (perc05 < 50)) {
+        if (median_y < 60 || stats.mean < 80 || perc95 < 130 ||
+            prop_low > 0.20) {
+          frame_cnt_dark_++;
         } else {
           frame_cnt_dark_ = 0;
         }
+      } else {
+        frame_cnt_dark_ = 0;
+      }
 
-        // Check if image is too bright
-        if ((std_y < 52) && (perc95 > 200) && (median_y > 160)) {
-          if (median_y > 185 || stats.mean > 185 || perc05 > 140 ||
-              prop_high > 0.25) {
-            frame_cnt_bright_++;
-          } else {
-            frame_cnt_bright_ = 0;
-          }
+      // Check if image is too bright
+      if ((std_y < 52) && (perc95 > 200) && (median_y > 160)) {
+        if (median_y > 185 || stats.mean > 185 || perc05 > 140 ||
+            prop_high > 0.25) {
+          frame_cnt_bright_++;
         } else {
           frame_cnt_bright_ = 0;
         }
+      } else {
+        frame_cnt_bright_ = 0;
+      }
     } else {
       frame_cnt_dark_ = 0;
       frame_cnt_bright_ = 0;
@@ -122,11 +125,11 @@ int32_t VPMBrightnessDetection::ProcessFrame(
   }
 
   if (frame_cnt_dark_ > frame_cnt_alarm) {
-    return VideoProcessingModule::kDarkWarning;
+    return VideoProcessing::kDarkWarning;
   } else if (frame_cnt_bright_ > frame_cnt_alarm) {
-    return VideoProcessingModule::kBrightWarning;
+    return VideoProcessing::kBrightWarning;
   } else {
-    return VideoProcessingModule::kNoWarning;
+    return VideoProcessing::kNoWarning;
   }
 }
 
diff --git a/webrtc/modules/video_processing/main/source/brightness_detection.h b/webrtc/modules/video_processing/brightness_detection.h
index 48532b4a20..78a7ac5e0b 100644
--- a/webrtc/modules/video_processing/main/source/brightness_detection.h
+++ b/webrtc/modules/video_processing/brightness_detection.h
@@ -8,12 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-/*
- * brightness_detection.h
- */
-#ifndef MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTNESS_DETECTION_H
-#define MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTNESS_DETECTION_H
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
+
+#include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -25,7 +23,7 @@ class VPMBrightnessDetection {
 
   void Reset();
   int32_t ProcessFrame(const VideoFrame& frame,
-                       const VideoProcessingModule::FrameStats& stats);
+                       const VideoProcessing::FrameStats& stats);
 
  private:
   uint32_t frame_cnt_bright_;
@@ -34,4 +32,4 @@ class VPMBrightnessDetection {
 
 }  // namespace webrtc
 
-#endif // MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTNESS_DETECTION_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
diff --git a/webrtc/modules/video_processing/main/source/content_analysis.cc b/webrtc/modules/video_processing/content_analysis.cc
index d29db27408..54c04da466 100644
--- a/webrtc/modules/video_processing/main/source/content_analysis.cc
+++ b/webrtc/modules/video_processing/content_analysis.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
 
 #include <math.h>
 #include <stdlib.h>
@@ -72,7 +72,7 @@ VideoContentMetrics* VPMContentAnalysis::ComputeContentMetrics(
   // Saving current frame as previous one: Y only.
   memcpy(prev_frame_, orig_frame_, width_ * height_);
 
-  first_frame_ =  false;
+  first_frame_ = false;
   ca_Init_ = true;
 
   return ContentMetrics();
@@ -85,7 +85,7 @@ int32_t VPMContentAnalysis::Release() {
   }
 
   if (prev_frame_ != NULL) {
-    delete [] prev_frame_;
+    delete[] prev_frame_;
     prev_frame_ = NULL;
   }
 
@@ -106,11 +106,11 @@ int32_t VPMContentAnalysis::Initialize(int width, int height) {
   skip_num_ = 1;
 
   // use skipNum = 2 for 4CIF, WHD
-  if ( (height_ >=  576) && (width_ >= 704) ) {
+  if ((height_ >= 576) && (width_ >= 704)) {
     skip_num_ = 2;
   }
   // use skipNum = 4 for FULLL_HD images
-  if ( (height_ >=  1080) && (width_ >= 1920) ) {
+  if ((height_ >= 1080) && (width_ >= 1920)) {
     skip_num_ = 4;
   }
 
@@ -119,7 +119,7 @@ int32_t VPMContentAnalysis::Initialize(int width, int height) {
   }
 
   if (prev_frame_ != NULL) {
-    delete [] prev_frame_;
+    delete[] prev_frame_;
   }
 
   // Spatial Metrics don't work on a border of 8. Minimum processing
@@ -135,12 +135,12 @@ int32_t VPMContentAnalysis::Initialize(int width, int height) {
   }
 
   prev_frame_ = new uint8_t[width_ * height_];  // Y only.
-  if (prev_frame_ == NULL) return VPM_MEMORY;
+  if (prev_frame_ == NULL)
+    return VPM_MEMORY;
 
   return VPM_OK;
 }
 
-
 // Compute motion metrics: magnitude over non-zero motion vectors,
 //  and size of zero cluster
 int32_t VPMContentAnalysis::ComputeMotionMetrics() {
@@ -163,36 +163,41 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_C() {
   uint64_t pixelSqSum = 0;
 
   uint32_t num_pixels = 0;  // Counter for # of pixels.
-  const int width_end = ((width_ - 2*border_) & -16) + border_;
+  const int width_end = ((width_ - 2 * border_) & -16) + border_;
 
   for (int i = border_; i < sizei - border_; i += skip_num_) {
     for (int j = border_; j < width_end; j++) {
       num_pixels += 1;
-      int ssn =  i * sizej + j;
+      int ssn = i * sizej + j;
 
-      uint8_t currPixel  = orig_frame_[ssn];
-      uint8_t prevPixel  = prev_frame_[ssn];
+      uint8_t currPixel = orig_frame_[ssn];
+      uint8_t prevPixel = prev_frame_[ssn];
 
-      tempDiffSum += (uint32_t)abs((int16_t)(currPixel - prevPixel));
-      pixelSum += (uint32_t) currPixel;
-      pixelSqSum += (uint64_t) (currPixel * currPixel);
+      tempDiffSum +=
+          static_cast<uint32_t>(abs((int16_t)(currPixel - prevPixel)));
+      pixelSum += static_cast<uint32_t>(currPixel);
+      pixelSqSum += static_cast<uint64_t>(currPixel * currPixel);
     }
   }
 
   // Default.
   motion_magnitude_ = 0.0f;
 
-  if (tempDiffSum == 0) return VPM_OK;
+  if (tempDiffSum == 0)
+    return VPM_OK;
 
   // Normalize over all pixels.
-  float const tempDiffAvg = (float)tempDiffSum / (float)(num_pixels);
-  float const pixelSumAvg = (float)pixelSum / (float)(num_pixels);
-  float const pixelSqSumAvg = (float)pixelSqSum / (float)(num_pixels);
+  float const tempDiffAvg =
+      static_cast<float>(tempDiffSum) / static_cast<float>(num_pixels);
+  float const pixelSumAvg =
+      static_cast<float>(pixelSum) / static_cast<float>(num_pixels);
+  float const pixelSqSumAvg =
+      static_cast<float>(pixelSqSum) / static_cast<float>(num_pixels);
   float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
 
   if (contrast > 0.0) {
     contrast = sqrt(contrast);
-    motion_magnitude_ = tempDiffAvg/contrast;
+    motion_magnitude_ = tempDiffAvg / contrast;
   }
   return VPM_OK;
 }
@@ -216,39 +221,40 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_C() {
   uint32_t spatialErrHSum = 0;
 
   // make sure work section is a multiple of 16
-  const int width_end = ((sizej - 2*border_) & -16) + border_;
+  const int width_end = ((sizej - 2 * border_) & -16) + border_;
 
   for (int i = border_; i < sizei - border_; i += skip_num_) {
     for (int j = border_; j < width_end; j++) {
-      int ssn1=  i * sizej + j;
-      int ssn2 = (i + 1) * sizej + j; // bottom
-      int ssn3 = (i - 1) * sizej + j; // top
-      int ssn4 = i * sizej + j + 1;   // right
-      int ssn5 = i * sizej + j - 1;   // left
+      int ssn1 = i * sizej + j;
+      int ssn2 = (i + 1) * sizej + j;  // bottom
+      int ssn3 = (i - 1) * sizej + j;  // top
+      int ssn4 = i * sizej + j + 1;    // right
+      int ssn5 = i * sizej + j - 1;    // left
 
-      uint16_t refPixel1  = orig_frame_[ssn1] << 1;
-      uint16_t refPixel2  = orig_frame_[ssn1] << 2;
+      uint16_t refPixel1 = orig_frame_[ssn1] << 1;
+      uint16_t refPixel2 = orig_frame_[ssn1] << 2;
 
       uint8_t bottPixel = orig_frame_[ssn2];
       uint8_t topPixel = orig_frame_[ssn3];
       uint8_t rightPixel = orig_frame_[ssn4];
       uint8_t leftPixel = orig_frame_[ssn5];
 
-      spatialErrSum  += (uint32_t) abs((int16_t)(refPixel2
-          - (uint16_t)(bottPixel + topPixel + leftPixel + rightPixel)));
-      spatialErrVSum += (uint32_t) abs((int16_t)(refPixel1
-          - (uint16_t)(bottPixel + topPixel)));
-      spatialErrHSum += (uint32_t) abs((int16_t)(refPixel1
-          - (uint16_t)(leftPixel + rightPixel)));
+      spatialErrSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
+          refPixel2 - static_cast<uint16_t>(bottPixel + topPixel + leftPixel +
+                                            rightPixel))));
+      spatialErrVSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
+          refPixel1 - static_cast<uint16_t>(bottPixel + topPixel))));
+      spatialErrHSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
+          refPixel1 - static_cast<uint16_t>(leftPixel + rightPixel))));
       pixelMSA += orig_frame_[ssn1];
     }
   }
 
   // Normalize over all pixels.
-  const float spatialErr = (float)(spatialErrSum >> 2);
-  const float spatialErrH = (float)(spatialErrHSum >> 1);
-  const float spatialErrV = (float)(spatialErrVSum >> 1);
-  const float norm = (float)pixelMSA;
+  const float spatialErr = static_cast<float>(spatialErrSum >> 2);
+  const float spatialErrH = static_cast<float>(spatialErrHSum >> 1);
+  const float spatialErrV = static_cast<float>(spatialErrVSum >> 1);
+  const float norm = static_cast<float>(pixelMSA);
 
   // 2X2:
   spatial_pred_err_ = spatialErr / norm;
@@ -260,7 +266,8 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_C() {
 }
 
 VideoContentMetrics* VPMContentAnalysis::ContentMetrics() {
-  if (ca_Init_ == false) return NULL;
+  if (ca_Init_ == false)
+    return NULL;
 
   content_metrics_->spatial_pred_err = spatial_pred_err_;
   content_metrics_->spatial_pred_err_h = spatial_pred_err_h_;
diff --git a/webrtc/modules/video_processing/main/source/content_analysis.h b/webrtc/modules/video_processing/content_analysis.h
index 510c1b4a55..d3a11bd091 100644
--- a/webrtc/modules/video_processing/main/source/content_analysis.h
+++ b/webrtc/modules/video_processing/content_analysis.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
@@ -72,16 +72,16 @@ class VPMContentAnalysis {
   int border_;
 
   // Content Metrics: Stores the local average of the metrics.
-  float motion_magnitude_;   // motion class
-  float spatial_pred_err_;   // spatial class
+  float motion_magnitude_;    // motion class
+  float spatial_pred_err_;    // spatial class
   float spatial_pred_err_h_;  // spatial class
   float spatial_pred_err_v_;  // spatial class
   bool first_frame_;
   bool ca_Init_;
 
-  VideoContentMetrics*   content_metrics_;
+  VideoContentMetrics* content_metrics_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
diff --git a/webrtc/modules/video_processing/main/source/content_analysis_sse2.cc b/webrtc/modules/video_processing/content_analysis_sse2.cc
index 17b64ff280..7a60a89b45 100644
--- a/webrtc/modules/video_processing/main/source/content_analysis_sse2.cc
+++ b/webrtc/modules/video_processing/content_analysis_sse2.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
 
 #include <emmintrin.h>
 #include <math.h>
@@ -16,22 +16,22 @@
 namespace webrtc {
 
 int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
-  uint32_t num_pixels = 0;       // counter for # of pixels
-  const uint8_t* imgBufO = orig_frame_ + border_*width_ + border_;
-  const uint8_t* imgBufP = prev_frame_ + border_*width_ + border_;
+  uint32_t num_pixels = 0;  // counter for # of pixels
+  const uint8_t* imgBufO = orig_frame_ + border_ * width_ + border_;
+  const uint8_t* imgBufP = prev_frame_ + border_ * width_ + border_;
 
-  const int32_t width_end = ((width_ - 2*border_) & -16) + border_;
+  const int32_t width_end = ((width_ - 2 * border_) & -16) + border_;
 
-  __m128i sad_64   = _mm_setzero_si128();
-  __m128i sum_64   = _mm_setzero_si128();
+  __m128i sad_64 = _mm_setzero_si128();
+  __m128i sum_64 = _mm_setzero_si128();
   __m128i sqsum_64 = _mm_setzero_si128();
-  const __m128i z  = _mm_setzero_si128();
+  const __m128i z = _mm_setzero_si128();
 
-  for (uint16_t i = 0; i < (height_ - 2*border_); i += skip_num_) {
-    __m128i sqsum_32  = _mm_setzero_si128();
+  for (uint16_t i = 0; i < (height_ - 2 * border_); i += skip_num_) {
+    __m128i sqsum_32 = _mm_setzero_si128();
 
-    const uint8_t *lineO = imgBufO;
-    const uint8_t *lineP = imgBufP;
+    const uint8_t* lineO = imgBufO;
+    const uint8_t* lineP = imgBufP;
 
     // Work on 16 pixels at a time.  For HD content with a width of 1920
     // this loop will run ~67 times (depending on border).  Maximum for
@@ -49,14 +49,14 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
       lineP += 16;
 
       // Abs pixel difference between frames.
-      sad_64 = _mm_add_epi64 (sad_64, _mm_sad_epu8(o, p));
+      sad_64 = _mm_add_epi64(sad_64, _mm_sad_epu8(o, p));
 
       // sum of all pixels in frame
-      sum_64 = _mm_add_epi64 (sum_64, _mm_sad_epu8(o, z));
+      sum_64 = _mm_add_epi64(sum_64, _mm_sad_epu8(o, z));
 
       // Squared sum of all pixels in frame.
-      const __m128i olo = _mm_unpacklo_epi8(o,z);
-      const __m128i ohi = _mm_unpackhi_epi8(o,z);
+      const __m128i olo = _mm_unpacklo_epi8(o, z);
+      const __m128i ohi = _mm_unpackhi_epi8(o, z);
 
       const __m128i sqsum_32_lo = _mm_madd_epi16(olo, olo);
       const __m128i sqsum_32_hi = _mm_madd_epi16(ohi, ohi);
@@ -66,9 +66,9 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
     }
 
     // Add to 64 bit running sum as to not roll over.
-    sqsum_64 = _mm_add_epi64(sqsum_64,
-        _mm_add_epi64(_mm_unpackhi_epi32(sqsum_32,z),
-        _mm_unpacklo_epi32(sqsum_32,z)));
+    sqsum_64 =
+        _mm_add_epi64(sqsum_64, _mm_add_epi64(_mm_unpackhi_epi32(sqsum_32, z),
+                                              _mm_unpacklo_epi32(sqsum_32, z)));
 
     imgBufO += width_ * skip_num_;
     imgBufP += width_ * skip_num_;
@@ -81,13 +81,13 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
 
   // Bring sums out of vector registers and into integer register
   // domain, summing them along the way.
-  _mm_store_si128 (&sad_final_128, sad_64);
-  _mm_store_si128 (&sum_final_128, sum_64);
-  _mm_store_si128 (&sqsum_final_128, sqsum_64);
+  _mm_store_si128(&sad_final_128, sad_64);
+  _mm_store_si128(&sum_final_128, sum_64);
+  _mm_store_si128(&sqsum_final_128, sqsum_64);
 
-  uint64_t *sad_final_64 = reinterpret_cast<uint64_t*>(&sad_final_128);
-  uint64_t *sum_final_64 = reinterpret_cast<uint64_t*>(&sum_final_128);
-  uint64_t *sqsum_final_64 = reinterpret_cast<uint64_t*>(&sqsum_final_128);
+  uint64_t* sad_final_64 = reinterpret_cast<uint64_t*>(&sad_final_128);
+  uint64_t* sum_final_64 = reinterpret_cast<uint64_t*>(&sum_final_128);
+  uint64_t* sqsum_final_64 = reinterpret_cast<uint64_t*>(&sqsum_final_128);
 
   const uint32_t pixelSum = sum_final_64[0] + sum_final_64[1];
   const uint64_t pixelSqSum = sqsum_final_64[0] + sqsum_final_64[1];
@@ -96,27 +96,31 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
   // Default.
   motion_magnitude_ = 0.0f;
 
-  if (tempDiffSum == 0) return VPM_OK;
+  if (tempDiffSum == 0)
+    return VPM_OK;
 
   // Normalize over all pixels.
-  const float tempDiffAvg = (float)tempDiffSum / (float)(num_pixels);
-  const float pixelSumAvg = (float)pixelSum / (float)(num_pixels);
-  const float pixelSqSumAvg = (float)pixelSqSum / (float)(num_pixels);
+  const float tempDiffAvg =
+      static_cast<float>(tempDiffSum) / static_cast<float>(num_pixels);
+  const float pixelSumAvg =
+      static_cast<float>(pixelSum) / static_cast<float>(num_pixels);
+  const float pixelSqSumAvg =
+      static_cast<float>(pixelSqSum) / static_cast<float>(num_pixels);
   float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
 
   if (contrast > 0.0) {
     contrast = sqrt(contrast);
-    motion_magnitude_ = tempDiffAvg/contrast;
+    motion_magnitude_ = tempDiffAvg / contrast;
   }
 
   return VPM_OK;
 }
 
 int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
-  const uint8_t* imgBuf = orig_frame_ + border_*width_;
+  const uint8_t* imgBuf = orig_frame_ + border_ * width_;
   const int32_t width_end = ((width_ - 2 * border_) & -16) + border_;
 
-  __m128i se_32  = _mm_setzero_si128();
+  __m128i se_32 = _mm_setzero_si128();
   __m128i sev_32 = _mm_setzero_si128();
   __m128i seh_32 = _mm_setzero_si128();
   __m128i msa_32 = _mm_setzero_si128();
@@ -127,8 +131,8 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
   // value is maxed out at 65529 for every row, 65529*1080 = 70777800, which
   // will not roll over a 32 bit accumulator.
   // skip_num_ is also used to reduce the number of rows
-  for (int32_t i = 0; i < (height_ - 2*border_); i += skip_num_) {
-    __m128i se_16  = _mm_setzero_si128();
+  for (int32_t i = 0; i < (height_ - 2 * border_); i += skip_num_) {
+    __m128i se_16 = _mm_setzero_si128();
     __m128i sev_16 = _mm_setzero_si128();
     __m128i seh_16 = _mm_setzero_si128();
     __m128i msa_16 = _mm_setzero_si128();
@@ -143,9 +147,9 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
     // border_ could also be adjusted to concentrate on just the center of
     // the images for an HD capture in order to reduce the possiblity of
     // rollover.
-    const uint8_t *lineTop = imgBuf - width_ + border_;
-    const uint8_t *lineCen = imgBuf + border_;
-    const uint8_t *lineBot = imgBuf + width_ + border_;
+    const uint8_t* lineTop = imgBuf - width_ + border_;
+    const uint8_t* lineCen = imgBuf + border_;
+    const uint8_t* lineBot = imgBuf + width_ + border_;
 
     for (int32_t j = 0; j < width_end - border_; j += 16) {
       const __m128i t = _mm_loadu_si128((__m128i*)(lineTop));
@@ -159,20 +163,20 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
       lineBot += 16;
 
       // center pixel unpacked
-      __m128i clo = _mm_unpacklo_epi8(c,z);
-      __m128i chi = _mm_unpackhi_epi8(c,z);
+      __m128i clo = _mm_unpacklo_epi8(c, z);
+      __m128i chi = _mm_unpackhi_epi8(c, z);
 
       // left right pixels unpacked and added together
-      const __m128i lrlo = _mm_add_epi16(_mm_unpacklo_epi8(l,z),
-                                        _mm_unpacklo_epi8(r,z));
-      const __m128i lrhi = _mm_add_epi16(_mm_unpackhi_epi8(l,z),
-                                         _mm_unpackhi_epi8(r,z));
+      const __m128i lrlo =
+          _mm_add_epi16(_mm_unpacklo_epi8(l, z), _mm_unpacklo_epi8(r, z));
+      const __m128i lrhi =
+          _mm_add_epi16(_mm_unpackhi_epi8(l, z), _mm_unpackhi_epi8(r, z));
 
       // top & bottom pixels unpacked and added together
-      const __m128i tblo = _mm_add_epi16(_mm_unpacklo_epi8(t,z),
-                                         _mm_unpacklo_epi8(b,z));
-      const __m128i tbhi = _mm_add_epi16(_mm_unpackhi_epi8(t,z),
-                                         _mm_unpackhi_epi8(b,z));
+      const __m128i tblo =
+          _mm_add_epi16(_mm_unpacklo_epi8(t, z), _mm_unpacklo_epi8(b, z));
+      const __m128i tbhi =
+          _mm_add_epi16(_mm_unpackhi_epi8(t, z), _mm_unpackhi_epi8(b, z));
 
       // running sum of all pixels
       msa_16 = _mm_add_epi16(msa_16, _mm_add_epi16(chi, clo));
@@ -190,29 +194,32 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
       const __m128i sethi = _mm_subs_epi16(chi, _mm_add_epi16(lrhi, tbhi));
 
       // Add to 16 bit running sum
-      se_16  = _mm_add_epi16(se_16, _mm_max_epi16(setlo,
-          _mm_subs_epi16(z, setlo)));
-      se_16  = _mm_add_epi16(se_16, _mm_max_epi16(sethi,
-          _mm_subs_epi16(z, sethi)));
-      sev_16 = _mm_add_epi16(sev_16, _mm_max_epi16(sevtlo,
-                                             _mm_subs_epi16(z, sevtlo)));
-      sev_16 = _mm_add_epi16(sev_16, _mm_max_epi16(sevthi,
-          _mm_subs_epi16(z, sevthi)));
-      seh_16 = _mm_add_epi16(seh_16, _mm_max_epi16(sehtlo,
-            _mm_subs_epi16(z, sehtlo)));
-      seh_16 = _mm_add_epi16(seh_16, _mm_max_epi16(sehthi,
-            _mm_subs_epi16(z, sehthi)));
+      se_16 =
+          _mm_add_epi16(se_16, _mm_max_epi16(setlo, _mm_subs_epi16(z, setlo)));
+      se_16 =
+          _mm_add_epi16(se_16, _mm_max_epi16(sethi, _mm_subs_epi16(z, sethi)));
+      sev_16 = _mm_add_epi16(sev_16,
+                             _mm_max_epi16(sevtlo, _mm_subs_epi16(z, sevtlo)));
+      sev_16 = _mm_add_epi16(sev_16,
+                             _mm_max_epi16(sevthi, _mm_subs_epi16(z, sevthi)));
+      seh_16 = _mm_add_epi16(seh_16,
+                             _mm_max_epi16(sehtlo, _mm_subs_epi16(z, sehtlo)));
+      seh_16 = _mm_add_epi16(seh_16,
+                             _mm_max_epi16(sehthi, _mm_subs_epi16(z, sehthi)));
     }
 
     // Add to 32 bit running sum as to not roll over.
-    se_32  = _mm_add_epi32(se_32, _mm_add_epi32(_mm_unpackhi_epi16(se_16,z),
-        _mm_unpacklo_epi16(se_16,z)));
-    sev_32 = _mm_add_epi32(sev_32, _mm_add_epi32(_mm_unpackhi_epi16(sev_16,z),
-        _mm_unpacklo_epi16(sev_16,z)));
-    seh_32 = _mm_add_epi32(seh_32, _mm_add_epi32(_mm_unpackhi_epi16(seh_16,z),
-        _mm_unpacklo_epi16(seh_16,z)));
-    msa_32 = _mm_add_epi32(msa_32, _mm_add_epi32(_mm_unpackhi_epi16(msa_16,z),
-        _mm_unpacklo_epi16(msa_16,z)));
+    se_32 = _mm_add_epi32(se_32, _mm_add_epi32(_mm_unpackhi_epi16(se_16, z),
+                                               _mm_unpacklo_epi16(se_16, z)));
+    sev_32 =
+        _mm_add_epi32(sev_32, _mm_add_epi32(_mm_unpackhi_epi16(sev_16, z),
+                                            _mm_unpacklo_epi16(sev_16, z)));
+    seh_32 =
+        _mm_add_epi32(seh_32, _mm_add_epi32(_mm_unpackhi_epi16(seh_16, z),
+                                            _mm_unpacklo_epi16(seh_16, z)));
+    msa_32 =
+        _mm_add_epi32(msa_32, _mm_add_epi32(_mm_unpackhi_epi16(msa_16, z),
+                                            _mm_unpacklo_epi16(msa_16, z)));
 
     imgBuf += width_ * skip_num_;
   }
@@ -224,30 +231,30 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
 
   // Bring sums out of vector registers and into integer register
   // domain, summing them along the way.
-  _mm_store_si128 (&se_128, _mm_add_epi64(_mm_unpackhi_epi32(se_32,z),
-        _mm_unpacklo_epi32(se_32,z)));
-  _mm_store_si128 (&sev_128, _mm_add_epi64(_mm_unpackhi_epi32(sev_32,z),
-      _mm_unpacklo_epi32(sev_32,z)));
-  _mm_store_si128 (&seh_128, _mm_add_epi64(_mm_unpackhi_epi32(seh_32,z),
-      _mm_unpacklo_epi32(seh_32,z)));
-  _mm_store_si128 (&msa_128, _mm_add_epi64(_mm_unpackhi_epi32(msa_32,z),
-      _mm_unpacklo_epi32(msa_32,z)));
-
-  uint64_t *se_64 = reinterpret_cast<uint64_t*>(&se_128);
-  uint64_t *sev_64 = reinterpret_cast<uint64_t*>(&sev_128);
-  uint64_t *seh_64 = reinterpret_cast<uint64_t*>(&seh_128);
-  uint64_t *msa_64 = reinterpret_cast<uint64_t*>(&msa_128);
-
-  const uint32_t spatialErrSum  = se_64[0] + se_64[1];
+  _mm_store_si128(&se_128, _mm_add_epi64(_mm_unpackhi_epi32(se_32, z),
+                                         _mm_unpacklo_epi32(se_32, z)));
+  _mm_store_si128(&sev_128, _mm_add_epi64(_mm_unpackhi_epi32(sev_32, z),
+                                          _mm_unpacklo_epi32(sev_32, z)));
+  _mm_store_si128(&seh_128, _mm_add_epi64(_mm_unpackhi_epi32(seh_32, z),
+                                          _mm_unpacklo_epi32(seh_32, z)));
+  _mm_store_si128(&msa_128, _mm_add_epi64(_mm_unpackhi_epi32(msa_32, z),
+                                          _mm_unpacklo_epi32(msa_32, z)));
+
+  uint64_t* se_64 = reinterpret_cast<uint64_t*>(&se_128);
+  uint64_t* sev_64 = reinterpret_cast<uint64_t*>(&sev_128);
+  uint64_t* seh_64 = reinterpret_cast<uint64_t*>(&seh_128);
+  uint64_t* msa_64 = reinterpret_cast<uint64_t*>(&msa_128);
+
+  const uint32_t spatialErrSum = se_64[0] + se_64[1];
   const uint32_t spatialErrVSum = sev_64[0] + sev_64[1];
   const uint32_t spatialErrHSum = seh_64[0] + seh_64[1];
   const uint32_t pixelMSA = msa_64[0] + msa_64[1];
 
   // Normalize over all pixels.
-  const float spatialErr  = (float)(spatialErrSum >> 2);
-  const float spatialErrH = (float)(spatialErrHSum >> 1);
-  const float spatialErrV = (float)(spatialErrVSum >> 1);
-  const float norm = (float)pixelMSA;
+  const float spatialErr = static_cast<float>(spatialErrSum >> 2);
+  const float spatialErrH = static_cast<float>(spatialErrHSum >> 1);
+  const float spatialErrV = static_cast<float>(spatialErrVSum >> 1);
+  const float norm = static_cast<float>(pixelMSA);
 
   // 2X2:
   spatial_pred_err_ = spatialErr / norm;
@@ -258,7 +265,7 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
   // 2X1:
   spatial_pred_err_v_ = spatialErrV / norm;
 
-    return VPM_OK;
+  return VPM_OK;
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/video_processing/main/source/deflickering.cc b/webrtc/modules/video_processing/deflickering.cc
index 19bc641ac9..0e936ce9b7 100644
--- a/webrtc/modules/video_processing/main/source/deflickering.cc
+++ b/webrtc/modules/video_processing/deflickering.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/source/deflickering.h"
+#include "webrtc/modules/video_processing/deflickering.h"
 
 #include <math.h>
 #include <stdlib.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/sort.h"
 
 namespace webrtc {
@@ -40,16 +40,17 @@ enum { kLog2OfDownsamplingFactor = 3 };
 // >> fprintf('%d, ', probUW16)
 // Resolution reduced to avoid overflow when multiplying with the
 // (potentially) large number of pixels.
-const uint16_t VPMDeflickering::prob_uw16_[kNumProbs] = {102, 205, 410, 614,
-    819, 1024, 1229, 1434, 1638, 1843, 1946, 1987}; // <Q11>
+const uint16_t VPMDeflickering::prob_uw16_[kNumProbs] = {
+    102,  205,  410,  614,  819,  1024,
+    1229, 1434, 1638, 1843, 1946, 1987};  // <Q11>
 
 // To generate in Matlab:
 // >> numQuants = 14; maxOnlyLength = 5;
 // >> weightUW16 = round(2^15 *
 //    [linspace(0.5, 1.0, numQuants - maxOnlyLength)]);
 // >> fprintf('%d, %d,\n ', weightUW16);
-const uint16_t VPMDeflickering::weight_uw16_[kNumQuants - kMaxOnlyLength] =
-    {16384, 18432, 20480, 22528, 24576, 26624, 28672, 30720, 32768}; // <Q15>
+const uint16_t VPMDeflickering::weight_uw16_[kNumQuants - kMaxOnlyLength] = {
+    16384, 18432, 20480, 22528, 24576, 26624, 28672, 30720, 32768};  // <Q15>
 
 VPMDeflickering::VPMDeflickering() {
   Reset();
@@ -70,8 +71,8 @@ void VPMDeflickering::Reset() {
   quant_hist_uw8_[0][kNumQuants - 1] = 255;
   for (int32_t i = 0; i < kNumProbs; i++) {
     // Unsigned round. <Q0>
-    quant_hist_uw8_[0][i + 1] = static_cast<uint8_t>(
-        (prob_uw16_[i] * 255 + (1 << 10)) >> 11);
+    quant_hist_uw8_[0][i + 1] =
+        static_cast<uint8_t>((prob_uw16_[i] * 255 + (1 << 10)) >> 11);
   }
 
   for (int32_t i = 1; i < kFrameHistory_size; i++) {
@@ -80,9 +81,8 @@ void VPMDeflickering::Reset() {
   }
 }
 
-int32_t VPMDeflickering::ProcessFrame(
-    VideoFrame* frame,
-    VideoProcessingModule::FrameStats* stats) {
+int32_t VPMDeflickering::ProcessFrame(VideoFrame* frame,
+                                      VideoProcessing::FrameStats* stats) {
   assert(frame);
   uint32_t frame_memory;
   uint8_t quant_uw8[kNumQuants];
@@ -107,11 +107,12 @@ int32_t VPMDeflickering::ProcessFrame(
     return VPM_GENERAL_ERROR;
   }
 
-  if (!VideoProcessingModule::ValidFrameStats(*stats)) {
+  if (!VideoProcessing::ValidFrameStats(*stats)) {
     return VPM_GENERAL_ERROR;
   }
 
-  if (PreDetection(frame->timestamp(), *stats) == -1) return VPM_GENERAL_ERROR;
+  if (PreDetection(frame->timestamp(), *stats) == -1)
+    return VPM_GENERAL_ERROR;
 
   // Flicker detection
   int32_t det_flicker = DetectFlicker();
@@ -124,13 +125,13 @@ int32_t VPMDeflickering::ProcessFrame(
   // Size of luminance component.
   const uint32_t y_size = height * width;
 
-  const uint32_t y_sub_size = width * (((height - 1) >>
-      kLog2OfDownsamplingFactor) + 1);
+  const uint32_t y_sub_size =
+      width * (((height - 1) >> kLog2OfDownsamplingFactor) + 1);
   uint8_t* y_sorted = new uint8_t[y_sub_size];
   uint32_t sort_row_idx = 0;
   for (int i = 0; i < height; i += kDownsamplingFactor) {
-    memcpy(y_sorted + sort_row_idx * width,
-        frame->buffer(kYPlane) + i * width, width);
+    memcpy(y_sorted + sort_row_idx * width, frame->buffer(kYPlane) + i * width,
+           width);
     sort_row_idx++;
   }
 
@@ -153,12 +154,12 @@ int32_t VPMDeflickering::ProcessFrame(
     quant_uw8[i + 1] = y_sorted[prob_idx_uw32];
   }
 
-  delete [] y_sorted;
+  delete[] y_sorted;
   y_sorted = NULL;
 
   // Shift history for new frame.
   memmove(quant_hist_uw8_[1], quant_hist_uw8_[0],
-      (kFrameHistory_size - 1) * kNumQuants * sizeof(uint8_t));
+          (kFrameHistory_size - 1) * kNumQuants * sizeof(uint8_t));
   // Store current frame in history.
   memcpy(quant_hist_uw8_[0], quant_uw8, kNumQuants * sizeof(uint8_t));
 
@@ -190,9 +191,10 @@ int32_t VPMDeflickering::ProcessFrame(
     // target = w * maxquant_uw8 + (1 - w) * minquant_uw8
     // Weights w = |weight_uw16_| are in Q15, hence the final output has to be
     // right shifted by 8 to end up in Q7.
-    target_quant_uw16[i] = static_cast<uint16_t>((
-        weight_uw16_[i] * maxquant_uw8[i] +
-        ((1 << 15) - weight_uw16_[i]) * minquant_uw8[i]) >> 8);  // <Q7>
+    target_quant_uw16[i] = static_cast<uint16_t>(
+        (weight_uw16_[i] * maxquant_uw8[i] +
+         ((1 << 15) - weight_uw16_[i]) * minquant_uw8[i]) >>
+        8);  // <Q7>
   }
 
   for (int32_t i = kNumQuants - kMaxOnlyLength; i < kNumQuants; i++) {
@@ -203,13 +205,14 @@ int32_t VPMDeflickering::ProcessFrame(
   uint16_t mapUW16;  // <Q7>
   for (int32_t i = 1; i < kNumQuants; i++) {
     // As quant and targetQuant are limited to UWord8, it's safe to use Q7 here.
-    tmp_uw32 = static_cast<uint32_t>(target_quant_uw16[i] -
-        target_quant_uw16[i - 1]);
+    tmp_uw32 =
+        static_cast<uint32_t>(target_quant_uw16[i] - target_quant_uw16[i - 1]);
     tmp_uw16 = static_cast<uint16_t>(quant_uw8[i] - quant_uw8[i - 1]);  // <Q0>
 
     if (tmp_uw16 > 0) {
-      increment_uw16 = static_cast<uint16_t>(WebRtcSpl_DivU32U16(tmp_uw32,
-          tmp_uw16)); // <Q7>
+      increment_uw16 =
+          static_cast<uint16_t>(WebRtcSpl_DivU32U16(tmp_uw32,
+                                                    tmp_uw16));  // <Q7>
     } else {
       // The value is irrelevant; the loop below will only iterate once.
       increment_uw16 = 0;
@@ -230,7 +233,7 @@ int32_t VPMDeflickering::ProcessFrame(
   }
 
   // Frame was altered, so reset stats.
-  VideoProcessingModule::ClearFrameStats(stats);
+  VideoProcessing::ClearFrameStats(stats);
 
   return VPM_OK;
 }
@@ -247,8 +250,9 @@ int32_t VPMDeflickering::ProcessFrame(
               zero.\n
           -1: Error
 */
-int32_t VPMDeflickering::PreDetection(const uint32_t timestamp,
-  const VideoProcessingModule::FrameStats& stats) {
+int32_t VPMDeflickering::PreDetection(
+    const uint32_t timestamp,
+    const VideoProcessing::FrameStats& stats) {
   int32_t mean_val;  // Mean value of frame (Q4)
   uint32_t frame_rate = 0;
   int32_t meanBufferLength;  // Temp variable.
@@ -257,16 +261,16 @@ int32_t VPMDeflickering::PreDetection(const uint32_t timestamp,
   // Update mean value buffer.
   // This should be done even though we might end up in an unreliable detection.
   memmove(mean_buffer_ + 1, mean_buffer_,
-      (kMeanBufferLength - 1) * sizeof(int32_t));
+          (kMeanBufferLength - 1) * sizeof(int32_t));
   mean_buffer_[0] = mean_val;
 
   // Update timestamp buffer.
   // This should be done even though we might end up in an unreliable detection.
-  memmove(timestamp_buffer_ + 1, timestamp_buffer_, (kMeanBufferLength - 1) *
-      sizeof(uint32_t));
+  memmove(timestamp_buffer_ + 1, timestamp_buffer_,
+          (kMeanBufferLength - 1) * sizeof(uint32_t));
   timestamp_buffer_[0] = timestamp;
 
-/* Compute current frame rate (Q4) */
+  /* Compute current frame rate (Q4) */
   if (timestamp_buffer_[kMeanBufferLength - 1] != 0) {
     frame_rate = ((90000 << 4) * (kMeanBufferLength - 1));
     frame_rate /=
@@ -315,22 +319,22 @@ int32_t VPMDeflickering::PreDetection(const uint32_t timestamp,
           -1: Error
 */
 int32_t VPMDeflickering::DetectFlicker() {
-  uint32_t  i;
-  int32_t  freqEst;       // (Q4) Frequency estimate to base detection upon
-  int32_t  ret_val = -1;
+  uint32_t i;
+  int32_t freqEst;  // (Q4) Frequency estimate to base detection upon
+  int32_t ret_val = -1;
 
   /* Sanity check for mean_buffer_length_ */
   if (mean_buffer_length_ < 2) {
     /* Not possible to estimate frequency */
-    return(2);
+    return 2;
   }
   // Count zero crossings with a dead zone to be robust against noise. If the
   // noise std is 2 pixel this corresponds to about 95% confidence interval.
   int32_t deadzone = (kZeroCrossingDeadzone << kmean_valueScaling);  // Q4
   int32_t meanOfBuffer = 0;  // Mean value of mean value buffer.
-  int32_t numZeros     = 0;  // Number of zeros that cross the dead-zone.
-  int32_t cntState     = 0;  // State variable for zero crossing regions.
-  int32_t cntStateOld  = 0;  // Previous state for zero crossing regions.
+  int32_t numZeros = 0;      // Number of zeros that cross the dead-zone.
+  int32_t cntState = 0;      // State variable for zero crossing regions.
+  int32_t cntStateOld = 0;   // Previous state for zero crossing regions.
 
   for (i = 0; i < mean_buffer_length_; i++) {
     meanOfBuffer += mean_buffer_[i];
@@ -371,7 +375,7 @@ int32_t VPMDeflickering::DetectFlicker() {
   int32_t freqAlias = freqEst;
   if (freqEst > kMinFrequencyToDetect) {
     uint8_t aliasState = 1;
-    while(freqState == 0) {
+    while (freqState == 0) {
       /* Increase frequency */
       freqAlias += (aliasState * frame_rate_);
       freqAlias += ((freqEst << 1) * (1 - (aliasState << 1)));
diff --git a/webrtc/modules/video_processing/main/source/deflickering.h b/webrtc/modules/video_processing/deflickering.h
index 36e6845d71..3ff2723aba 100644
--- a/webrtc/modules/video_processing/main/source/deflickering.h
+++ b/webrtc/modules/video_processing/deflickering.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCEdeflickering__H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCEdeflickering__H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
 
 #include <string.h>  // NULL
 
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -24,12 +24,11 @@ class VPMDeflickering {
   ~VPMDeflickering();
 
   void Reset();
-  int32_t ProcessFrame(VideoFrame* frame,
-                       VideoProcessingModule::FrameStats* stats);
+  int32_t ProcessFrame(VideoFrame* frame, VideoProcessing::FrameStats* stats);
 
  private:
   int32_t PreDetection(uint32_t timestamp,
-                       const VideoProcessingModule::FrameStats& stats);
+                       const VideoProcessing::FrameStats& stats);
 
   int32_t DetectFlicker();
 
@@ -39,13 +38,13 @@ class VPMDeflickering {
   enum { kNumQuants = kNumProbs + 2 };
   enum { kMaxOnlyLength = 5 };
 
-  uint32_t  mean_buffer_length_;
-  uint8_t   detection_state_;    // 0: No flickering
-                                // 1: Flickering detected
-                                // 2: In flickering
-  int32_t    mean_buffer_[kMeanBufferLength];
-  uint32_t   timestamp_buffer_[kMeanBufferLength];
-  uint32_t   frame_rate_;
+  uint32_t mean_buffer_length_;
+  uint8_t detection_state_;  // 0: No flickering
+                             // 1: Flickering detected
+                             // 2: In flickering
+  int32_t mean_buffer_[kMeanBufferLength];
+  uint32_t timestamp_buffer_[kMeanBufferLength];
+  uint32_t frame_rate_;
   static const uint16_t prob_uw16_[kNumProbs];
   static const uint16_t weight_uw16_[kNumQuants - kMaxOnlyLength];
   uint8_t quant_hist_uw8_[kFrameHistory_size][kNumQuants];
@@ -53,4 +52,4 @@ class VPMDeflickering {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCEdeflickering__H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
diff --git a/webrtc/modules/video_processing/main/source/frame_preprocessor.cc b/webrtc/modules/video_processing/frame_preprocessor.cc
index a9d77c2e0c..6778a597be 100644
--- a/webrtc/modules/video_processing/main/source/frame_preprocessor.cc
+++ b/webrtc/modules/video_processing/frame_preprocessor.cc
@@ -8,12 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/source/frame_preprocessor.h"
+#include "webrtc/modules/video_processing/frame_preprocessor.h"
+
+#include "webrtc/modules/video_processing/video_denoiser.h"
 
 namespace webrtc {
 
 VPMFramePreprocessor::VPMFramePreprocessor()
-    : content_metrics_(NULL),
+    : content_metrics_(nullptr),
       resampled_frame_(),
       enable_ca_(false),
       frame_cnt_(0) {
@@ -24,21 +26,21 @@ VPMFramePreprocessor::VPMFramePreprocessor()
 
 VPMFramePreprocessor::~VPMFramePreprocessor() {
   Reset();
-  delete spatial_resampler_;
   delete ca_;
   delete vd_;
+  delete spatial_resampler_;
 }
 
-void  VPMFramePreprocessor::Reset() {
+void VPMFramePreprocessor::Reset() {
   ca_->Release();
   vd_->Reset();
-  content_metrics_ = NULL;
+  content_metrics_ = nullptr;
   spatial_resampler_->Reset();
   enable_ca_ = false;
   frame_cnt_ = 0;
 }
 
-void  VPMFramePreprocessor::EnableTemporalDecimation(bool enable) {
+void VPMFramePreprocessor::EnableTemporalDecimation(bool enable) {
   vd_->EnableTemporalDecimation(enable);
 }
 
@@ -46,20 +48,22 @@ void VPMFramePreprocessor::EnableContentAnalysis(bool enable) {
   enable_ca_ = enable;
 }
 
-void  VPMFramePreprocessor::SetInputFrameResampleMode(
+void VPMFramePreprocessor::SetInputFrameResampleMode(
     VideoFrameResampling resampling_mode) {
   spatial_resampler_->SetInputFrameResampleMode(resampling_mode);
 }
 
-int32_t VPMFramePreprocessor::SetTargetResolution(
-    uint32_t width, uint32_t height, uint32_t frame_rate) {
-  if ( (width == 0) || (height == 0) || (frame_rate == 0)) {
+int32_t VPMFramePreprocessor::SetTargetResolution(uint32_t width,
+                                                  uint32_t height,
+                                                  uint32_t frame_rate) {
+  if ((width == 0) || (height == 0) || (frame_rate == 0)) {
     return VPM_PARAMETER_ERROR;
   }
   int32_t ret_val = 0;
   ret_val = spatial_resampler_->SetTargetFrameSize(width, height);
 
-  if (ret_val < 0) return ret_val;
+  if (ret_val < 0)
+    return ret_val;
 
   vd_->SetTargetFramerate(frame_rate);
   return VPM_OK;
@@ -78,59 +82,60 @@ void VPMFramePreprocessor::UpdateIncomingframe_rate() {
   vd_->UpdateIncomingframe_rate();
 }
 
-uint32_t VPMFramePreprocessor::Decimatedframe_rate() {
-  return vd_->Decimatedframe_rate();
+uint32_t VPMFramePreprocessor::GetDecimatedFrameRate() {
+  return vd_->GetDecimatedFrameRate();
 }
 
-
-uint32_t VPMFramePreprocessor::DecimatedWidth() const {
+uint32_t VPMFramePreprocessor::GetDecimatedWidth() const {
   return spatial_resampler_->TargetWidth();
 }
 
-
-uint32_t VPMFramePreprocessor::DecimatedHeight() const {
+uint32_t VPMFramePreprocessor::GetDecimatedHeight() const {
   return spatial_resampler_->TargetHeight();
 }
 
-int32_t VPMFramePreprocessor::PreprocessFrame(const VideoFrame& frame,
-                                              VideoFrame** processed_frame) {
+void VPMFramePreprocessor::EnableDenosing(bool enable) {
+  denoiser_.reset(new VideoDenoiser(true));
+}
+
+const VideoFrame* VPMFramePreprocessor::PreprocessFrame(
+    const VideoFrame& frame) {
   if (frame.IsZeroSize()) {
-    return VPM_PARAMETER_ERROR;
+    return nullptr;
   }
 
   vd_->UpdateIncomingframe_rate();
-
   if (vd_->DropFrame()) {
-    return 1;  // drop 1 frame
+    return nullptr;
   }
 
-  // Resizing incoming frame if needed. Otherwise, remains NULL.
-  // We are not allowed to resample the input frame (must make a copy of it).
-  *processed_frame = NULL;
-  if (spatial_resampler_->ApplyResample(frame.width(), frame.height()))  {
-    int32_t ret = spatial_resampler_->ResampleFrame(frame, &resampled_frame_);
-    if (ret != VPM_OK) return ret;
-    *processed_frame = &resampled_frame_;
+  const VideoFrame* current_frame = &frame;
+  if (denoiser_) {
+    denoiser_->DenoiseFrame(*current_frame, &denoised_frame_);
+    current_frame = &denoised_frame_;
+  }
+
+  if (spatial_resampler_->ApplyResample(current_frame->width(),
+                                        current_frame->height())) {
+    if (spatial_resampler_->ResampleFrame(*current_frame, &resampled_frame_) !=
+        VPM_OK) {
+      return nullptr;
+    }
+    current_frame = &resampled_frame_;
   }
 
   // Perform content analysis on the frame to be encoded.
-  if (enable_ca_) {
+  if (enable_ca_ && frame_cnt_ % kSkipFrameCA == 0) {
     // Compute new metrics every |kSkipFramesCA| frames, starting with
     // the first frame.
-    if (frame_cnt_ % kSkipFrameCA == 0) {
-      if (*processed_frame == NULL)  {
-        content_metrics_ = ca_->ComputeContentMetrics(frame);
-      } else {
-        content_metrics_ = ca_->ComputeContentMetrics(resampled_frame_);
-      }
-    }
-    ++frame_cnt_;
+    content_metrics_ = ca_->ComputeContentMetrics(*current_frame);
   }
-  return VPM_OK;
+  ++frame_cnt_;
+  return current_frame;
 }
 
-VideoContentMetrics* VPMFramePreprocessor::ContentMetrics() const {
+VideoContentMetrics* VPMFramePreprocessor::GetContentMetrics() const {
   return content_metrics_;
 }
 
-}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/main/source/frame_preprocessor.h b/webrtc/modules/video_processing/frame_preprocessor.h
index 895e457cc6..5bdc576f37 100644
--- a/webrtc/modules/video_processing/main/source/frame_preprocessor.h
+++ b/webrtc/modules/video_processing/frame_preprocessor.h
@@ -8,20 +8,23 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-/*
- * frame_preprocessor.h
- */
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_FRAME_PREPROCESSOR_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_FRAME_PREPROCESSOR_H
-
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
-#include "webrtc/modules/video_processing/main/source/spatial_resampler.h"
-#include "webrtc/modules/video_processing/main/source/video_decimator.h"
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_FRAME_PREPROCESSOR_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_FRAME_PREPROCESSOR_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
+#include "webrtc/modules/video_processing/spatial_resampler.h"
+#include "webrtc/modules/video_processing/video_decimator.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
+class VideoDenoiser;
+
+// All pointers/members in this class are assumed to be protected by the class
+// owner.
 class VPMFramePreprocessor {
  public:
   VPMFramePreprocessor();
@@ -38,7 +41,8 @@ class VPMFramePreprocessor {
   void EnableContentAnalysis(bool enable);
 
   // Set target resolution: frame rate and dimension.
-  int32_t SetTargetResolution(uint32_t width, uint32_t height,
+  int32_t SetTargetResolution(uint32_t width,
+                              uint32_t height,
                               uint32_t frame_rate);
 
   // Set target frame rate.
@@ -50,14 +54,14 @@ class VPMFramePreprocessor {
   int32_t updateIncomingFrameSize(uint32_t width, uint32_t height);
 
   // Set decimated values: frame rate/dimension.
-  uint32_t Decimatedframe_rate();
-  uint32_t DecimatedWidth() const;
-  uint32_t DecimatedHeight() const;
+  uint32_t GetDecimatedFrameRate();
+  uint32_t GetDecimatedWidth() const;
+  uint32_t GetDecimatedHeight() const;
 
   // Preprocess output:
-  int32_t PreprocessFrame(const VideoFrame& frame,
-                          VideoFrame** processed_frame);
-  VideoContentMetrics* ContentMetrics() const;
+  void EnableDenosing(bool enable);
+  const VideoFrame* PreprocessFrame(const VideoFrame& frame);
+  VideoContentMetrics* GetContentMetrics() const;
 
  private:
   // The content does not change so much every frame, so to reduce complexity
@@ -65,15 +69,16 @@ class VPMFramePreprocessor {
   enum { kSkipFrameCA = 2 };
 
   VideoContentMetrics* content_metrics_;
+  VideoFrame denoised_frame_;
   VideoFrame resampled_frame_;
   VPMSpatialResampler* spatial_resampler_;
   VPMContentAnalysis* ca_;
   VPMVideoDecimator* vd_;
+  rtc::scoped_ptr<VideoDenoiser> denoiser_;
   bool enable_ca_;
-  int frame_cnt_;
-
+  uint32_t frame_cnt_;
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_FRAME_PREPROCESSOR_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_FRAME_PREPROCESSOR_H_
diff --git a/webrtc/modules/video_processing/include/video_processing.h b/webrtc/modules/video_processing/include/video_processing.h
new file mode 100644
index 0000000000..a8d6358887
--- /dev/null
+++ b/webrtc/modules/video_processing/include/video_processing.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
+#include "webrtc/video_frame.h"
+
+// The module is largely intended to process video streams, except functionality
+// provided by static functions which operate independent of previous frames. It
+// is recommended, but not required that a unique instance be used for each
+// concurrently processed stream. Similarly, it is recommended to call Reset()
+// before switching to a new stream, but this is not absolutely required.
+//
+// The module provides basic thread safety by permitting only a single function
+// to execute concurrently.
+
+namespace webrtc {
+
+class VideoProcessing {
+ public:
+  struct FrameStats {
+    uint32_t hist[256];  // Frame histogram.
+    uint32_t mean;
+    uint32_t sum;
+    uint32_t num_pixels;
+    uint32_t sub_sampling_factor;  // Sub-sampling factor, in powers of 2.
+  };
+
+  enum BrightnessWarning { kNoWarning, kDarkWarning, kBrightWarning };
+
+  static VideoProcessing* Create();
+  virtual ~VideoProcessing() {}
+
+  // Retrieves statistics for the input frame. This function must be used to
+  // prepare a FrameStats struct for use in certain VPM functions.
+  static void GetFrameStats(const VideoFrame& frame, FrameStats* stats);
+
+  // Checks the validity of a FrameStats struct. Currently, valid implies only
+  // that is had changed from its initialized state.
+  static bool ValidFrameStats(const FrameStats& stats);
+
+  static void ClearFrameStats(FrameStats* stats);
+
+  // Increases/decreases the luminance value. 'delta' can be in the range {}
+  static void Brighten(int delta, VideoFrame* frame);
+
+  // Detects and removes camera flicker from a video stream. Every frame from
+  // the stream must be passed in. A frame will only be altered if flicker has
+  // been detected. Has a fixed-point implementation.
+  // Frame statistics provided by GetFrameStats(). On return the stats will
+  // be reset to zero if the frame was altered. Call GetFrameStats() again
+  // if the statistics for the altered frame are required.
+  virtual int32_t Deflickering(VideoFrame* frame, FrameStats* stats) = 0;
+
+  // Detects if a video frame is excessively bright or dark. Returns a
+  // warning if this is the case. Multiple frames should be passed in before
+  // expecting a warning. Has a floating-point implementation.
+  virtual int32_t BrightnessDetection(const VideoFrame& frame,
+                                      const FrameStats& stats) = 0;
+
+  // The following functions refer to the pre-processor unit within VPM. The
+  // pre-processor perfoms spatial/temporal decimation and content analysis on
+  // the frames prior to encoding.
+
+  // Enable/disable temporal decimation
+  virtual void EnableTemporalDecimation(bool enable) = 0;
+
+  virtual int32_t SetTargetResolution(uint32_t width,
+                                      uint32_t height,
+                                      uint32_t frame_rate) = 0;
+
+  virtual void SetTargetFramerate(int frame_rate) = 0;
+
+  virtual uint32_t GetDecimatedFrameRate() = 0;
+  virtual uint32_t GetDecimatedWidth() const = 0;
+  virtual uint32_t GetDecimatedHeight() const = 0;
+
+  // Set the spatial resampling settings of the VPM according to
+  // VideoFrameResampling.
+  virtual void SetInputFrameResampleMode(
+      VideoFrameResampling resampling_mode) = 0;
+
+  virtual void EnableDenosing(bool enable) = 0;
+  virtual const VideoFrame* PreprocessFrame(const VideoFrame& frame) = 0;
+
+  virtual VideoContentMetrics* GetContentMetrics() const = 0;
+  virtual void EnableContentAnalysis(bool enable) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
diff --git a/webrtc/modules/video_processing/main/interface/video_processing_defines.h b/webrtc/modules/video_processing/include/video_processing_defines.h
index 93a0658966..9cc71bde27 100644
--- a/webrtc/modules/video_processing/main/interface/video_processing_defines.h
+++ b/webrtc/modules/video_processing/include/video_processing_defines.h
@@ -13,29 +13,29 @@
  * This header file includes the definitions used in the video processor module
  */
 
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
-#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_DEFINES_H_
 
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Error codes
-#define VPM_OK                   0
-#define VPM_GENERAL_ERROR       -1
-#define VPM_MEMORY              -2
-#define VPM_PARAMETER_ERROR     -3
-#define VPM_SCALE_ERROR         -4
-#define VPM_UNINITIALIZED       -5
-#define VPM_UNIMPLEMENTED       -6
+#define VPM_OK 0
+#define VPM_GENERAL_ERROR -1
+#define VPM_MEMORY -2
+#define VPM_PARAMETER_ERROR -3
+#define VPM_SCALE_ERROR -4
+#define VPM_UNINITIALIZED -5
+#define VPM_UNIMPLEMENTED -6
 
 enum VideoFrameResampling {
-  kNoRescaling,         // Disables rescaling.
-  kFastRescaling,       // Point filter.
-  kBiLinear,            // Bi-linear interpolation.
-  kBox,                 // Box inteprolation.
+  kNoRescaling,    // Disables rescaling.
+  kFastRescaling,  // Point filter.
+  kBiLinear,       // Bi-linear interpolation.
+  kBox,            // Box inteprolation.
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_DEFINES_H_
diff --git a/webrtc/modules/video_processing/main/interface/video_processing.h b/webrtc/modules/video_processing/main/interface/video_processing.h
deleted file mode 100644
index 30af99fb8e..0000000000
--- a/webrtc/modules/video_processing/main/interface/video_processing.h
+++ /dev/null
@@ -1,270 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_processing.h
- * This header file contains the API required for the video
- * processing module class.
- */
-
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
-#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
-
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
-#include "webrtc/video_frame.h"
-
-/**
-   The module is largely intended to process video streams, except functionality
-   provided by static functions which operate independent of previous frames. It
-   is recommended, but not required that a unique instance be used for each
-   concurrently processed stream. Similarly, it is recommended to call Reset()
-   before switching to a new stream, but this is not absolutely required.
-
-   The module provides basic thread safety by permitting only a single function
-   to execute concurrently.
-*/
-
-namespace webrtc {
-
-class VideoProcessingModule : public Module {
- public:
-  /**
-     Structure to hold frame statistics. Populate it with GetFrameStats().
-  */
-  struct FrameStats {
-      FrameStats() :
-          mean(0),
-          sum(0),
-          num_pixels(0),
-          subSamplWidth(0),
-          subSamplHeight(0) {
-    memset(hist, 0, sizeof(hist));
-  }
-
-  uint32_t hist[256];       // FRame histogram.
-  uint32_t mean;            // Frame Mean value.
-  uint32_t sum;             // Sum of frame.
-  uint32_t num_pixels;       // Number of pixels.
-  uint8_t  subSamplWidth;   // Subsampling rate of width in powers of 2.
-  uint8_t  subSamplHeight;  // Subsampling rate of height in powers of 2.
-};
-
-  /**
-     Specifies the warning types returned by BrightnessDetection().
-  */
-  enum BrightnessWarning {
-    kNoWarning,     // Frame has acceptable brightness.
-    kDarkWarning,   // Frame is too dark.
-    kBrightWarning  // Frame is too bright.
-  };
-
-  /*
-     Creates a VPM object.
-
-     \param[in] id
-         Unique identifier of this object.
-
-     \return Pointer to a VPM object.
-  */
-  static VideoProcessingModule* Create();
-
-  /**
-     Destroys a VPM object.
-
-     \param[in] module
-         Pointer to the VPM object to destroy.
-  */
-  static void Destroy(VideoProcessingModule* module);
-
-  /**
-     Not supported.
-  */
-  int64_t TimeUntilNextProcess() override { return -1; }
-
-  /**
-     Not supported.
-  */
-  int32_t Process() override { return -1; }
-
-  /**
-     Resets all processing components to their initial states. This should be
-     called whenever a new video stream is started.
-  */
-  virtual void Reset() = 0;
-
-  /**
-     Retrieves statistics for the input frame. This function must be used to
-     prepare a FrameStats struct for use in certain VPM functions.
-
-     \param[out] stats
-         The frame statistics will be stored here on return.
-
-     \param[in]  frame
-         Reference to the video frame.
-
-     \return 0 on success, -1 on failure.
-  */
-  static int32_t GetFrameStats(FrameStats* stats, const VideoFrame& frame);
-
-  /**
-     Checks the validity of a FrameStats struct. Currently, valid implies only
-     that is had changed from its initialized state.
-
-     \param[in] stats
-         Frame statistics.
-
-     \return True on valid stats, false on invalid stats.
-  */
-  static bool ValidFrameStats(const FrameStats& stats);
-
-  /**
-     Returns a FrameStats struct to its intialized state.
-
-     \param[in,out] stats
-         Frame statistics.
-  */
-  static void ClearFrameStats(FrameStats* stats);
-
-  /**
-     Increases/decreases the luminance value.
-
-     \param[in,out] frame
-         Pointer to the video frame.
-
-    \param[in] delta
-         The amount to change the chrominance value of every single pixel.
-         Can be < 0 also.
-
-     \return 0 on success, -1 on failure.
-  */
-  static int32_t Brighten(VideoFrame* frame, int delta);
-
-  /**
-     Detects and removes camera flicker from a video stream. Every frame from
-     the stream must be passed in. A frame will only be altered if flicker has
-     been detected. Has a fixed-point implementation.
-
-     \param[in,out] frame
-         Pointer to the video frame.
-
-     \param[in,out] stats
-         Frame statistics provided by GetFrameStats(). On return the stats will
-         be reset to zero if the frame was altered. Call GetFrameStats() again
-         if the statistics for the altered frame are required.
-
-     \return 0 on success, -1 on failure.
-  */
-  virtual int32_t Deflickering(VideoFrame* frame, FrameStats* stats) = 0;
-
-  /**
-     Detects if a video frame is excessively bright or dark. Returns a
-     warning if this is the case. Multiple frames should be passed in before
-     expecting a warning. Has a floating-point implementation.
-
-     \param[in] frame
-         Pointer to the video frame.
-
-     \param[in] stats
-         Frame statistics provided by GetFrameStats().
-
-     \return A member of BrightnessWarning on success, -1 on error
-  */
-  virtual int32_t BrightnessDetection(const VideoFrame& frame,
-                                      const FrameStats& stats) = 0;
-
-  /**
-  The following functions refer to the pre-processor unit within VPM. The
-  pre-processor perfoms spatial/temporal decimation and content analysis on
-  the frames prior to encoding.
-  */
-
-  /**
-  Enable/disable temporal decimation
-
-  \param[in] enable when true, temporal decimation is enabled
-  */
-  virtual void EnableTemporalDecimation(bool enable) = 0;
-
-  /**
- Set target resolution
-
- \param[in] width
- Target width
-
- \param[in] height
- Target height
-
-  \param[in] frame_rate
-  Target frame_rate
-
-  \return VPM_OK on success, a negative value on error (see error codes)
-
-  */
-  virtual int32_t SetTargetResolution(uint32_t width,
-                                      uint32_t height,
-                                      uint32_t frame_rate) = 0;
-
-  virtual void SetTargetFramerate(int frame_rate) {}
-
-  /**
-  Get decimated(target) frame rate
-  */
-  virtual uint32_t Decimatedframe_rate() = 0;
-
-  /**
-  Get decimated(target) frame width
-  */
-  virtual uint32_t DecimatedWidth() const = 0;
-
-  /**
-  Get decimated(target) frame height
-  */
-  virtual uint32_t DecimatedHeight() const = 0 ;
-
-  /**
-  Set the spatial resampling settings of the VPM: The resampler may either be
-  disabled or one of the following:
-  scaling to a close to target dimension followed by crop/pad
-
-  \param[in] resampling_mode
-  Set resampling mode (a member of VideoFrameResampling)
-  */
-  virtual void SetInputFrameResampleMode(VideoFrameResampling
-                                         resampling_mode) = 0;
-
-  /**
-  Get Processed (decimated) frame
-
-  \param[in] frame pointer to the video frame.
-  \param[in] processed_frame pointer (double) to the processed frame. If no
-             processing is required, processed_frame will be NULL.
-
-  \return VPM_OK on success, a negative value on error (see error codes)
-  */
-  virtual int32_t PreprocessFrame(const VideoFrame& frame,
-                                  VideoFrame** processed_frame) = 0;
-
-  /**
-  Return content metrics for the last processed frame
-  */
-  virtual VideoContentMetrics* ContentMetrics() const = 0 ;
-
-  /**
-  Enable content analysis
-  */
-  virtual void EnableContentAnalysis(bool enable) = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
diff --git a/webrtc/modules/video_processing/main/source/OWNERS b/webrtc/modules/video_processing/main/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/webrtc/modules/video_processing/main/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/webrtc/modules/video_processing/main/source/brighten.cc b/webrtc/modules/video_processing/main/source/brighten.cc
deleted file mode 100644
index 1fe813e7b0..0000000000
--- a/webrtc/modules/video_processing/main/source/brighten.cc
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_processing/main/source/brighten.h"
-
-#include <stdlib.h>
-
-namespace webrtc {
-namespace VideoProcessing {
-
-int32_t Brighten(VideoFrame* frame, int delta) {
-  assert(frame);
-  if (frame->IsZeroSize()) {
-    return VPM_PARAMETER_ERROR;
-  }
-  if (frame->width() <= 0 || frame->height() <= 0) {
-    return VPM_PARAMETER_ERROR;
-  }
-
-  int num_pixels = frame->width() * frame->height();
-
-  int look_up[256];
-  for (int i = 0; i < 256; i++) {
-    int val = i + delta;
-    look_up[i] = ((((val < 0) ? 0 : val) > 255) ? 255 : val);
-  }
-
-  uint8_t* temp_ptr = frame->buffer(kYPlane);
-
-  for (int i = 0; i < num_pixels; i++) {
-    *temp_ptr = static_cast<uint8_t>(look_up[*temp_ptr]);
-    temp_ptr++;
-  }
-  return VPM_OK;
-}
-
-}  // namespace VideoProcessing
-}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/main/source/brighten.h b/webrtc/modules/video_processing/main/source/brighten.h
deleted file mode 100644
index 151d7a3b51..0000000000
--- a/webrtc/modules/video_processing/main/source/brighten.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
-#define MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
-
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-namespace VideoProcessing {
-
-int32_t Brighten(VideoFrame* frame, int delta);
-
-}  // namespace VideoProcessing
-}  // namespace webrtc
-
-#endif  // MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
diff --git a/webrtc/modules/video_processing/main/source/video_processing_impl.cc b/webrtc/modules/video_processing/main/source/video_processing_impl.cc
deleted file mode 100644
index eaaf14f6ad..0000000000
--- a/webrtc/modules/video_processing/main/source/video_processing_impl.cc
+++ /dev/null
@@ -1,183 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "webrtc/modules/video_processing/main/source/video_processing_impl.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
-
-#include <assert.h>
-
-namespace webrtc {
-
-namespace {
-void  SetSubSampling(VideoProcessingModule::FrameStats* stats,
-                     const int32_t width,
-                     const int32_t height) {
-  if (width * height >= 640 * 480) {
-    stats->subSamplWidth = 3;
-    stats->subSamplHeight = 3;
-  } else if (width * height >= 352 * 288) {
-    stats->subSamplWidth = 2;
-    stats->subSamplHeight = 2;
-  } else if (width * height >= 176 * 144) {
-    stats->subSamplWidth = 1;
-    stats->subSamplHeight = 1;
-  } else {
-    stats->subSamplWidth = 0;
-    stats->subSamplHeight = 0;
-  }
-}
-}  // namespace
-
-VideoProcessingModule* VideoProcessingModule::Create() {
-  return new VideoProcessingModuleImpl();
-}
-
-void VideoProcessingModule::Destroy(VideoProcessingModule* module) {
-  if (module)
-    delete static_cast<VideoProcessingModuleImpl*>(module);
-}
-
-VideoProcessingModuleImpl::VideoProcessingModuleImpl() {}
-VideoProcessingModuleImpl::~VideoProcessingModuleImpl() {}
-
-void VideoProcessingModuleImpl::Reset() {
-  rtc::CritScope mutex(&mutex_);
-  deflickering_.Reset();
-  brightness_detection_.Reset();
-  frame_pre_processor_.Reset();
-}
-
-int32_t VideoProcessingModule::GetFrameStats(FrameStats* stats,
-                                             const VideoFrame& frame) {
-  if (frame.IsZeroSize()) {
-    LOG(LS_ERROR) << "Zero size frame.";
-    return VPM_PARAMETER_ERROR;
-  }
-
-  int width = frame.width();
-  int height = frame.height();
-
-  ClearFrameStats(stats);  // The histogram needs to be zeroed out.
-  SetSubSampling(stats, width, height);
-
-  const uint8_t* buffer = frame.buffer(kYPlane);
-  // Compute histogram and sum of frame
-  for (int i = 0; i < height; i += (1 << stats->subSamplHeight)) {
-    int k = i * width;
-    for (int j = 0; j < width; j += (1 << stats->subSamplWidth)) {
-      stats->hist[buffer[k + j]]++;
-      stats->sum += buffer[k + j];
-    }
-  }
-
-  stats->num_pixels = (width * height) / ((1 << stats->subSamplWidth) *
-                     (1 << stats->subSamplHeight));
-  assert(stats->num_pixels > 0);
-
-  // Compute mean value of frame
-  stats->mean = stats->sum / stats->num_pixels;
-
-  return VPM_OK;
-}
-
-bool VideoProcessingModule::ValidFrameStats(const FrameStats& stats) {
-  if (stats.num_pixels == 0) {
-    LOG(LS_WARNING) << "Invalid frame stats.";
-    return false;
-  }
-  return true;
-}
-
-void VideoProcessingModule::ClearFrameStats(FrameStats* stats) {
-  stats->mean = 0;
-  stats->sum = 0;
-  stats->num_pixels = 0;
-  stats->subSamplWidth = 0;
-  stats->subSamplHeight = 0;
-  memset(stats->hist, 0, sizeof(stats->hist));
-}
-
-int32_t VideoProcessingModule::Brighten(VideoFrame* frame, int delta) {
-  return VideoProcessing::Brighten(frame, delta);
-}
-
-int32_t VideoProcessingModuleImpl::Deflickering(VideoFrame* frame,
-                                                FrameStats* stats) {
-  rtc::CritScope mutex(&mutex_);
-  return deflickering_.ProcessFrame(frame, stats);
-}
-
-int32_t VideoProcessingModuleImpl::BrightnessDetection(
-    const VideoFrame& frame,
-    const FrameStats& stats) {
-  rtc::CritScope mutex(&mutex_);
-  return brightness_detection_.ProcessFrame(frame, stats);
-}
-
-
-void VideoProcessingModuleImpl::EnableTemporalDecimation(bool enable) {
-  rtc::CritScope mutex(&mutex_);
-  frame_pre_processor_.EnableTemporalDecimation(enable);
-}
-
-
-void VideoProcessingModuleImpl::SetInputFrameResampleMode(VideoFrameResampling
-                                                          resampling_mode) {
-  rtc::CritScope cs(&mutex_);
-  frame_pre_processor_.SetInputFrameResampleMode(resampling_mode);
-}
-
-int32_t VideoProcessingModuleImpl::SetTargetResolution(uint32_t width,
-                                                       uint32_t height,
-                                                       uint32_t frame_rate) {
-  rtc::CritScope cs(&mutex_);
-  return frame_pre_processor_.SetTargetResolution(width, height, frame_rate);
-}
-
-void VideoProcessingModuleImpl::SetTargetFramerate(int frame_rate) {
-  rtc::CritScope cs(&mutex_);
-  frame_pre_processor_.SetTargetFramerate(frame_rate);
-}
-
-uint32_t VideoProcessingModuleImpl::Decimatedframe_rate() {
-  rtc::CritScope cs(&mutex_);
-  return  frame_pre_processor_.Decimatedframe_rate();
-}
-
-uint32_t VideoProcessingModuleImpl::DecimatedWidth() const {
-  rtc::CritScope cs(&mutex_);
-  return frame_pre_processor_.DecimatedWidth();
-}
-
-uint32_t VideoProcessingModuleImpl::DecimatedHeight() const {
-  rtc::CritScope cs(&mutex_);
-  return frame_pre_processor_.DecimatedHeight();
-}
-
-int32_t VideoProcessingModuleImpl::PreprocessFrame(
-    const VideoFrame& frame,
-    VideoFrame** processed_frame) {
-  rtc::CritScope mutex(&mutex_);
-  return frame_pre_processor_.PreprocessFrame(frame, processed_frame);
-}
-
-VideoContentMetrics* VideoProcessingModuleImpl::ContentMetrics() const {
-  rtc::CritScope mutex(&mutex_);
-  return frame_pre_processor_.ContentMetrics();
-}
-
-void VideoProcessingModuleImpl::EnableContentAnalysis(bool enable) {
-  rtc::CritScope mutex(&mutex_);
-  frame_pre_processor_.EnableContentAnalysis(enable);
-}
-
-}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/main/source/video_processing_impl.h b/webrtc/modules/video_processing/main/source/video_processing_impl.h
deleted file mode 100644
index fed5197f49..0000000000
--- a/webrtc/modules/video_processing/main/source/video_processing_impl.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
-#define WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/brighten.h"
-#include "webrtc/modules/video_processing/main/source/brightness_detection.h"
-#include "webrtc/modules/video_processing/main/source/deflickering.h"
-#include "webrtc/modules/video_processing/main/source/frame_preprocessor.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class VideoProcessingModuleImpl : public VideoProcessingModule {
- public:
-  VideoProcessingModuleImpl();
-  ~VideoProcessingModuleImpl() override;
-
-  void Reset() override;
-
-  int32_t Deflickering(VideoFrame* frame, FrameStats* stats) override;
-
-  int32_t BrightnessDetection(const VideoFrame& frame,
-                              const FrameStats& stats) override;
-
-  // Frame pre-processor functions
-
-  // Enable temporal decimation
-  void EnableTemporalDecimation(bool enable) override;
-
-  void SetInputFrameResampleMode(VideoFrameResampling resampling_mode) override;
-
-  // Enable content analysis
-  void EnableContentAnalysis(bool enable) override;
-
-  // Set Target Resolution: frame rate and dimension
-  int32_t SetTargetResolution(uint32_t width,
-                              uint32_t height,
-                              uint32_t frame_rate) override;
-
-  void SetTargetFramerate(int frame_rate) override;
-
-  // Get decimated values: frame rate/dimension
-  uint32_t Decimatedframe_rate() override;
-  uint32_t DecimatedWidth() const override;
-  uint32_t DecimatedHeight() const override;
-
-  // Preprocess:
-  // Pre-process incoming frame: Sample when needed and compute content
-  // metrics when enabled.
-  // If no resampling takes place - processed_frame is set to NULL.
-  int32_t PreprocessFrame(const VideoFrame& frame,
-                          VideoFrame** processed_frame) override;
-  VideoContentMetrics* ContentMetrics() const override;
-
- private:
-  mutable rtc::CriticalSection mutex_;
-  VPMDeflickering deflickering_ GUARDED_BY(mutex_);
-  VPMBrightnessDetection brightness_detection_;
-  VPMFramePreprocessor frame_pre_processor_;
-};
-
-}  // namespace
-
-#endif
diff --git a/webrtc/modules/video_processing/main/test/unit_test/brightness_detection_test.cc b/webrtc/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
deleted file mode 100644
index 4d0de3ac98..0000000000
--- a/webrtc/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-using namespace webrtc;
-
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(BrightnessDetection))
-{
-    uint32_t frameNum = 0;
-    int32_t brightnessWarning = 0;
-    uint32_t warningCount = 0;
-    rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-           frame_length_)
-    {
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        frameNum++;
-        VideoProcessingModule::FrameStats stats;
-        ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-        ASSERT_GE(brightnessWarning = vpm_->BrightnessDetection(video_frame_,
-                                                                stats), 0);
-        if (brightnessWarning != VideoProcessingModule::kNoWarning)
-        {
-            warningCount++;
-        }
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    // Expect few warnings
-    float warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-    printf("\nWarning proportions:\n");
-    printf("Stock foreman: %.1f %%\n", warningProportion);
-    EXPECT_LT(warningProportion, 10);
-
-    rewind(source_file_);
-    frameNum = 0;
-    warningCount = 0;
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-        frame_length_ &&
-        frameNum < 300)
-    {
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        frameNum++;
-
-        uint8_t* frame = video_frame_.buffer(kYPlane);
-        uint32_t yTmp = 0;
-        for (int yIdx = 0; yIdx < width_ * height_; yIdx++)
-        {
-            yTmp = frame[yIdx] << 1;
-            if (yTmp > 255)
-            {
-                yTmp = 255;
-            }
-            frame[yIdx] = static_cast<uint8_t>(yTmp);
-        }
-
-        VideoProcessingModule::FrameStats stats;
-        ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-        ASSERT_GE(brightnessWarning = vpm_->BrightnessDetection(video_frame_,
-                                                                stats), 0);
-        EXPECT_NE(VideoProcessingModule::kDarkWarning, brightnessWarning);
-        if (brightnessWarning == VideoProcessingModule::kBrightWarning)
-        {
-            warningCount++;
-        }
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    // Expect many brightness warnings
-    warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-    printf("Bright foreman: %.1f %%\n", warningProportion);
-    EXPECT_GT(warningProportion, 95);
-
-    rewind(source_file_);
-    frameNum = 0;
-    warningCount = 0;
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-        frame_length_ && frameNum < 300)
-    {
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        frameNum++;
-
-        uint8_t* y_plane = video_frame_.buffer(kYPlane);
-        int32_t yTmp = 0;
-        for (int yIdx = 0; yIdx < width_ * height_; yIdx++)
-        {
-            yTmp = y_plane[yIdx] >> 1;
-            y_plane[yIdx] = static_cast<uint8_t>(yTmp);
-        }
-
-        VideoProcessingModule::FrameStats stats;
-        ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-        ASSERT_GE(brightnessWarning = vpm_->BrightnessDetection(video_frame_,
-                                                                stats), 0);
-        EXPECT_NE(VideoProcessingModule::kBrightWarning, brightnessWarning);
-        if (brightnessWarning == VideoProcessingModule::kDarkWarning)
-        {
-            warningCount++;
-        }
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    // Expect many darkness warnings
-    warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-    printf("Dark foreman: %.1f %%\n\n", warningProportion);
-    EXPECT_GT(warningProportion, 90);
-}
diff --git a/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc b/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc
deleted file mode 100644
index 83d09ef486..0000000000
--- a/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(Deflickering))
-{
-    enum { NumRuns = 30 };
-    uint32_t frameNum = 0;
-    const uint32_t frame_rate = 15;
-
-    int64_t min_runtime = 0;
-    int64_t avg_runtime = 0;
-
-    // Close automatically opened Foreman.
-    fclose(source_file_);
-    const std::string input_file =
-        webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
-    source_file_  = fopen(input_file.c_str(), "rb");
-    ASSERT_TRUE(source_file_ != NULL) <<
-        "Cannot read input file: " << input_file << "\n";
-
-    const std::string output_file =
-        webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
-    FILE* deflickerFile = fopen(output_file.c_str(), "wb");
-    ASSERT_TRUE(deflickerFile != NULL) <<
-        "Could not open output file: " << output_file << "\n";
-
-    printf("\nRun time [us / frame]:\n");
-    rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-    for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++)
-    {
-        TickTime t0;
-        TickTime t1;
-        TickInterval acc_ticks;
-        uint32_t timeStamp = 1;
-
-        frameNum = 0;
-        while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-               frame_length_)
-        {
-            frameNum++;
-            EXPECT_EQ(
-                0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-            video_frame_.set_timestamp(timeStamp);
-
-            t0 = TickTime::Now();
-            VideoProcessingModule::FrameStats stats;
-            ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-            ASSERT_EQ(0, vpm_->Deflickering(&video_frame_, &stats));
-            t1 = TickTime::Now();
-            acc_ticks += (t1 - t0);
-
-            if (run_idx == 0)
-            {
-              if (PrintVideoFrame(video_frame_, deflickerFile) < 0) {
-                return;
-              }
-            }
-            timeStamp += (90000 / frame_rate);
-        }
-        ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-        printf("%u\n", static_cast<int>(acc_ticks.Microseconds() / frameNum));
-        if (acc_ticks.Microseconds() < min_runtime || run_idx == 0)
-        {
-            min_runtime = acc_ticks.Microseconds();
-        }
-        avg_runtime += acc_ticks.Microseconds();
-
-        rewind(source_file_);
-    }
-    ASSERT_EQ(0, fclose(deflickerFile));
-    // TODO(kjellander): Add verification of deflicker output file.
-
-    printf("\nAverage run time = %d us / frame\n",
-        static_cast<int>(avg_runtime / frameNum / NumRuns));
-    printf("Min run time = %d us / frame\n\n",
-        static_cast<int>(min_runtime / frameNum));
-}
-
-}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/main/source/spatial_resampler.cc b/webrtc/modules/video_processing/spatial_resampler.cc
index 9360e68b41..cdbe0efac1 100644
--- a/webrtc/modules/video_processing/main/source/spatial_resampler.cc
+++ b/webrtc/modules/video_processing/spatial_resampler.cc
@@ -8,8 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/source/spatial_resampler.h"
-
+#include "webrtc/modules/video_processing/spatial_resampler.h"
 
 namespace webrtc {
 
@@ -21,12 +20,13 @@ VPMSimpleSpatialResampler::VPMSimpleSpatialResampler()
 
 VPMSimpleSpatialResampler::~VPMSimpleSpatialResampler() {}
 
-
 int32_t VPMSimpleSpatialResampler::SetTargetFrameSize(int32_t width,
                                                       int32_t height) {
-  if (resampling_mode_ == kNoRescaling) return VPM_OK;
+  if (resampling_mode_ == kNoRescaling)
+    return VPM_OK;
 
-  if (width < 1 || height < 1) return VPM_PARAMETER_ERROR;
+  if (width < 1 || height < 1)
+    return VPM_PARAMETER_ERROR;
 
   target_width_ = width;
   target_height_ = height;
@@ -48,11 +48,11 @@ void VPMSimpleSpatialResampler::Reset() {
 int32_t VPMSimpleSpatialResampler::ResampleFrame(const VideoFrame& inFrame,
                                                  VideoFrame* outFrame) {
   // Don't copy if frame remains as is.
-  if (resampling_mode_ == kNoRescaling)
-     return VPM_OK;
+  if (resampling_mode_ == kNoRescaling) {
+    return VPM_OK;
   // Check if re-sampling is needed
-  else if ((inFrame.width() == target_width_) &&
-    (inFrame.height() == target_height_))  {
+  } else if ((inFrame.width() == target_width_) &&
+             (inFrame.height() == target_height_)) {
     return VPM_OK;
   }
 
@@ -60,8 +60,8 @@ int32_t VPMSimpleSpatialResampler::ResampleFrame(const VideoFrame& inFrame,
   // TODO(mikhal/marpan): Should we allow for setting the filter mode in
   // _scale.Set() with |resampling_mode_|?
   int ret_val = 0;
-  ret_val = scaler_.Set(inFrame.width(), inFrame.height(),
-                       target_width_, target_height_, kI420, kI420, kScaleBox);
+  ret_val = scaler_.Set(inFrame.width(), inFrame.height(), target_width_,
+                        target_height_, kI420, kI420, kScaleBox);
   if (ret_val < 0)
     return ret_val;
 
@@ -86,10 +86,9 @@ int32_t VPMSimpleSpatialResampler::TargetWidth() {
   return target_width_;
 }
 
-bool VPMSimpleSpatialResampler::ApplyResample(int32_t width,
-                                              int32_t height) {
+bool VPMSimpleSpatialResampler::ApplyResample(int32_t width, int32_t height) {
   if ((width == target_width_ && height == target_height_) ||
-       resampling_mode_ == kNoRescaling)
+      resampling_mode_ == kNoRescaling)
     return false;
   else
     return true;
diff --git a/webrtc/modules/video_processing/main/source/spatial_resampler.h b/webrtc/modules/video_processing/spatial_resampler.h
index f965a40a83..51820e24e5 100644
--- a/webrtc/modules/video_processing/main/source/spatial_resampler.h
+++ b/webrtc/modules/video_processing/spatial_resampler.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_SPATIAL_RESAMPLER_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_SPATIAL_RESAMPLER_H_
 
 #include "webrtc/typedefs.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
 
 #include "webrtc/common_video/libyuv/include/scaler.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@@ -23,10 +23,10 @@ namespace webrtc {
 
 class VPMSpatialResampler {
  public:
-  virtual ~VPMSpatialResampler() {};
+  virtual ~VPMSpatialResampler() {}
   virtual int32_t SetTargetFrameSize(int32_t width, int32_t height) = 0;
-  virtual void SetInputFrameResampleMode(VideoFrameResampling
-                                         resampling_mode) = 0;
+  virtual void SetInputFrameResampleMode(
+      VideoFrameResampling resampling_mode) = 0;
   virtual void Reset() = 0;
   virtual int32_t ResampleFrame(const VideoFrame& inFrame,
                                 VideoFrame* outFrame) = 0;
@@ -49,13 +49,12 @@ class VPMSimpleSpatialResampler : public VPMSpatialResampler {
   virtual bool ApplyResample(int32_t width, int32_t height);
 
  private:
-
-  VideoFrameResampling        resampling_mode_;
-  int32_t                     target_width_;
-  int32_t                     target_height_;
-  Scaler                      scaler_;
+  VideoFrameResampling resampling_mode_;
+  int32_t target_width_;
+  int32_t target_height_;
+  Scaler scaler_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_SPATIAL_RESAMPLER_H_
diff --git a/webrtc/modules/video_processing/test/brightness_detection_test.cc b/webrtc/modules/video_processing/test/brightness_detection_test.cc
new file mode 100644
index 0000000000..669bb183e5
--- /dev/null
+++ b/webrtc/modules/video_processing/test/brightness_detection_test.cc
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_BrightnessDetection DISABLED_BrightnessDetection
+#else
+#define MAYBE_BrightnessDetection BrightnessDetection
+#endif
+TEST_F(VideoProcessingTest, MAYBE_BrightnessDetection) {
+  uint32_t frameNum = 0;
+  int32_t brightnessWarning = 0;
+  uint32_t warningCount = 0;
+  rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+         frame_length_) {
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+    frameNum++;
+    VideoProcessing::FrameStats stats;
+    vp_->GetFrameStats(video_frame_, &stats);
+    EXPECT_GT(stats.num_pixels, 0u);
+    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
+              0);
+    if (brightnessWarning != VideoProcessing::kNoWarning) {
+      warningCount++;
+    }
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+  // Expect few warnings
+  float warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+  printf("\nWarning proportions:\n");
+  printf("Stock foreman: %.1f %%\n", warningProportion);
+  EXPECT_LT(warningProportion, 10);
+
+  rewind(source_file_);
+  frameNum = 0;
+  warningCount = 0;
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+             frame_length_ &&
+         frameNum < 300) {
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+    frameNum++;
+
+    uint8_t* frame = video_frame_.buffer(kYPlane);
+    uint32_t yTmp = 0;
+    for (int yIdx = 0; yIdx < width_ * height_; yIdx++) {
+      yTmp = frame[yIdx] << 1;
+      if (yTmp > 255) {
+        yTmp = 255;
+      }
+      frame[yIdx] = static_cast<uint8_t>(yTmp);
+    }
+
+    VideoProcessing::FrameStats stats;
+    vp_->GetFrameStats(video_frame_, &stats);
+    EXPECT_GT(stats.num_pixels, 0u);
+    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
+              0);
+    EXPECT_NE(VideoProcessing::kDarkWarning, brightnessWarning);
+    if (brightnessWarning == VideoProcessing::kBrightWarning) {
+      warningCount++;
+    }
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+  // Expect many brightness warnings
+  warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+  printf("Bright foreman: %.1f %%\n", warningProportion);
+  EXPECT_GT(warningProportion, 95);
+
+  rewind(source_file_);
+  frameNum = 0;
+  warningCount = 0;
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+             frame_length_ &&
+         frameNum < 300) {
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+    frameNum++;
+
+    uint8_t* y_plane = video_frame_.buffer(kYPlane);
+    int32_t yTmp = 0;
+    for (int yIdx = 0; yIdx < width_ * height_; yIdx++) {
+      yTmp = y_plane[yIdx] >> 1;
+      y_plane[yIdx] = static_cast<uint8_t>(yTmp);
+    }
+
+    VideoProcessing::FrameStats stats;
+    vp_->GetFrameStats(video_frame_, &stats);
+    EXPECT_GT(stats.num_pixels, 0u);
+    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
+              0);
+    EXPECT_NE(VideoProcessing::kBrightWarning, brightnessWarning);
+    if (brightnessWarning == VideoProcessing::kDarkWarning) {
+      warningCount++;
+    }
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+  // Expect many darkness warnings
+  warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+  printf("Dark foreman: %.1f %%\n\n", warningProportion);
+  EXPECT_GT(warningProportion, 90);
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/main/test/unit_test/content_metrics_test.cc b/webrtc/modules/video_processing/test/content_metrics_test.cc
index d9c1309d9b..782f9cff59 100644
--- a/webrtc/modules/video_processing/main/test/unit_test/content_metrics_test.cc
+++ b/webrtc/modules/video_processing/test/content_metrics_test.cc
@@ -9,28 +9,32 @@
  */
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
 
 namespace webrtc {
 
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(ContentAnalysis)) {
-  VPMContentAnalysis    ca__c(false);
-  VPMContentAnalysis    ca__sse(true);
-  VideoContentMetrics  *_cM_c, *_cM_SSE;
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_ContentAnalysis) {
+#else
+TEST_F(VideoProcessingTest, ContentAnalysis) {
+#endif
+  VPMContentAnalysis ca__c(false);
+  VPMContentAnalysis ca__sse(true);
+  VideoContentMetrics* _cM_c;
+  VideoContentMetrics* _cM_SSE;
 
-  ca__c.Initialize(width_,height_);
-  ca__sse.Initialize(width_,height_);
+  ca__c.Initialize(width_, height_);
+  ca__sse.Initialize(width_, height_);
 
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  while (fread(video_buffer.get(), 1, frame_length_, source_file_)
-       == frame_length_) {
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+         frame_length_) {
     // Using ConvertToI420 to add stride to the image.
     EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                                0, kVideoRotation_0, &video_frame_));
-    _cM_c   = ca__c.ComputeContentMetrics(video_frame_);
+    _cM_c = ca__c.ComputeContentMetrics(video_frame_);
     _cM_SSE = ca__sse.ComputeContentMetrics(video_frame_);
 
     ASSERT_EQ(_cM_c->spatial_pred_err, _cM_SSE->spatial_pred_err);
diff --git a/webrtc/modules/video_processing/main/test/unit_test/createTable.m b/webrtc/modules/video_processing/test/createTable.m
index 2c7fb522f6..fe8777ee71 100644
--- a/webrtc/modules/video_processing/main/test/unit_test/createTable.m
+++ b/webrtc/modules/video_processing/test/createTable.m
@@ -31,7 +31,7 @@ A=(1-B)/r0;
 f0=A*x0.^2+B*x0; % compander function in zone 1
 
 % equation system for finding second zone parameters
-M=[r0^3 r0^2 r0 1; 
+M=[r0^3 r0^2 r0 1;
     3*r0^2 2*r0 1 0;
     3*r1^2 2*r1 1 0;
     r1^3 r1^2 r1 1];
@@ -173,7 +173,7 @@ for k=1:size(y,3)
         end
     end
 end
-      
+
 fprintf('\nWriting modified test file...')
 writeYUV420file('../out/Debug/foremanColorEnhanced.yuv',y,unew,vnew);
 fprintf(' done\n');
diff --git a/webrtc/modules/video_processing/test/deflickering_test.cc b/webrtc/modules/video_processing/test/deflickering_test.cc
new file mode 100644
index 0000000000..5410015b06
--- /dev/null
+++ b/webrtc/modules/video_processing/test/deflickering_test.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_Deflickering) {
+#else
+TEST_F(VideoProcessingTest, Deflickering) {
+#endif
+  enum { NumRuns = 30 };
+  uint32_t frameNum = 0;
+  const uint32_t frame_rate = 15;
+
+  int64_t min_runtime = 0;
+  int64_t avg_runtime = 0;
+
+  // Close automatically opened Foreman.
+  fclose(source_file_);
+  const std::string input_file =
+      webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
+  source_file_ = fopen(input_file.c_str(), "rb");
+  ASSERT_TRUE(source_file_ != NULL) << "Cannot read input file: " << input_file
+                                    << "\n";
+
+  const std::string output_file =
+      webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
+  FILE* deflickerFile = fopen(output_file.c_str(), "wb");
+  ASSERT_TRUE(deflickerFile != NULL)
+      << "Could not open output file: " << output_file << "\n";
+
+  printf("\nRun time [us / frame]:\n");
+  rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
+  for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++) {
+    TickTime t0;
+    TickTime t1;
+    TickInterval acc_ticks;
+    uint32_t timeStamp = 1;
+
+    frameNum = 0;
+    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+           frame_length_) {
+      frameNum++;
+      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
+                                 height_, 0, kVideoRotation_0, &video_frame_));
+      video_frame_.set_timestamp(timeStamp);
+
+      t0 = TickTime::Now();
+      VideoProcessing::FrameStats stats;
+      vp_->GetFrameStats(video_frame_, &stats);
+      EXPECT_GT(stats.num_pixels, 0u);
+      ASSERT_EQ(0, vp_->Deflickering(&video_frame_, &stats));
+      t1 = TickTime::Now();
+      acc_ticks += (t1 - t0);
+
+      if (run_idx == 0) {
+        if (PrintVideoFrame(video_frame_, deflickerFile) < 0) {
+          return;
+        }
+      }
+      timeStamp += (90000 / frame_rate);
+    }
+    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+    printf("%u\n", static_cast<int>(acc_ticks.Microseconds() / frameNum));
+    if (acc_ticks.Microseconds() < min_runtime || run_idx == 0) {
+      min_runtime = acc_ticks.Microseconds();
+    }
+    avg_runtime += acc_ticks.Microseconds();
+
+    rewind(source_file_);
+  }
+  ASSERT_EQ(0, fclose(deflickerFile));
+  // TODO(kjellander): Add verification of deflicker output file.
+
+  printf("\nAverage run time = %d us / frame\n",
+         static_cast<int>(avg_runtime / frameNum / NumRuns));
+  printf("Min run time = %d us / frame\n\n",
+         static_cast<int>(min_runtime / frameNum));
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/test/denoiser_test.cc b/webrtc/modules/video_processing/test/denoiser_test.cc
new file mode 100644
index 0000000000..551a77617d
--- /dev/null
+++ b/webrtc/modules/video_processing/test/denoiser_test.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+#include "webrtc/modules/video_processing/video_denoiser.h"
+
+namespace webrtc {
+
+TEST_F(VideoProcessingTest, CopyMem) {
+  rtc::scoped_ptr<DenoiserFilter> df_c(DenoiserFilter::Create(false));
+  rtc::scoped_ptr<DenoiserFilter> df_sse_neon(DenoiserFilter::Create(true));
+  uint8_t src[16 * 16], dst[16 * 16];
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      src[i * 16 + j] = i * 16 + j;
+    }
+  }
+
+  memset(dst, 0, 8 * 8);
+  df_c->CopyMem8x8(src, 8, dst, 8);
+  EXPECT_EQ(0, memcmp(src, dst, 8 * 8));
+
+  memset(dst, 0, 16 * 16);
+  df_c->CopyMem16x16(src, 16, dst, 16);
+  EXPECT_EQ(0, memcmp(src, dst, 16 * 16));
+
+  memset(dst, 0, 8 * 8);
+  df_sse_neon->CopyMem16x16(src, 8, dst, 8);
+  EXPECT_EQ(0, memcmp(src, dst, 8 * 8));
+
+  memset(dst, 0, 16 * 16);
+  df_sse_neon->CopyMem16x16(src, 16, dst, 16);
+  EXPECT_EQ(0, memcmp(src, dst, 16 * 16));
+}
+
+TEST_F(VideoProcessingTest, Variance) {
+  rtc::scoped_ptr<DenoiserFilter> df_c(DenoiserFilter::Create(false));
+  rtc::scoped_ptr<DenoiserFilter> df_sse_neon(DenoiserFilter::Create(true));
+  uint8_t src[16 * 16], dst[16 * 16];
+  uint32_t sum = 0, sse = 0, var;
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      src[i * 16 + j] = i * 16 + j;
+    }
+  }
+  // Compute the 16x8 variance of the 16x16 block.
+  for (int i = 0; i < 8; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      sum += (i * 32 + j);
+      sse += (i * 32 + j) * (i * 32 + j);
+    }
+  }
+  var = sse - ((sum * sum) >> 7);
+  memset(dst, 0, 16 * 16);
+  EXPECT_EQ(var, df_c->Variance16x8(src, 16, dst, 16, &sse));
+  EXPECT_EQ(var, df_sse_neon->Variance16x8(src, 16, dst, 16, &sse));
+}
+
+TEST_F(VideoProcessingTest, MbDenoise) {
+  rtc::scoped_ptr<DenoiserFilter> df_c(DenoiserFilter::Create(false));
+  rtc::scoped_ptr<DenoiserFilter> df_sse_neon(DenoiserFilter::Create(true));
+  uint8_t running_src[16 * 16], src[16 * 16], dst[16 * 16], dst_ref[16 * 16];
+
+  // Test case: |diff| <= |3 + shift_inc1|
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 2;
+      dst_ref[i * 16 + j] = running_src[i * 16 + j];
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+
+  // Test case: |diff| >= |4 + shift_inc1|
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 5;
+      dst_ref[i * 16 + j] = src[i * 16 + j] - 2;
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+  memset(dst, 0, 16 * 16);
+  df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+
+  // Test case: |diff| >= 8
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 8;
+      dst_ref[i * 16 + j] = src[i * 16 + j] - 6;
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+  memset(dst, 0, 16 * 16);
+  df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+
+  // Test case: |diff| > 15
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 16;
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  DenoiserDecision decision =
+      df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(COPY_BLOCK, decision);
+  decision = df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(COPY_BLOCK, decision);
+}
+
+TEST_F(VideoProcessingTest, Denoiser) {
+  // Create pure C denoiser.
+  VideoDenoiser denoiser_c(false);
+  // Create SSE or NEON denoiser.
+  VideoDenoiser denoiser_sse_neon(true);
+  VideoFrame denoised_frame_c;
+  VideoFrame denoised_frame_sse_neon;
+
+  rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+         frame_length_) {
+    // Using ConvertToI420 to add stride to the image.
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+
+    denoiser_c.DenoiseFrame(video_frame_, &denoised_frame_c);
+    denoiser_sse_neon.DenoiseFrame(video_frame_, &denoised_frame_sse_neon);
+
+    // Denoising results should be the same for C and SSE/NEON denoiser.
+    ASSERT_EQ(true, denoised_frame_c.EqualsFrame(denoised_frame_sse_neon));
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/main/test/unit_test/readYUV420file.m b/webrtc/modules/video_processing/test/readYUV420file.m
index 03013efd3a..f409820283 100644
--- a/webrtc/modules/video_processing/main/test/unit_test/readYUV420file.m
+++ b/webrtc/modules/video_processing/test/readYUV420file.m
@@ -10,7 +10,7 @@ end
 nPx=width*height;
 
 % nPx bytes luminance, nPx/4 bytes U, nPx/4 bytes V
-frameSizeBytes = nPx*1.5; 
+frameSizeBytes = nPx*1.5;
 
 % calculate number of frames
 fseek(fid,0,'eof'); % move to end of file
@@ -27,19 +27,19 @@ V=uint8(zeros(height/2,width/2,numFrames));
 [X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
 
 for k=1:numFrames
-    
+
     % Store luminance
     Y(:,:,k)=uint8(reshape(X(1:nPx), width, height).');
-    
+
     % Store U channel
     U(:,:,k)=uint8(reshape(X(nPx + (1:nPx/4)), width/2, height/2).');
 
     % Store V channel
     V(:,:,k)=uint8(reshape(X(nPx + nPx/4 + (1:nPx/4)), width/2, height/2).');
-    
+
     % Read next frame
     [X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
 end
 
-    
+
 fclose(fid);
diff --git a/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.cc b/webrtc/modules/video_processing/test/video_processing_unittest.cc
index 11ccc4891b..2fd8fb6673 100644
--- a/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
+++ b/webrtc/modules/video_processing/test/video_processing_unittest.cc
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+
+#include <gflags/gflags.h>
 
 #include <string>
 
-#include <gflags/gflags.h>
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -30,8 +30,8 @@ DEFINE_bool(gen_files, false, "Output files for visual inspection.");
 static void PreprocessFrameAndVerify(const VideoFrame& source,
                                      int target_width,
                                      int target_height,
-                                     VideoProcessingModule* vpm,
-                                     VideoFrame** out_frame);
+                                     VideoProcessing* vpm,
+                                     const VideoFrame* out_frame);
 static void CropFrame(const uint8_t* source_data,
                       int source_width,
                       int source_height,
@@ -49,14 +49,14 @@ static void TestSize(const VideoFrame& source_frame,
                      int target_width,
                      int target_height,
                      double expected_psnr,
-                     VideoProcessingModule* vpm);
+                     VideoProcessing* vpm);
 static bool CompareFrames(const webrtc::VideoFrame& frame1,
                           const webrtc::VideoFrame& frame2);
 static void WriteProcessedFrameForVisualInspection(const VideoFrame& source,
                                                    const VideoFrame& processed);
 
-VideoProcessingModuleTest::VideoProcessingModuleTest()
-    : vpm_(NULL),
+VideoProcessingTest::VideoProcessingTest()
+    : vp_(NULL),
       source_file_(NULL),
       width_(352),
       half_width_((width_ + 1) / 2),
@@ -65,155 +65,179 @@ VideoProcessingModuleTest::VideoProcessingModuleTest()
       size_uv_(half_width_ * ((height_ + 1) / 2)),
       frame_length_(CalcBufferSize(kI420, width_, height_)) {}
 
-void VideoProcessingModuleTest::SetUp() {
-  vpm_ = VideoProcessingModule::Create();
-  ASSERT_TRUE(vpm_ != NULL);
+void VideoProcessingTest::SetUp() {
+  vp_ = VideoProcessing::Create();
+  ASSERT_TRUE(vp_ != NULL);
 
   ASSERT_EQ(0, video_frame_.CreateEmptyFrame(width_, height_, width_,
-                                            half_width_, half_width_));
+                                             half_width_, half_width_));
   // Clear video frame so DrMemory/Valgrind will allow reads of the buffer.
   memset(video_frame_.buffer(kYPlane), 0, video_frame_.allocated_size(kYPlane));
   memset(video_frame_.buffer(kUPlane), 0, video_frame_.allocated_size(kUPlane));
   memset(video_frame_.buffer(kVPlane), 0, video_frame_.allocated_size(kVPlane));
   const std::string video_file =
       webrtc::test::ResourcePath("foreman_cif", "yuv");
-  source_file_  = fopen(video_file.c_str(),"rb");
-  ASSERT_TRUE(source_file_ != NULL) <<
-      "Cannot read source file: " + video_file + "\n";
+  source_file_ = fopen(video_file.c_str(), "rb");
+  ASSERT_TRUE(source_file_ != NULL)
+      << "Cannot read source file: " + video_file + "\n";
 }
 
-void VideoProcessingModuleTest::TearDown() {
-  if (source_file_ != NULL)  {
+void VideoProcessingTest::TearDown() {
+  if (source_file_ != NULL) {
     ASSERT_EQ(0, fclose(source_file_));
   }
   source_file_ = NULL;
-
-  if (vpm_ != NULL)  {
-    VideoProcessingModule::Destroy(vpm_);
-  }
-  vpm_ = NULL;
+  delete vp_;
+  vp_ = NULL;
 }
 
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(HandleNullBuffer)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_HandleNullBuffer) {
+#else
+TEST_F(VideoProcessingTest, HandleNullBuffer) {
+#endif
   // TODO(mikhal/stefan): Do we need this one?
-  VideoProcessingModule::FrameStats stats;
+  VideoProcessing::FrameStats stats;
   // Video frame with unallocated buffer.
   VideoFrame videoFrame;
 
-  EXPECT_EQ(-3, vpm_->GetFrameStats(&stats, videoFrame));
+  vp_->GetFrameStats(videoFrame, &stats);
+  EXPECT_EQ(stats.num_pixels, 0u);
 
-  EXPECT_EQ(-1, vpm_->Deflickering(&videoFrame, &stats));
+  EXPECT_EQ(-1, vp_->Deflickering(&videoFrame, &stats));
 
-  EXPECT_EQ(-3, vpm_->BrightnessDetection(videoFrame, stats));
+  EXPECT_EQ(-3, vp_->BrightnessDetection(videoFrame, stats));
 }
 
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(HandleBadStats)) {
-  VideoProcessingModule::FrameStats stats;
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_HandleBadStats) {
+#else
+TEST_F(VideoProcessingTest, HandleBadStats) {
+#endif
+  VideoProcessing::FrameStats stats;
+  vp_->ClearFrameStats(&stats);
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
 
-  EXPECT_EQ(-1, vpm_->Deflickering(&video_frame_, &stats));
+  EXPECT_EQ(-1, vp_->Deflickering(&video_frame_, &stats));
 
-  EXPECT_EQ(-3, vpm_->BrightnessDetection(video_frame_, stats));
+  EXPECT_EQ(-3, vp_->BrightnessDetection(video_frame_, stats));
 }
 
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(IdenticalResultsAfterReset)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_IdenticalResultsAfterReset) {
+#else
+TEST_F(VideoProcessingTest, IdenticalResultsAfterReset) {
+#endif
   VideoFrame video_frame2;
-  VideoProcessingModule::FrameStats stats;
+  VideoProcessing::FrameStats stats;
   // Only testing non-static functions here.
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
-  ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
+  vp_->GetFrameStats(video_frame_, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
   ASSERT_EQ(0, video_frame2.CopyFrame(video_frame_));
-  ASSERT_EQ(0, vpm_->Deflickering(&video_frame_, &stats));
-  vpm_->Reset();
+  ASSERT_EQ(0, vp_->Deflickering(&video_frame_, &stats));
+
   // Retrieve frame stats again in case Deflickering() has zeroed them.
-  ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame2));
-  ASSERT_EQ(0, vpm_->Deflickering(&video_frame2, &stats));
+  vp_->GetFrameStats(video_frame2, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
+  ASSERT_EQ(0, vp_->Deflickering(&video_frame2, &stats));
   EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
 
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
-  ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
+  vp_->GetFrameStats(video_frame_, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
   video_frame2.CopyFrame(video_frame_);
-  ASSERT_EQ(0, vpm_->BrightnessDetection(video_frame_, stats));
-  vpm_->Reset();
-  ASSERT_EQ(0, vpm_->BrightnessDetection(video_frame2, stats));
+  ASSERT_EQ(0, vp_->BrightnessDetection(video_frame_, stats));
+
+  ASSERT_EQ(0, vp_->BrightnessDetection(video_frame2, stats));
   EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
 }
 
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(FrameStats)) {
-  VideoProcessingModule::FrameStats stats;
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_FrameStats) {
+#else
+TEST_F(VideoProcessingTest, FrameStats) {
+#endif
+  VideoProcessing::FrameStats stats;
+  vp_->ClearFrameStats(&stats);
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
 
-  EXPECT_FALSE(vpm_->ValidFrameStats(stats));
-  EXPECT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-  EXPECT_TRUE(vpm_->ValidFrameStats(stats));
+  EXPECT_FALSE(vp_->ValidFrameStats(stats));
+  vp_->GetFrameStats(video_frame_, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
+  EXPECT_TRUE(vp_->ValidFrameStats(stats));
 
   printf("\nFrameStats\n");
-  printf("mean: %u\nnum_pixels: %u\nsubSamplWidth: "
-         "%u\nsumSamplHeight: %u\nsum: %u\n\n",
+  printf("mean: %u\nnum_pixels: %u\nsubSamplFactor: %u\nsum: %u\n\n",
          static_cast<unsigned int>(stats.mean),
          static_cast<unsigned int>(stats.num_pixels),
-         static_cast<unsigned int>(stats.subSamplHeight),
-         static_cast<unsigned int>(stats.subSamplWidth),
+         static_cast<unsigned int>(stats.sub_sampling_factor),
          static_cast<unsigned int>(stats.sum));
 
-  vpm_->ClearFrameStats(&stats);
-  EXPECT_FALSE(vpm_->ValidFrameStats(stats));
+  vp_->ClearFrameStats(&stats);
+  EXPECT_FALSE(vp_->ValidFrameStats(stats));
 }
 
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(PreprocessorLogic)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_PreprocessorLogic) {
+#else
+TEST_F(VideoProcessingTest, PreprocessorLogic) {
+#endif
   // Disable temporal sampling (frame dropping).
-  vpm_->EnableTemporalDecimation(false);
+  vp_->EnableTemporalDecimation(false);
   int resolution = 100;
-  EXPECT_EQ(VPM_OK, vpm_->SetTargetResolution(resolution, resolution, 15));
-  EXPECT_EQ(VPM_OK, vpm_->SetTargetResolution(resolution, resolution, 30));
+  EXPECT_EQ(VPM_OK, vp_->SetTargetResolution(resolution, resolution, 15));
+  EXPECT_EQ(VPM_OK, vp_->SetTargetResolution(resolution, resolution, 30));
   // Disable spatial sampling.
-  vpm_->SetInputFrameResampleMode(kNoRescaling);
-  EXPECT_EQ(VPM_OK, vpm_->SetTargetResolution(resolution, resolution, 30));
+  vp_->SetInputFrameResampleMode(kNoRescaling);
+  EXPECT_EQ(VPM_OK, vp_->SetTargetResolution(resolution, resolution, 30));
   VideoFrame* out_frame = NULL;
   // Set rescaling => output frame != NULL.
-  vpm_->SetInputFrameResampleMode(kFastRescaling);
-  PreprocessFrameAndVerify(video_frame_, resolution, resolution, vpm_,
-                           &out_frame);
+  vp_->SetInputFrameResampleMode(kFastRescaling);
+  PreprocessFrameAndVerify(video_frame_, resolution, resolution, vp_,
+                           out_frame);
   // No rescaling=> output frame = NULL.
-  vpm_->SetInputFrameResampleMode(kNoRescaling);
-  EXPECT_EQ(VPM_OK, vpm_->PreprocessFrame(video_frame_, &out_frame));
-  EXPECT_TRUE(out_frame == NULL);
+  vp_->SetInputFrameResampleMode(kNoRescaling);
+  EXPECT_TRUE(vp_->PreprocessFrame(video_frame_) != nullptr);
 }
 
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(Resampler)) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_Resampler) {
+#else
+TEST_F(VideoProcessingTest, Resampler) {
+#endif
   enum { NumRuns = 1 };
 
   int64_t min_runtime = 0;
   int64_t total_runtime = 0;
 
   rewind(source_file_);
-  ASSERT_TRUE(source_file_ != NULL) <<
-      "Cannot read input file \n";
+  ASSERT_TRUE(source_file_ != NULL) << "Cannot read input file \n";
 
   // CA not needed here
-  vpm_->EnableContentAnalysis(false);
+  vp_->EnableContentAnalysis(false);
   // no temporal decimation
-  vpm_->EnableTemporalDecimation(false);
+  vp_->EnableTemporalDecimation(false);
 
   // Reading test frame
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   // Using ConvertToI420 to add stride to the image.
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
@@ -231,43 +255,43 @@ TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(Resampler)) {
 
     // Test scaling to different sizes: source is of |width|/|height| = 352/288.
     // Pure scaling:
-    TestSize(video_frame_, video_frame_, width_ / 4, height_ / 4, 25.2, vpm_);
-    TestSize(video_frame_, video_frame_, width_ / 2, height_ / 2, 28.1, vpm_);
+    TestSize(video_frame_, video_frame_, width_ / 4, height_ / 4, 25.2, vp_);
+    TestSize(video_frame_, video_frame_, width_ / 2, height_ / 2, 28.1, vp_);
     // No resampling:
-    TestSize(video_frame_, video_frame_, width_, height_, -1, vpm_);
-    TestSize(video_frame_, video_frame_, 2 * width_, 2 * height_, 32.2, vpm_);
+    TestSize(video_frame_, video_frame_, width_, height_, -1, vp_);
+    TestSize(video_frame_, video_frame_, 2 * width_, 2 * height_, 32.2, vp_);
 
     // Scaling and cropping. The cropped source frame is the largest center
     // aligned region that can be used from the source while preserving aspect
     // ratio.
     CropFrame(video_buffer.get(), width_, height_, 0, 56, 352, 176,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 100, 50, 24.0, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 100, 50, 24.0, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 0, 30, 352, 225,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 400, 256, 31.3, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 400, 256, 31.3, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 68, 0, 216, 288,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 480, 640, 32.15, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 480, 640, 32.15, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 0, 12, 352, 264,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 960, 720, 32.2, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 960, 720, 32.2, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 0, 44, 352, 198,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 1280, 720, 32.15, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 1280, 720, 32.15, vp_);
 
     // Upsampling to odd size.
     CropFrame(video_buffer.get(), width_, height_, 0, 26, 352, 233,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 501, 333, 32.05, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 501, 333, 32.05, vp_);
     // Downsample to odd size.
     CropFrame(video_buffer.get(), width_, height_, 0, 34, 352, 219,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 281, 175, 29.3, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 281, 175, 29.3, vp_);
 
     // Stop timer.
     const int64_t runtime = (TickTime::Now() - time_start).Microseconds();
@@ -279,30 +303,30 @@ TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(Resampler)) {
 
   printf("\nAverage run time = %d us / frame\n",
          static_cast<int>(total_runtime));
-  printf("Min run time = %d us / frame\n\n",
-         static_cast<int>(min_runtime));
+  printf("Min run time = %d us / frame\n\n", static_cast<int>(min_runtime));
 }
 
 void PreprocessFrameAndVerify(const VideoFrame& source,
                               int target_width,
                               int target_height,
-                              VideoProcessingModule* vpm,
-                              VideoFrame** out_frame) {
+                              VideoProcessing* vpm,
+                              const VideoFrame* out_frame) {
   ASSERT_EQ(VPM_OK, vpm->SetTargetResolution(target_width, target_height, 30));
-  ASSERT_EQ(VPM_OK, vpm->PreprocessFrame(source, out_frame));
+  out_frame = vpm->PreprocessFrame(source);
+  EXPECT_TRUE(out_frame != nullptr);
 
-  // If no resizing is needed, expect NULL.
+  // If no resizing is needed, expect the original frame.
   if (target_width == source.width() && target_height == source.height()) {
-    EXPECT_EQ(NULL, *out_frame);
+    EXPECT_EQ(&source, out_frame);
     return;
   }
 
   // Verify the resampled frame.
-  EXPECT_TRUE(*out_frame != NULL);
-  EXPECT_EQ(source.render_time_ms(), (*out_frame)->render_time_ms());
-  EXPECT_EQ(source.timestamp(), (*out_frame)->timestamp());
-  EXPECT_EQ(target_width, (*out_frame)->width());
-  EXPECT_EQ(target_height, (*out_frame)->height());
+  EXPECT_TRUE(out_frame != NULL);
+  EXPECT_EQ(source.render_time_ms(), (out_frame)->render_time_ms());
+  EXPECT_EQ(source.timestamp(), (out_frame)->timestamp());
+  EXPECT_EQ(target_width, (out_frame)->width());
+  EXPECT_EQ(target_height, (out_frame)->height());
 }
 
 void CropFrame(const uint8_t* source_data,
@@ -326,12 +350,12 @@ void TestSize(const VideoFrame& source_frame,
               int target_width,
               int target_height,
               double expected_psnr,
-              VideoProcessingModule* vpm) {
+              VideoProcessing* vpm) {
   // Resample source_frame to out_frame.
   VideoFrame* out_frame = NULL;
   vpm->SetInputFrameResampleMode(kBox);
   PreprocessFrameAndVerify(source_frame, target_width, target_height, vpm,
-                           &out_frame);
+                           out_frame);
   if (out_frame == NULL)
     return;
   WriteProcessedFrameForVisualInspection(source_frame, *out_frame);
@@ -340,21 +364,22 @@ void TestSize(const VideoFrame& source_frame,
   VideoFrame resampled_source_frame;
   resampled_source_frame.CopyFrame(*out_frame);
   PreprocessFrameAndVerify(resampled_source_frame, cropped_source_frame.width(),
-                           cropped_source_frame.height(), vpm, &out_frame);
+                           cropped_source_frame.height(), vpm, out_frame);
   WriteProcessedFrameForVisualInspection(resampled_source_frame, *out_frame);
 
   // Compute PSNR against the cropped source frame and check expectation.
   double psnr = I420PSNR(&cropped_source_frame, out_frame);
   EXPECT_GT(psnr, expected_psnr);
-  printf("PSNR: %f. PSNR is between source of size %d %d, and a modified "
-         "source which is scaled down/up to: %d %d, and back to source size \n",
-         psnr, source_frame.width(), source_frame.height(),
-         target_width, target_height);
+  printf(
+      "PSNR: %f. PSNR is between source of size %d %d, and a modified "
+      "source which is scaled down/up to: %d %d, and back to source size \n",
+      psnr, source_frame.width(), source_frame.height(), target_width,
+      target_height);
 }
 
 bool CompareFrames(const webrtc::VideoFrame& frame1,
                    const webrtc::VideoFrame& frame2) {
-  for (int plane = 0; plane < webrtc::kNumOfPlanes; plane ++) {
+  for (int plane = 0; plane < webrtc::kNumOfPlanes; plane++) {
     webrtc::PlaneType plane_type = static_cast<webrtc::PlaneType>(plane);
     int allocated_size1 = frame1.allocated_size(plane_type);
     int allocated_size2 = frame2.allocated_size(plane_type);
diff --git a/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h b/webrtc/modules/video_processing/test/video_processing_unittest.h
index 4a4fda41e6..3433c6ca86 100644
--- a/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h
+++ b/webrtc/modules/video_processing/test/video_processing_unittest.h
@@ -8,19 +8,21 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VIDEO_PROCESSING_UNITTEST_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VIDEO_PROCESSING_UNITTEST_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_TEST_VIDEO_PROCESSING_UNITTEST_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_TEST_VIDEO_PROCESSING_UNITTEST_H_
+
+#include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
-class VideoProcessingModuleTest : public ::testing::Test {
+class VideoProcessingTest : public ::testing::Test {
  protected:
-  VideoProcessingModuleTest();
+  VideoProcessingTest();
   virtual void SetUp();
   virtual void TearDown();
   static void SetUpTestCase() {
@@ -28,10 +30,8 @@ class VideoProcessingModuleTest : public ::testing::Test {
     std::string trace_file = webrtc::test::OutputPath() + "VPMTrace.txt";
     ASSERT_EQ(0, Trace::SetTraceFile(trace_file.c_str()));
   }
-  static void TearDownTestCase() {
-    Trace::ReturnTrace();
-  }
-  VideoProcessingModule* vpm_;
+  static void TearDownTestCase() { Trace::ReturnTrace(); }
+  VideoProcessing* vp_;
   FILE* source_file_;
   VideoFrame video_frame_;
   const int width_;
@@ -44,4 +44,4 @@ class VideoProcessingModuleTest : public ::testing::Test {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VIDEO_PROCESSING_UNITTEST_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_TEST_VIDEO_PROCESSING_UNITTEST_H_
diff --git a/webrtc/modules/video_processing/main/test/unit_test/writeYUV420file.m b/webrtc/modules/video_processing/test/writeYUV420file.m
index 69a8808338..359445009b 100644
--- a/webrtc/modules/video_processing/main/test/unit_test/writeYUV420file.m
+++ b/webrtc/modules/video_processing/test/writeYUV420file.m
@@ -11,10 +11,10 @@ numFrames=size(Y,3);
 for k=1:numFrames
    % Write luminance
    fwrite(fid,uint8(Y(:,:,k).'), 'uchar');
-   
+
    % Write U channel
    fwrite(fid,uint8(U(:,:,k).'), 'uchar');
-   
+
    % Write V channel
    fwrite(fid,uint8(V(:,:,k).'), 'uchar');
 end
diff --git a/webrtc/modules/video_processing/util/denoiser_filter.cc b/webrtc/modules/video_processing/util/denoiser_filter.cc
new file mode 100644
index 0000000000..fbc2435cb5
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter_c.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter_neon.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter_sse2.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+
+namespace webrtc {
+
+const int kMotionMagnitudeThreshold = 8 * 3;
+const int kSumDiffThreshold = 16 * 16 * 2;
+const int kSumDiffThresholdHigh = 600;
+
+rtc::scoped_ptr<DenoiserFilter> DenoiserFilter::Create(
+    bool runtime_cpu_detection) {
+  rtc::scoped_ptr<DenoiserFilter> filter;
+
+  if (runtime_cpu_detection) {
+// If we know the minimum architecture at compile time, avoid CPU detection.
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    // x86 CPU detection required.
+    if (WebRtc_GetCPUInfo(kSSE2)) {
+      filter.reset(new DenoiserFilterSSE2());
+    } else {
+      filter.reset(new DenoiserFilterC());
+    }
+#elif defined(WEBRTC_DETECT_NEON)
+    if (WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) {
+      filter.reset(new DenoiserFilterNEON());
+    } else {
+      filter.reset(new DenoiserFilterC());
+    }
+#else
+    filter.reset(new DenoiserFilterC());
+#endif
+  } else {
+    filter.reset(new DenoiserFilterC());
+  }
+
+  RTC_DCHECK(filter.get() != nullptr);
+  return filter;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/util/denoiser_filter.h b/webrtc/modules/video_processing/util/denoiser_filter.h
new file mode 100644
index 0000000000..5d5a61c59c
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
+
+namespace webrtc {
+
+extern const int kMotionMagnitudeThreshold;
+extern const int kSumDiffThreshold;
+extern const int kSumDiffThresholdHigh;
+
+enum DenoiserDecision { COPY_BLOCK, FILTER_BLOCK };
+struct DenoiseMetrics {
+  uint32_t var;
+  uint32_t sad;
+  uint8_t denoise;
+  bool is_skin;
+};
+
+class DenoiserFilter {
+ public:
+  static rtc::scoped_ptr<DenoiserFilter> Create(bool runtime_cpu_detection);
+
+  virtual ~DenoiserFilter() {}
+
+  virtual void CopyMem16x16(const uint8_t* src,
+                            int src_stride,
+                            uint8_t* dst,
+                            int dst_stride) = 0;
+  virtual void CopyMem8x8(const uint8_t* src,
+                          int src_stride,
+                          uint8_t* dst,
+                          int dst_stride) = 0;
+  virtual uint32_t Variance16x8(const uint8_t* a,
+                                int a_stride,
+                                const uint8_t* b,
+                                int b_stride,
+                                unsigned int* sse) = 0;
+  virtual DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                                     int mc_avg_y_stride,
+                                     uint8_t* running_avg_y,
+                                     int avg_y_stride,
+                                     const uint8_t* sig,
+                                     int sig_stride,
+                                     uint8_t motion_magnitude,
+                                     int increase_denoising) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_H_
diff --git a/webrtc/modules/video_processing/util/denoiser_filter_c.cc b/webrtc/modules/video_processing/util/denoiser_filter_c.cc
new file mode 100644
index 0000000000..6323980e18
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter_c.cc
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+
+#include "webrtc/modules/video_processing/util/denoiser_filter_c.h"
+
+namespace webrtc {
+
+void DenoiserFilterC::CopyMem16x16(const uint8_t* src,
+                                   int src_stride,
+                                   uint8_t* dst,
+                                   int dst_stride) {
+  for (int i = 0; i < 16; i++) {
+    memcpy(dst, src, 16);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+void DenoiserFilterC::CopyMem8x8(const uint8_t* src,
+                                 int src_stride,
+                                 uint8_t* dst,
+                                 int dst_stride) {
+  for (int i = 0; i < 8; i++) {
+    memcpy(dst, src, 8);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+uint32_t DenoiserFilterC::Variance16x8(const uint8_t* a,
+                                       int a_stride,
+                                       const uint8_t* b,
+                                       int b_stride,
+                                       uint32_t* sse) {
+  int sum = 0;
+  *sse = 0;
+  a_stride <<= 1;
+  b_stride <<= 1;
+
+  for (int i = 0; i < 8; i++) {
+    for (int j = 0; j < 16; j++) {
+      const int diff = a[j] - b[j];
+      sum += diff;
+      *sse += diff * diff;
+    }
+
+    a += a_stride;
+    b += b_stride;
+  }
+  return *sse - ((static_cast<int64_t>(sum) * sum) >> 7);
+}
+
+DenoiserDecision DenoiserFilterC::MbDenoise(uint8_t* mc_running_avg_y,
+                                            int mc_avg_y_stride,
+                                            uint8_t* running_avg_y,
+                                            int avg_y_stride,
+                                            const uint8_t* sig,
+                                            int sig_stride,
+                                            uint8_t motion_magnitude,
+                                            int increase_denoising) {
+  int sum_diff_thresh = 0;
+  int sum_diff = 0;
+  int adj_val[3] = {3, 4, 6};
+  int shift_inc1 = 0;
+  int shift_inc2 = 1;
+  int col_sum[16] = {0};
+  if (motion_magnitude <= kMotionMagnitudeThreshold) {
+    if (increase_denoising) {
+      shift_inc1 = 1;
+      shift_inc2 = 2;
+    }
+    adj_val[0] += shift_inc2;
+    adj_val[1] += shift_inc2;
+    adj_val[2] += shift_inc2;
+  }
+
+  for (int r = 0; r < 16; ++r) {
+    for (int c = 0; c < 16; ++c) {
+      int diff = 0;
+      int adjustment = 0;
+      int absdiff = 0;
+
+      diff = mc_running_avg_y[c] - sig[c];
+      absdiff = abs(diff);
+
+      // When |diff| <= |3 + shift_inc1|, use pixel value from
+      // last denoised raw.
+      if (absdiff <= 3 + shift_inc1) {
+        running_avg_y[c] = mc_running_avg_y[c];
+        col_sum[c] += diff;
+      } else {
+        if (absdiff >= 4 + shift_inc1 && absdiff <= 7)
+          adjustment = adj_val[0];
+        else if (absdiff >= 8 && absdiff <= 15)
+          adjustment = adj_val[1];
+        else
+          adjustment = adj_val[2];
+
+        if (diff > 0) {
+          if ((sig[c] + adjustment) > 255)
+            running_avg_y[c] = 255;
+          else
+            running_avg_y[c] = sig[c] + adjustment;
+
+          col_sum[c] += adjustment;
+        } else {
+          if ((sig[c] - adjustment) < 0)
+            running_avg_y[c] = 0;
+          else
+            running_avg_y[c] = sig[c] - adjustment;
+
+          col_sum[c] -= adjustment;
+        }
+      }
+    }
+
+    // Update pointers for next iteration.
+    sig += sig_stride;
+    mc_running_avg_y += mc_avg_y_stride;
+    running_avg_y += avg_y_stride;
+  }
+
+  for (int c = 0; c < 16; ++c) {
+    if (col_sum[c] >= 128) {
+      col_sum[c] = 127;
+    }
+    sum_diff += col_sum[c];
+  }
+
+  sum_diff_thresh = kSumDiffThreshold;
+  if (increase_denoising)
+    sum_diff_thresh = kSumDiffThresholdHigh;
+  if (abs(sum_diff) > sum_diff_thresh) {
+    int delta = ((abs(sum_diff) - sum_diff_thresh) >> 8) + 1;
+    // Only apply the adjustment for max delta up to 3.
+    if (delta < 4) {
+      sig -= sig_stride * 16;
+      mc_running_avg_y -= mc_avg_y_stride * 16;
+      running_avg_y -= avg_y_stride * 16;
+      for (int r = 0; r < 16; ++r) {
+        for (int c = 0; c < 16; ++c) {
+          int diff = mc_running_avg_y[c] - sig[c];
+          int adjustment = abs(diff);
+          if (adjustment > delta)
+            adjustment = delta;
+          if (diff > 0) {
+            // Bring denoised signal down.
+            if (running_avg_y[c] - adjustment < 0)
+              running_avg_y[c] = 0;
+            else
+              running_avg_y[c] = running_avg_y[c] - adjustment;
+            col_sum[c] -= adjustment;
+          } else if (diff < 0) {
+            // Bring denoised signal up.
+            if (running_avg_y[c] + adjustment > 255)
+              running_avg_y[c] = 255;
+            else
+              running_avg_y[c] = running_avg_y[c] + adjustment;
+            col_sum[c] += adjustment;
+          }
+        }
+        sig += sig_stride;
+        mc_running_avg_y += mc_avg_y_stride;
+        running_avg_y += avg_y_stride;
+      }
+
+      sum_diff = 0;
+      for (int c = 0; c < 16; ++c) {
+        if (col_sum[c] >= 128) {
+          col_sum[c] = 127;
+        }
+        sum_diff += col_sum[c];
+      }
+
+      if (abs(sum_diff) > sum_diff_thresh)
+        return COPY_BLOCK;
+    } else {
+      return COPY_BLOCK;
+    }
+  }
+
+  return FILTER_BLOCK;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/util/denoiser_filter_c.h b/webrtc/modules/video_processing/util/denoiser_filter_c.h
new file mode 100644
index 0000000000..fe46ac38ec
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter_c.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_C_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_C_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+
+namespace webrtc {
+
+class DenoiserFilterC : public DenoiserFilter {
+ public:
+  DenoiserFilterC() {}
+  void CopyMem16x16(const uint8_t* src,
+                    int src_stride,
+                    uint8_t* dst,
+                    int dst_stride) override;
+  void CopyMem8x8(const uint8_t* src,
+                  int src_stride,
+                  uint8_t* dst,
+                  int dst_stride) override;
+  uint32_t Variance16x8(const uint8_t* a,
+                        int a_stride,
+                        const uint8_t* b,
+                        int b_stride,
+                        unsigned int* sse) override;
+  DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                             int mc_avg_y_stride,
+                             uint8_t* running_avg_y,
+                             int avg_y_stride,
+                             const uint8_t* sig,
+                             int sig_stride,
+                             uint8_t motion_magnitude,
+                             int increase_denoising) override;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_C_H_
diff --git a/webrtc/modules/video_processing/util/denoiser_filter_neon.cc b/webrtc/modules/video_processing/util/denoiser_filter_neon.cc
new file mode 100644
index 0000000000..b522bf002b
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter_neon.cc
@@ -0,0 +1,283 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <arm_neon.h>
+
+#include "webrtc/modules/video_processing/util/denoiser_filter_neon.h"
+
+namespace webrtc {
+
+static int HorizontalAddS16x8(const int16x8_t v_16x8) {
+  const int32x4_t a = vpaddlq_s16(v_16x8);
+  const int64x2_t b = vpaddlq_s32(a);
+  const int32x2_t c = vadd_s32(vreinterpret_s32_s64(vget_low_s64(b)),
+                               vreinterpret_s32_s64(vget_high_s64(b)));
+  return vget_lane_s32(c, 0);
+}
+
+static int HorizontalAddS32x4(const int32x4_t v_32x4) {
+  const int64x2_t b = vpaddlq_s32(v_32x4);
+  const int32x2_t c = vadd_s32(vreinterpret_s32_s64(vget_low_s64(b)),
+                               vreinterpret_s32_s64(vget_high_s64(b)));
+  return vget_lane_s32(c, 0);
+}
+
+static void VarianceNeonW8(const uint8_t* a,
+                           int a_stride,
+                           const uint8_t* b,
+                           int b_stride,
+                           int w,
+                           int h,
+                           uint32_t* sse,
+                           int64_t* sum) {
+  int16x8_t v_sum = vdupq_n_s16(0);
+  int32x4_t v_sse_lo = vdupq_n_s32(0);
+  int32x4_t v_sse_hi = vdupq_n_s32(0);
+
+  for (int i = 0; i < h; ++i) {
+    for (int j = 0; j < w; j += 8) {
+      const uint8x8_t v_a = vld1_u8(&a[j]);
+      const uint8x8_t v_b = vld1_u8(&b[j]);
+      const uint16x8_t v_diff = vsubl_u8(v_a, v_b);
+      const int16x8_t sv_diff = vreinterpretq_s16_u16(v_diff);
+      v_sum = vaddq_s16(v_sum, sv_diff);
+      v_sse_lo =
+          vmlal_s16(v_sse_lo, vget_low_s16(sv_diff), vget_low_s16(sv_diff));
+      v_sse_hi =
+          vmlal_s16(v_sse_hi, vget_high_s16(sv_diff), vget_high_s16(sv_diff));
+    }
+    a += a_stride;
+    b += b_stride;
+  }
+
+  *sum = HorizontalAddS16x8(v_sum);
+  *sse =
+      static_cast<uint32_t>(HorizontalAddS32x4(vaddq_s32(v_sse_lo, v_sse_hi)));
+}
+
+void DenoiserFilterNEON::CopyMem16x16(const uint8_t* src,
+                                      int src_stride,
+                                      uint8_t* dst,
+                                      int dst_stride) {
+  uint8x16_t qtmp;
+  for (int r = 0; r < 16; r++) {
+    qtmp = vld1q_u8(src);
+    vst1q_u8(dst, qtmp);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+void DenoiserFilterNEON::CopyMem8x8(const uint8_t* src,
+                                    int src_stride,
+                                    uint8_t* dst,
+                                    int dst_stride) {
+  uint8x8_t vtmp;
+
+  for (int r = 0; r < 8; r++) {
+    vtmp = vld1_u8(src);
+    vst1_u8(dst, vtmp);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+uint32_t DenoiserFilterNEON::Variance16x8(const uint8_t* a,
+                                          int a_stride,
+                                          const uint8_t* b,
+                                          int b_stride,
+                                          uint32_t* sse) {
+  int64_t sum = 0;
+  VarianceNeonW8(a, a_stride << 1, b, b_stride << 1, 16, 8, sse, &sum);
+  return *sse - ((sum * sum) >> 7);
+}
+
+DenoiserDecision DenoiserFilterNEON::MbDenoise(uint8_t* mc_running_avg_y,
+                                               int mc_running_avg_y_stride,
+                                               uint8_t* running_avg_y,
+                                               int running_avg_y_stride,
+                                               const uint8_t* sig,
+                                               int sig_stride,
+                                               uint8_t motion_magnitude,
+                                               int increase_denoising) {
+  // If motion_magnitude is small, making the denoiser more aggressive by
+  // increasing the adjustment for each level, level1 adjustment is
+  // increased, the deltas stay the same.
+  int shift_inc =
+      (increase_denoising && motion_magnitude <= kMotionMagnitudeThreshold) ? 1
+                                                                            : 0;
+  const uint8x16_t v_level1_adjustment = vmovq_n_u8(
+      (motion_magnitude <= kMotionMagnitudeThreshold) ? 4 + shift_inc : 3);
+  const uint8x16_t v_delta_level_1_and_2 = vdupq_n_u8(1);
+  const uint8x16_t v_delta_level_2_and_3 = vdupq_n_u8(2);
+  const uint8x16_t v_level1_threshold = vmovq_n_u8(4 + shift_inc);
+  const uint8x16_t v_level2_threshold = vdupq_n_u8(8);
+  const uint8x16_t v_level3_threshold = vdupq_n_u8(16);
+  int64x2_t v_sum_diff_total = vdupq_n_s64(0);
+
+  // Go over lines.
+  for (int r = 0; r < 16; ++r) {
+    // Load inputs.
+    const uint8x16_t v_sig = vld1q_u8(sig);
+    const uint8x16_t v_mc_running_avg_y = vld1q_u8(mc_running_avg_y);
+
+    // Calculate absolute difference and sign masks.
+    const uint8x16_t v_abs_diff = vabdq_u8(v_sig, v_mc_running_avg_y);
+    const uint8x16_t v_diff_pos_mask = vcltq_u8(v_sig, v_mc_running_avg_y);
+    const uint8x16_t v_diff_neg_mask = vcgtq_u8(v_sig, v_mc_running_avg_y);
+
+    // Figure out which level that put us in.
+    const uint8x16_t v_level1_mask = vcleq_u8(v_level1_threshold, v_abs_diff);
+    const uint8x16_t v_level2_mask = vcleq_u8(v_level2_threshold, v_abs_diff);
+    const uint8x16_t v_level3_mask = vcleq_u8(v_level3_threshold, v_abs_diff);
+
+    // Calculate absolute adjustments for level 1, 2 and 3.
+    const uint8x16_t v_level2_adjustment =
+        vandq_u8(v_level2_mask, v_delta_level_1_and_2);
+    const uint8x16_t v_level3_adjustment =
+        vandq_u8(v_level3_mask, v_delta_level_2_and_3);
+    const uint8x16_t v_level1and2_adjustment =
+        vaddq_u8(v_level1_adjustment, v_level2_adjustment);
+    const uint8x16_t v_level1and2and3_adjustment =
+        vaddq_u8(v_level1and2_adjustment, v_level3_adjustment);
+
+    // Figure adjustment absolute value by selecting between the absolute
+    // difference if in level0 or the value for level 1, 2 and 3.
+    const uint8x16_t v_abs_adjustment =
+        vbslq_u8(v_level1_mask, v_level1and2and3_adjustment, v_abs_diff);
+
+    // Calculate positive and negative adjustments. Apply them to the signal
+    // and accumulate them. Adjustments are less than eight and the maximum
+    // sum of them (7 * 16) can fit in a signed char.
+    const uint8x16_t v_pos_adjustment =
+        vandq_u8(v_diff_pos_mask, v_abs_adjustment);
+    const uint8x16_t v_neg_adjustment =
+        vandq_u8(v_diff_neg_mask, v_abs_adjustment);
+
+    uint8x16_t v_running_avg_y = vqaddq_u8(v_sig, v_pos_adjustment);
+    v_running_avg_y = vqsubq_u8(v_running_avg_y, v_neg_adjustment);
+
+    // Store results.
+    vst1q_u8(running_avg_y, v_running_avg_y);
+
+    // Sum all the accumulators to have the sum of all pixel differences
+    // for this macroblock.
+    {
+      const int8x16_t v_sum_diff =
+          vqsubq_s8(vreinterpretq_s8_u8(v_pos_adjustment),
+                    vreinterpretq_s8_u8(v_neg_adjustment));
+      const int16x8_t fe_dc_ba_98_76_54_32_10 = vpaddlq_s8(v_sum_diff);
+      const int32x4_t fedc_ba98_7654_3210 =
+          vpaddlq_s16(fe_dc_ba_98_76_54_32_10);
+      const int64x2_t fedcba98_76543210 = vpaddlq_s32(fedc_ba98_7654_3210);
+
+      v_sum_diff_total = vqaddq_s64(v_sum_diff_total, fedcba98_76543210);
+    }
+
+    // Update pointers for next iteration.
+    sig += sig_stride;
+    mc_running_avg_y += mc_running_avg_y_stride;
+    running_avg_y += running_avg_y_stride;
+  }
+
+  // Too much adjustments => copy block.
+  {
+    int64x1_t x = vqadd_s64(vget_high_s64(v_sum_diff_total),
+                            vget_low_s64(v_sum_diff_total));
+    int sum_diff = vget_lane_s32(vabs_s32(vreinterpret_s32_s64(x)), 0);
+    int sum_diff_thresh = kSumDiffThreshold;
+
+    if (increase_denoising)
+      sum_diff_thresh = kSumDiffThresholdHigh;
+    if (sum_diff > sum_diff_thresh) {
+      // Before returning to copy the block (i.e., apply no denoising),
+      // checK if we can still apply some (weaker) temporal filtering to
+      // this block, that would otherwise not be denoised at all. Simplest
+      // is to apply an additional adjustment to running_avg_y to bring it
+      // closer to sig. The adjustment is capped by a maximum delta, and
+      // chosen such that in most cases the resulting sum_diff will be
+      // within the accceptable range given by sum_diff_thresh.
+
+      // The delta is set by the excess of absolute pixel diff over the
+      // threshold.
+      int delta = ((sum_diff - sum_diff_thresh) >> 8) + 1;
+      // Only apply the adjustment for max delta up to 3.
+      if (delta < 4) {
+        const uint8x16_t k_delta = vmovq_n_u8(delta);
+        sig -= sig_stride * 16;
+        mc_running_avg_y -= mc_running_avg_y_stride * 16;
+        running_avg_y -= running_avg_y_stride * 16;
+        for (int r = 0; r < 16; ++r) {
+          uint8x16_t v_running_avg_y = vld1q_u8(running_avg_y);
+          const uint8x16_t v_sig = vld1q_u8(sig);
+          const uint8x16_t v_mc_running_avg_y = vld1q_u8(mc_running_avg_y);
+
+          // Calculate absolute difference and sign masks.
+          const uint8x16_t v_abs_diff = vabdq_u8(v_sig, v_mc_running_avg_y);
+          const uint8x16_t v_diff_pos_mask =
+              vcltq_u8(v_sig, v_mc_running_avg_y);
+          const uint8x16_t v_diff_neg_mask =
+              vcgtq_u8(v_sig, v_mc_running_avg_y);
+          // Clamp absolute difference to delta to get the adjustment.
+          const uint8x16_t v_abs_adjustment = vminq_u8(v_abs_diff, (k_delta));
+
+          const uint8x16_t v_pos_adjustment =
+              vandq_u8(v_diff_pos_mask, v_abs_adjustment);
+          const uint8x16_t v_neg_adjustment =
+              vandq_u8(v_diff_neg_mask, v_abs_adjustment);
+
+          v_running_avg_y = vqsubq_u8(v_running_avg_y, v_pos_adjustment);
+          v_running_avg_y = vqaddq_u8(v_running_avg_y, v_neg_adjustment);
+
+          // Store results.
+          vst1q_u8(running_avg_y, v_running_avg_y);
+
+          {
+            const int8x16_t v_sum_diff =
+                vqsubq_s8(vreinterpretq_s8_u8(v_neg_adjustment),
+                          vreinterpretq_s8_u8(v_pos_adjustment));
+
+            const int16x8_t fe_dc_ba_98_76_54_32_10 = vpaddlq_s8(v_sum_diff);
+            const int32x4_t fedc_ba98_7654_3210 =
+                vpaddlq_s16(fe_dc_ba_98_76_54_32_10);
+            const int64x2_t fedcba98_76543210 =
+                vpaddlq_s32(fedc_ba98_7654_3210);
+
+            v_sum_diff_total = vqaddq_s64(v_sum_diff_total, fedcba98_76543210);
+          }
+          // Update pointers for next iteration.
+          sig += sig_stride;
+          mc_running_avg_y += mc_running_avg_y_stride;
+          running_avg_y += running_avg_y_stride;
+        }
+        {
+          // Update the sum of all pixel differences of this MB.
+          x = vqadd_s64(vget_high_s64(v_sum_diff_total),
+                        vget_low_s64(v_sum_diff_total));
+          sum_diff = vget_lane_s32(vabs_s32(vreinterpret_s32_s64(x)), 0);
+
+          if (sum_diff > sum_diff_thresh) {
+            return COPY_BLOCK;
+          }
+        }
+      } else {
+        return COPY_BLOCK;
+      }
+    }
+  }
+
+  // Tell above level that block was filtered.
+  running_avg_y -= running_avg_y_stride * 16;
+  sig -= sig_stride * 16;
+
+  return FILTER_BLOCK;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/util/denoiser_filter_neon.h b/webrtc/modules/video_processing/util/denoiser_filter_neon.h
new file mode 100644
index 0000000000..bc87ba788e
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter_neon.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_NEON_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_NEON_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+
+namespace webrtc {
+
+class DenoiserFilterNEON : public DenoiserFilter {
+ public:
+  DenoiserFilterNEON() {}
+  void CopyMem16x16(const uint8_t* src,
+                    int src_stride,
+                    uint8_t* dst,
+                    int dst_stride) override;
+  void CopyMem8x8(const uint8_t* src,
+                  int src_stride,
+                  uint8_t* dst,
+                  int dst_stride) override;
+  uint32_t Variance16x8(const uint8_t* a,
+                        int a_stride,
+                        const uint8_t* b,
+                        int b_stride,
+                        unsigned int* sse) override;
+  DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                             int mc_avg_y_stride,
+                             uint8_t* running_avg_y,
+                             int avg_y_stride,
+                             const uint8_t* sig,
+                             int sig_stride,
+                             uint8_t motion_magnitude,
+                             int increase_denoising) override;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_NEON_H_
diff --git a/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc b/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
new file mode 100644
index 0000000000..903d7b1ec6
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
@@ -0,0 +1,280 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <emmintrin.h>
+
+#include "webrtc/modules/video_processing/util/denoiser_filter_sse2.h"
+
+namespace webrtc {
+
+static void Get8x8varSse2(const uint8_t* src,
+                          int src_stride,
+                          const uint8_t* ref,
+                          int ref_stride,
+                          unsigned int* sse,
+                          int* sum) {
+  const __m128i zero = _mm_setzero_si128();
+  __m128i vsum = _mm_setzero_si128();
+  __m128i vsse = _mm_setzero_si128();
+
+  for (int i = 0; i < 8; i += 2) {
+    const __m128i src0 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(src + i * src_stride)), zero);
+    const __m128i ref0 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(ref + i * ref_stride)), zero);
+    const __m128i diff0 = _mm_sub_epi16(src0, ref0);
+
+    const __m128i src1 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(src + (i + 1) * src_stride)), zero);
+    const __m128i ref1 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(ref + (i + 1) * ref_stride)), zero);
+    const __m128i diff1 = _mm_sub_epi16(src1, ref1);
+
+    vsum = _mm_add_epi16(vsum, diff0);
+    vsum = _mm_add_epi16(vsum, diff1);
+    vsse = _mm_add_epi32(vsse, _mm_madd_epi16(diff0, diff0));
+    vsse = _mm_add_epi32(vsse, _mm_madd_epi16(diff1, diff1));
+  }
+
+  // sum
+  vsum = _mm_add_epi16(vsum, _mm_srli_si128(vsum, 8));
+  vsum = _mm_add_epi16(vsum, _mm_srli_si128(vsum, 4));
+  vsum = _mm_add_epi16(vsum, _mm_srli_si128(vsum, 2));
+  *sum = static_cast<int16_t>(_mm_extract_epi16(vsum, 0));
+
+  // sse
+  vsse = _mm_add_epi32(vsse, _mm_srli_si128(vsse, 8));
+  vsse = _mm_add_epi32(vsse, _mm_srli_si128(vsse, 4));
+  *sse = _mm_cvtsi128_si32(vsse);
+}
+
+static void VarianceSSE2(const unsigned char* src,
+                         int src_stride,
+                         const unsigned char* ref,
+                         int ref_stride,
+                         int w,
+                         int h,
+                         uint32_t* sse,
+                         int64_t* sum,
+                         int block_size) {
+  *sse = 0;
+  *sum = 0;
+
+  for (int i = 0; i < h; i += block_size) {
+    for (int j = 0; j < w; j += block_size) {
+      uint32_t sse0 = 0;
+      int32_t sum0 = 0;
+
+      Get8x8varSse2(src + src_stride * i + j, src_stride,
+                    ref + ref_stride * i + j, ref_stride, &sse0, &sum0);
+      *sse += sse0;
+      *sum += sum0;
+    }
+  }
+}
+
+// Compute the sum of all pixel differences of this MB.
+static uint32_t AbsSumDiff16x1(__m128i acc_diff) {
+  const __m128i k_1 = _mm_set1_epi16(1);
+  const __m128i acc_diff_lo =
+      _mm_srai_epi16(_mm_unpacklo_epi8(acc_diff, acc_diff), 8);
+  const __m128i acc_diff_hi =
+      _mm_srai_epi16(_mm_unpackhi_epi8(acc_diff, acc_diff), 8);
+  const __m128i acc_diff_16 = _mm_add_epi16(acc_diff_lo, acc_diff_hi);
+  const __m128i hg_fe_dc_ba = _mm_madd_epi16(acc_diff_16, k_1);
+  const __m128i hgfe_dcba =
+      _mm_add_epi32(hg_fe_dc_ba, _mm_srli_si128(hg_fe_dc_ba, 8));
+  const __m128i hgfedcba =
+      _mm_add_epi32(hgfe_dcba, _mm_srli_si128(hgfe_dcba, 4));
+  unsigned int sum_diff = abs(_mm_cvtsi128_si32(hgfedcba));
+
+  return sum_diff;
+}
+
+// TODO(jackychen): Optimize this function using SSE2.
+void DenoiserFilterSSE2::CopyMem16x16(const uint8_t* src,
+                                      int src_stride,
+                                      uint8_t* dst,
+                                      int dst_stride) {
+  for (int i = 0; i < 16; i++) {
+    memcpy(dst, src, 16);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+// TODO(jackychen): Optimize this function using SSE2.
+void DenoiserFilterSSE2::CopyMem8x8(const uint8_t* src,
+                                    int src_stride,
+                                    uint8_t* dst,
+                                    int dst_stride) {
+  for (int i = 0; i < 8; i++) {
+    memcpy(dst, src, 8);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+uint32_t DenoiserFilterSSE2::Variance16x8(const uint8_t* src,
+                                          int src_stride,
+                                          const uint8_t* ref,
+                                          int ref_stride,
+                                          uint32_t* sse) {
+  int64_t sum = 0;
+  VarianceSSE2(src, src_stride << 1, ref, ref_stride << 1, 16, 8, sse, &sum, 8);
+  return *sse - ((sum * sum) >> 7);
+}
+
+DenoiserDecision DenoiserFilterSSE2::MbDenoise(uint8_t* mc_running_avg_y,
+                                               int mc_avg_y_stride,
+                                               uint8_t* running_avg_y,
+                                               int avg_y_stride,
+                                               const uint8_t* sig,
+                                               int sig_stride,
+                                               uint8_t motion_magnitude,
+                                               int increase_denoising) {
+  int shift_inc =
+      (increase_denoising && motion_magnitude <= kMotionMagnitudeThreshold) ? 1
+                                                                            : 0;
+  __m128i acc_diff = _mm_setzero_si128();
+  const __m128i k_0 = _mm_setzero_si128();
+  const __m128i k_4 = _mm_set1_epi8(4 + shift_inc);
+  const __m128i k_8 = _mm_set1_epi8(8);
+  const __m128i k_16 = _mm_set1_epi8(16);
+  // Modify each level's adjustment according to motion_magnitude.
+  const __m128i l3 = _mm_set1_epi8(
+      (motion_magnitude <= kMotionMagnitudeThreshold) ? 7 + shift_inc : 6);
+  // Difference between level 3 and level 2 is 2.
+  const __m128i l32 = _mm_set1_epi8(2);
+  // Difference between level 2 and level 1 is 1.
+  const __m128i l21 = _mm_set1_epi8(1);
+
+  for (int r = 0; r < 16; ++r) {
+    // Calculate differences.
+    const __m128i v_sig =
+        _mm_loadu_si128(reinterpret_cast<const __m128i*>(&sig[0]));
+    const __m128i v_mc_running_avg_y =
+        _mm_loadu_si128(reinterpret_cast<__m128i*>(&mc_running_avg_y[0]));
+    __m128i v_running_avg_y;
+    const __m128i pdiff = _mm_subs_epu8(v_mc_running_avg_y, v_sig);
+    const __m128i ndiff = _mm_subs_epu8(v_sig, v_mc_running_avg_y);
+    // Obtain the sign. FF if diff is negative.
+    const __m128i diff_sign = _mm_cmpeq_epi8(pdiff, k_0);
+    // Clamp absolute difference to 16 to be used to get mask. Doing this
+    // allows us to use _mm_cmpgt_epi8, which operates on signed byte.
+    const __m128i clamped_absdiff =
+        _mm_min_epu8(_mm_or_si128(pdiff, ndiff), k_16);
+    // Get masks for l2 l1 and l0 adjustments.
+    const __m128i mask2 = _mm_cmpgt_epi8(k_16, clamped_absdiff);
+    const __m128i mask1 = _mm_cmpgt_epi8(k_8, clamped_absdiff);
+    const __m128i mask0 = _mm_cmpgt_epi8(k_4, clamped_absdiff);
+    // Get adjustments for l2, l1, and l0.
+    __m128i adj2 = _mm_and_si128(mask2, l32);
+    const __m128i adj1 = _mm_and_si128(mask1, l21);
+    const __m128i adj0 = _mm_and_si128(mask0, clamped_absdiff);
+    __m128i adj, padj, nadj;
+
+    // Combine the adjustments and get absolute adjustments.
+    adj2 = _mm_add_epi8(adj2, adj1);
+    adj = _mm_sub_epi8(l3, adj2);
+    adj = _mm_andnot_si128(mask0, adj);
+    adj = _mm_or_si128(adj, adj0);
+
+    // Restore the sign and get positive and negative adjustments.
+    padj = _mm_andnot_si128(diff_sign, adj);
+    nadj = _mm_and_si128(diff_sign, adj);
+
+    // Calculate filtered value.
+    v_running_avg_y = _mm_adds_epu8(v_sig, padj);
+    v_running_avg_y = _mm_subs_epu8(v_running_avg_y, nadj);
+    _mm_storeu_si128(reinterpret_cast<__m128i*>(running_avg_y),
+                     v_running_avg_y);
+
+    // Adjustments <=7, and each element in acc_diff can fit in signed
+    // char.
+    acc_diff = _mm_adds_epi8(acc_diff, padj);
+    acc_diff = _mm_subs_epi8(acc_diff, nadj);
+
+    // Update pointers for next iteration.
+    sig += sig_stride;
+    mc_running_avg_y += mc_avg_y_stride;
+    running_avg_y += avg_y_stride;
+  }
+
+  {
+    // Compute the sum of all pixel differences of this MB.
+    unsigned int abs_sum_diff = AbsSumDiff16x1(acc_diff);
+    unsigned int sum_diff_thresh = kSumDiffThreshold;
+    if (increase_denoising)
+      sum_diff_thresh = kSumDiffThresholdHigh;
+    if (abs_sum_diff > sum_diff_thresh) {
+      // Before returning to copy the block (i.e., apply no denoising),
+      // check if we can still apply some (weaker) temporal filtering to
+      // this block, that would otherwise not be denoised at all. Simplest
+      // is to apply an additional adjustment to running_avg_y to bring it
+      // closer to sig. The adjustment is capped by a maximum delta, and
+      // chosen such that in most cases the resulting sum_diff will be
+      // within the acceptable range given by sum_diff_thresh.
+
+      // The delta is set by the excess of absolute pixel diff over the
+      // threshold.
+      int delta = ((abs_sum_diff - sum_diff_thresh) >> 8) + 1;
+      // Only apply the adjustment for max delta up to 3.
+      if (delta < 4) {
+        const __m128i k_delta = _mm_set1_epi8(delta);
+        sig -= sig_stride * 16;
+        mc_running_avg_y -= mc_avg_y_stride * 16;
+        running_avg_y -= avg_y_stride * 16;
+        for (int r = 0; r < 16; ++r) {
+          __m128i v_running_avg_y =
+              _mm_loadu_si128(reinterpret_cast<__m128i*>(&running_avg_y[0]));
+          // Calculate differences.
+          const __m128i v_sig =
+              _mm_loadu_si128(reinterpret_cast<const __m128i*>(&sig[0]));
+          const __m128i v_mc_running_avg_y =
+              _mm_loadu_si128(reinterpret_cast<__m128i*>(&mc_running_avg_y[0]));
+          const __m128i pdiff = _mm_subs_epu8(v_mc_running_avg_y, v_sig);
+          const __m128i ndiff = _mm_subs_epu8(v_sig, v_mc_running_avg_y);
+          // Obtain the sign. FF if diff is negative.
+          const __m128i diff_sign = _mm_cmpeq_epi8(pdiff, k_0);
+          // Clamp absolute difference to delta to get the adjustment.
+          const __m128i adj = _mm_min_epu8(_mm_or_si128(pdiff, ndiff), k_delta);
+          // Restore the sign and get positive and negative adjustments.
+          __m128i padj, nadj;
+          padj = _mm_andnot_si128(diff_sign, adj);
+          nadj = _mm_and_si128(diff_sign, adj);
+          // Calculate filtered value.
+          v_running_avg_y = _mm_subs_epu8(v_running_avg_y, padj);
+          v_running_avg_y = _mm_adds_epu8(v_running_avg_y, nadj);
+          _mm_storeu_si128(reinterpret_cast<__m128i*>(running_avg_y),
+                           v_running_avg_y);
+
+          // Accumulate the adjustments.
+          acc_diff = _mm_subs_epi8(acc_diff, padj);
+          acc_diff = _mm_adds_epi8(acc_diff, nadj);
+
+          // Update pointers for next iteration.
+          sig += sig_stride;
+          mc_running_avg_y += mc_avg_y_stride;
+          running_avg_y += avg_y_stride;
+        }
+        abs_sum_diff = AbsSumDiff16x1(acc_diff);
+        if (abs_sum_diff > sum_diff_thresh) {
+          return COPY_BLOCK;
+        }
+      } else {
+        return COPY_BLOCK;
+      }
+    }
+  }
+  return FILTER_BLOCK;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/util/denoiser_filter_sse2.h b/webrtc/modules/video_processing/util/denoiser_filter_sse2.h
new file mode 100644
index 0000000000..31d8510902
--- /dev/null
+++ b/webrtc/modules/video_processing/util/denoiser_filter_sse2.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_SSE2_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_SSE2_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+
+namespace webrtc {
+
+class DenoiserFilterSSE2 : public DenoiserFilter {
+ public:
+  DenoiserFilterSSE2() {}
+  void CopyMem16x16(const uint8_t* src,
+                    int src_stride,
+                    uint8_t* dst,
+                    int dst_stride) override;
+  void CopyMem8x8(const uint8_t* src,
+                  int src_stride,
+                  uint8_t* dst,
+                  int dst_stride) override;
+  uint32_t Variance16x8(const uint8_t* a,
+                        int a_stride,
+                        const uint8_t* b,
+                        int b_stride,
+                        unsigned int* sse) override;
+  DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                             int mc_avg_y_stride,
+                             uint8_t* running_avg_y,
+                             int avg_y_stride,
+                             const uint8_t* sig,
+                             int sig_stride,
+                             uint8_t motion_magnitude,
+                             int increase_denoising) override;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_SSE2_H_
diff --git a/webrtc/modules/video_processing/util/skin_detection.cc b/webrtc/modules/video_processing/util/skin_detection.cc
new file mode 100644
index 0000000000..bf631ce2f6
--- /dev/null
+++ b/webrtc/modules/video_processing/util/skin_detection.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <limits.h>
+#include <math.h>
+
+#include "webrtc/modules/video_processing/util/skin_detection.h"
+
+namespace webrtc {
+
+// Fixed-point skin color model parameters.
+static const int skin_mean[2] = {7463, 9614};                 // q6
+static const int skin_inv_cov[4] = {4107, 1663, 1663, 2157};  // q16
+static const int skin_threshold = 1570636;                    // q18
+
+// Thresholds on luminance.
+static const int y_low = 20;
+static const int y_high = 220;
+
+// Evaluates the Mahalanobis distance measure for the input CbCr values.
+static int EvaluateSkinColorDifference(int cb, int cr) {
+  const int cb_q6 = cb << 6;
+  const int cr_q6 = cr << 6;
+  const int cb_diff_q12 = (cb_q6 - skin_mean[0]) * (cb_q6 - skin_mean[0]);
+  const int cbcr_diff_q12 = (cb_q6 - skin_mean[0]) * (cr_q6 - skin_mean[1]);
+  const int cr_diff_q12 = (cr_q6 - skin_mean[1]) * (cr_q6 - skin_mean[1]);
+  const int cb_diff_q2 = (cb_diff_q12 + (1 << 9)) >> 10;
+  const int cbcr_diff_q2 = (cbcr_diff_q12 + (1 << 9)) >> 10;
+  const int cr_diff_q2 = (cr_diff_q12 + (1 << 9)) >> 10;
+  const int skin_diff =
+      skin_inv_cov[0] * cb_diff_q2 + skin_inv_cov[1] * cbcr_diff_q2 +
+      skin_inv_cov[2] * cbcr_diff_q2 + skin_inv_cov[3] * cr_diff_q2;
+  return skin_diff;
+}
+
+bool MbHasSkinColor(const uint8_t* y_src,
+                    const uint8_t* u_src,
+                    const uint8_t* v_src,
+                    const int stride_y,
+                    const int stride_u,
+                    const int stride_v,
+                    const int mb_row,
+                    const int mb_col) {
+  const uint8_t* y = y_src + ((mb_row << 4) + 8) * stride_y + (mb_col << 4) + 8;
+  const uint8_t* u = u_src + ((mb_row << 3) + 4) * stride_u + (mb_col << 3) + 4;
+  const uint8_t* v = v_src + ((mb_row << 3) + 4) * stride_v + (mb_col << 3) + 4;
+  // Use 2x2 average of center pixel to compute skin area.
+  uint8_t y_avg = (*y + *(y + 1) + *(y + stride_y) + *(y + stride_y + 1)) >> 2;
+  uint8_t u_avg = (*u + *(u + 1) + *(u + stride_u) + *(u + stride_u + 1)) >> 2;
+  uint8_t v_avg = (*v + *(v + 1) + *(v + stride_v) + *(v + stride_v + 1)) >> 2;
+  // Ignore MB with too high or low brightness.
+  if (y_avg < y_low || y_avg > y_high)
+    return false;
+  else
+    return (EvaluateSkinColorDifference(u_avg, v_avg) < skin_threshold);
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/util/skin_detection.h b/webrtc/modules/video_processing/util/skin_detection.h
new file mode 100755
index 0000000000..561c03c425
--- /dev/null
+++ b/webrtc/modules/video_processing/util/skin_detection.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_SKIN_DETECTION_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_SKIN_DETECTION_H_
+
+namespace webrtc {
+
+typedef unsigned char uint8_t;
+bool MbHasSkinColor(const uint8_t* y_src,
+                    const uint8_t* u_src,
+                    const uint8_t* v_src,
+                    const int stride_y,
+                    const int stride_u,
+                    const int stride_v,
+                    const int mb_row,
+                    const int mb_col);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_SKIN_DETECTION_H_
diff --git a/webrtc/modules/video_processing/main/source/video_decimator.cc b/webrtc/modules/video_processing/video_decimator.cc
index 34c29c1677..63e347b026 100644
--- a/webrtc/modules/video_processing/main/source/video_decimator.cc
+++ b/webrtc/modules/video_processing/video_decimator.cc
@@ -9,8 +9,8 @@
  */
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/video_decimator.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/video_decimator.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
 #define VD_MIN(a, b) ((a) < (b)) ? (a) : (b)
@@ -23,7 +23,7 @@ VPMVideoDecimator::VPMVideoDecimator() {
 
 VPMVideoDecimator::~VPMVideoDecimator() {}
 
-void VPMVideoDecimator::Reset()  {
+void VPMVideoDecimator::Reset() {
   overshoot_modifier_ = 0;
   drop_count_ = 0;
   keep_count_ = 0;
@@ -43,14 +43,17 @@ void VPMVideoDecimator::SetTargetFramerate(int frame_rate) {
 }
 
 bool VPMVideoDecimator::DropFrame() {
-  if (!enable_temporal_decimation_) return false;
+  if (!enable_temporal_decimation_)
+    return false;
 
-  if (incoming_frame_rate_ <= 0) return false;
+  if (incoming_frame_rate_ <= 0)
+    return false;
 
   const uint32_t incomingframe_rate =
       static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
 
-  if (target_frame_rate_ == 0) return true;
+  if (target_frame_rate_ == 0)
+    return true;
 
   bool drop = false;
   if (incomingframe_rate > target_frame_rate_) {
@@ -61,44 +64,43 @@ bool VPMVideoDecimator::DropFrame() {
       overshoot_modifier_ = 0;
     }
 
-    if (overshoot && 2 * overshoot < (int32_t) incomingframe_rate) {
+    if (overshoot && 2 * overshoot < (int32_t)incomingframe_rate) {
       if (drop_count_) {  // Just got here so drop to be sure.
-          drop_count_ = 0;
-          return true;
+        drop_count_ = 0;
+        return true;
       }
       const uint32_t dropVar = incomingframe_rate / overshoot;
 
       if (keep_count_ >= dropVar) {
-          drop = true;
-          overshoot_modifier_ = -((int32_t) incomingframe_rate % overshoot) / 3;
-          keep_count_ = 1;
+        drop = true;
+        overshoot_modifier_ = -((int32_t)incomingframe_rate % overshoot) / 3;
+        keep_count_ = 1;
       } else {
-          keep_count_++;
+        keep_count_++;
       }
     } else {
       keep_count_ = 0;
       const uint32_t dropVar = overshoot / target_frame_rate_;
       if (drop_count_ < dropVar) {
-          drop = true;
-          drop_count_++;
+        drop = true;
+        drop_count_++;
       } else {
-          overshoot_modifier_ = overshoot % target_frame_rate_;
-          drop = false;
-          drop_count_ = 0;
+        overshoot_modifier_ = overshoot % target_frame_rate_;
+        drop = false;
+        drop_count_ = 0;
       }
     }
   }
   return drop;
 }
 
-
-uint32_t VPMVideoDecimator::Decimatedframe_rate() {
-ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
+uint32_t VPMVideoDecimator::GetDecimatedFrameRate() {
+  ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
   if (!enable_temporal_decimation_) {
     return static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
   }
   return VD_MIN(target_frame_rate_,
-      static_cast<uint32_t>(incoming_frame_rate_ + 0.5f));
+                static_cast<uint32_t>(incoming_frame_rate_ + 0.5f));
 }
 
 uint32_t VPMVideoDecimator::Inputframe_rate() {
@@ -113,7 +115,7 @@ void VPMVideoDecimator::UpdateIncomingframe_rate() {
   } else {
     // Shift.
     for (int i = kFrameCountHistory_size - 2; i >= 0; i--) {
-        incoming_frame_times_[i+1] = incoming_frame_times_[i];
+      incoming_frame_times_[i + 1] = incoming_frame_times_[i];
     }
   }
   incoming_frame_times_[0] = now;
@@ -133,7 +135,7 @@ void VPMVideoDecimator::ProcessIncomingframe_rate(int64_t now) {
     }
   }
   if (num > 1) {
-    int64_t diff = now - incoming_frame_times_[num-1];
+    int64_t diff = now - incoming_frame_times_[num - 1];
     incoming_frame_rate_ = 1.0;
     if (diff > 0) {
       incoming_frame_rate_ = nrOfFrames * 1000.0f / static_cast<float>(diff);
diff --git a/webrtc/modules/video_processing/main/source/video_decimator.h b/webrtc/modules/video_processing/video_decimator.h
index 3d4573caf8..1b871df8c3 100644
--- a/webrtc/modules/video_processing/main/source/video_decimator.h
+++ b/webrtc/modules/video_processing/video_decimator.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DECIMATOR_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DECIMATOR_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -32,7 +32,7 @@ class VPMVideoDecimator {
   void UpdateIncomingframe_rate();
 
   // Get Decimated Frame Rate/Dimensions.
-  uint32_t Decimatedframe_rate();
+  uint32_t GetDecimatedFrameRate();
 
   // Get input frame rate.
   uint32_t Inputframe_rate();
@@ -40,8 +40,8 @@ class VPMVideoDecimator {
  private:
   void ProcessIncomingframe_rate(int64_t now);
 
-  enum { kFrameCountHistory_size = 90};
-  enum { kFrameHistoryWindowMs = 2000};
+  enum { kFrameCountHistory_size = 90 };
+  enum { kFrameHistoryWindowMs = 2000 };
 
   // Temporal decimation.
   int32_t overshoot_modifier_;
@@ -55,4 +55,4 @@ class VPMVideoDecimator {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DECIMATOR_H_
diff --git a/webrtc/modules/video_processing/video_denoiser.cc b/webrtc/modules/video_processing/video_denoiser.cc
new file mode 100644
index 0000000000..4902a89491
--- /dev/null
+++ b/webrtc/modules/video_processing/video_denoiser.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/common_video/libyuv/include/scaler.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/video_denoiser.h"
+
+namespace webrtc {
+
+VideoDenoiser::VideoDenoiser(bool runtime_cpu_detection)
+    : width_(0),
+      height_(0),
+      filter_(DenoiserFilter::Create(runtime_cpu_detection)) {}
+
+void VideoDenoiser::TrailingReduction(int mb_rows,
+                                      int mb_cols,
+                                      const uint8_t* y_src,
+                                      int stride_y,
+                                      uint8_t* y_dst) {
+  for (int mb_row = 1; mb_row < mb_rows - 1; ++mb_row) {
+    for (int mb_col = 1; mb_col < mb_cols - 1; ++mb_col) {
+      int mb_index = mb_row * mb_cols + mb_col;
+      uint8_t* mb_dst = y_dst + (mb_row << 4) * stride_y + (mb_col << 4);
+      const uint8_t* mb_src = y_src + (mb_row << 4) * stride_y + (mb_col << 4);
+      // If the number of denoised neighbors is less than a threshold,
+      // do NOT denoise for the block. Set different threshold for skin MB.
+      // The change of denoising status will not propagate.
+      if (metrics_[mb_index].is_skin) {
+        // The threshold is high (more strict) for non-skin MB where the
+        // trailing usually happen.
+        if (metrics_[mb_index].denoise &&
+            metrics_[mb_index + 1].denoise + metrics_[mb_index - 1].denoise +
+                    metrics_[mb_index + mb_cols].denoise +
+                    metrics_[mb_index - mb_cols].denoise <=
+                2) {
+          metrics_[mb_index].denoise = 0;
+          filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+        }
+      } else if (metrics_[mb_index].denoise &&
+                 metrics_[mb_index + 1].denoise +
+                         metrics_[mb_index - 1].denoise +
+                         metrics_[mb_index + mb_cols + 1].denoise +
+                         metrics_[mb_index + mb_cols - 1].denoise +
+                         metrics_[mb_index - mb_cols + 1].denoise +
+                         metrics_[mb_index - mb_cols - 1].denoise +
+                         metrics_[mb_index + mb_cols].denoise +
+                         metrics_[mb_index - mb_cols].denoise <=
+                     7) {
+        filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+      }
+    }
+  }
+}
+
+void VideoDenoiser::DenoiseFrame(const VideoFrame& frame,
+                                 VideoFrame* denoised_frame) {
+  int stride_y = frame.stride(kYPlane);
+  int stride_u = frame.stride(kUPlane);
+  int stride_v = frame.stride(kVPlane);
+  // If previous width and height are different from current frame's, then no
+  // denoising for the current frame.
+  if (width_ != frame.width() || height_ != frame.height()) {
+    width_ = frame.width();
+    height_ = frame.height();
+    denoised_frame->CreateFrame(frame.buffer(kYPlane), frame.buffer(kUPlane),
+                                frame.buffer(kVPlane), width_, height_,
+                                stride_y, stride_u, stride_v);
+    // Setting time parameters to the output frame.
+    denoised_frame->set_timestamp(frame.timestamp());
+    denoised_frame->set_render_time_ms(frame.render_time_ms());
+    return;
+  }
+  // For 16x16 block.
+  int mb_cols = width_ >> 4;
+  int mb_rows = height_ >> 4;
+  if (metrics_.get() == nullptr)
+    metrics_.reset(new DenoiseMetrics[mb_cols * mb_rows]());
+  // Denoise on Y plane.
+  uint8_t* y_dst = denoised_frame->buffer(kYPlane);
+  uint8_t* u_dst = denoised_frame->buffer(kUPlane);
+  uint8_t* v_dst = denoised_frame->buffer(kVPlane);
+  const uint8_t* y_src = frame.buffer(kYPlane);
+  const uint8_t* u_src = frame.buffer(kUPlane);
+  const uint8_t* v_src = frame.buffer(kVPlane);
+  // Temporary buffer to store denoising result.
+  uint8_t y_tmp[16 * 16] = {0};
+  for (int mb_row = 0; mb_row < mb_rows; ++mb_row) {
+    for (int mb_col = 0; mb_col < mb_cols; ++mb_col) {
+      const uint8_t* mb_src = y_src + (mb_row << 4) * stride_y + (mb_col << 4);
+      uint8_t* mb_dst = y_dst + (mb_row << 4) * stride_y + (mb_col << 4);
+      int mb_index = mb_row * mb_cols + mb_col;
+      // Denoise each MB at the very start and save the result to a temporary
+      // buffer.
+      if (filter_->MbDenoise(mb_dst, stride_y, y_tmp, 16, mb_src, stride_y, 0,
+                             1) == FILTER_BLOCK) {
+        uint32_t thr_var = 0;
+        // Save var and sad to the buffer.
+        metrics_[mb_index].var = filter_->Variance16x8(
+            mb_dst, stride_y, y_tmp, 16, &metrics_[mb_index].sad);
+        // Get skin map.
+        metrics_[mb_index].is_skin = MbHasSkinColor(
+            y_src, u_src, v_src, stride_y, stride_u, stride_v, mb_row, mb_col);
+        // Variance threshold for skin/non-skin MB is different.
+        // Skin MB use a small threshold to reduce blockiness.
+        thr_var = metrics_[mb_index].is_skin ? 128 : 12 * 128;
+        if (metrics_[mb_index].var > thr_var) {
+          metrics_[mb_index].denoise = 0;
+          // Use the source MB.
+          filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+        } else {
+          metrics_[mb_index].denoise = 1;
+          // Use the denoised MB.
+          filter_->CopyMem16x16(y_tmp, 16, mb_dst, stride_y);
+        }
+      } else {
+        metrics_[mb_index].denoise = 0;
+        filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+      }
+      // Copy source U/V plane.
+      const uint8_t* mb_src_u =
+          u_src + (mb_row << 3) * stride_u + (mb_col << 3);
+      const uint8_t* mb_src_v =
+          v_src + (mb_row << 3) * stride_v + (mb_col << 3);
+      uint8_t* mb_dst_u = u_dst + (mb_row << 3) * stride_u + (mb_col << 3);
+      uint8_t* mb_dst_v = v_dst + (mb_row << 3) * stride_v + (mb_col << 3);
+      filter_->CopyMem8x8(mb_src_u, stride_u, mb_dst_u, stride_u);
+      filter_->CopyMem8x8(mb_src_v, stride_v, mb_dst_v, stride_v);
+    }
+  }
+  // Second round.
+  // This is to reduce the trailing artifact and blockiness by referring
+  // neighbors' denoising status.
+  TrailingReduction(mb_rows, mb_cols, y_src, stride_y, y_dst);
+
+  // Setting time parameters to the output frame.
+  denoised_frame->set_timestamp(frame.timestamp());
+  denoised_frame->set_render_time_ms(frame.render_time_ms());
+  return;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/video_denoiser.h b/webrtc/modules/video_processing/video_denoiser.h
new file mode 100644
index 0000000000..107a15ca07
--- /dev/null
+++ b/webrtc/modules/video_processing/video_denoiser.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DENOISER_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DENOISER_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+#include "webrtc/modules/video_processing/util/skin_detection.h"
+
+namespace webrtc {
+
+class VideoDenoiser {
+ public:
+  explicit VideoDenoiser(bool runtime_cpu_detection);
+  void DenoiseFrame(const VideoFrame& frame, VideoFrame* denoised_frame);
+
+ private:
+  void TrailingReduction(int mb_rows,
+                         int mb_cols,
+                         const uint8_t* y_src,
+                         int stride_y,
+                         uint8_t* y_dst);
+  int width_;
+  int height_;
+  rtc::scoped_ptr<DenoiseMetrics[]> metrics_;
+  rtc::scoped_ptr<DenoiserFilter> filter_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DENOISER_H_
diff --git a/webrtc/modules/video_processing/video_processing.gypi b/webrtc/modules/video_processing/video_processing.gypi
index 5827a5b1a6..7418c455a2 100644
--- a/webrtc/modules/video_processing/video_processing.gypi
+++ b/webrtc/modules/video_processing/video_processing.gypi
@@ -18,29 +18,38 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'main/interface/video_processing.h',
-        'main/interface/video_processing_defines.h',
-        'main/source/brighten.cc',
-        'main/source/brighten.h',
-        'main/source/brightness_detection.cc',
-        'main/source/brightness_detection.h',
-        'main/source/content_analysis.cc',
-        'main/source/content_analysis.h',
-        'main/source/deflickering.cc',
-        'main/source/deflickering.h',
-        'main/source/frame_preprocessor.cc',
-        'main/source/frame_preprocessor.h',
-        'main/source/spatial_resampler.cc',
-        'main/source/spatial_resampler.h',
-        'main/source/video_decimator.cc',
-        'main/source/video_decimator.h',
-        'main/source/video_processing_impl.cc',
-        'main/source/video_processing_impl.h',
+        'include/video_processing.h',
+        'include/video_processing_defines.h',
+        'brightness_detection.cc',
+        'brightness_detection.h',
+        'content_analysis.cc',
+        'content_analysis.h',
+        'deflickering.cc',
+        'deflickering.h',
+        'frame_preprocessor.cc',
+        'frame_preprocessor.h',
+        'spatial_resampler.cc',
+        'spatial_resampler.h',
+        'video_decimator.cc',
+        'video_decimator.h',
+        'video_processing_impl.cc',
+        'video_processing_impl.h',
+        'video_denoiser.cc',
+        'video_denoiser.h',
+        'util/denoiser_filter.cc',
+        'util/denoiser_filter.h',
+        'util/denoiser_filter_c.cc',
+        'util/denoiser_filter_c.h',
+        'util/skin_detection.cc',
+        'util/skin_detection.h',
       ],
       'conditions': [
         ['target_arch=="ia32" or target_arch=="x64"', {
           'dependencies': [ 'video_processing_sse2', ],
         }],
+        ['target_arch=="arm" or target_arch == "arm64"', {
+          'dependencies': [ 'video_processing_neon', ],
+        }],
       ],
     },
   ],
@@ -51,7 +60,9 @@
           'target_name': 'video_processing_sse2',
           'type': 'static_library',
           'sources': [
-            'main/source/content_analysis_sse2.cc',
+            'content_analysis_sse2.cc',
+            'util/denoiser_filter_sse2.cc',
+            'util/denoiser_filter_sse2.h',
           ],
           'conditions': [
             ['os_posix==1 and OS!="mac"', {
@@ -66,6 +77,19 @@
         },
       ],
     }],
+    ['target_arch=="arm" or target_arch == "arm64"', {
+      'targets': [
+        {
+          'target_name': 'video_processing_neon',
+          'type': 'static_library',
+          'includes': [ '../../build/arm_neon.gypi', ],
+          'sources': [
+            'util/denoiser_filter_neon.cc',
+            'util/denoiser_filter_neon.h',
+          ],
+        },
+      ],
+    }],
   ],
 }
 
diff --git a/webrtc/modules/video_processing/video_processing_impl.cc b/webrtc/modules/video_processing/video_processing_impl.cc
new file mode 100644
index 0000000000..f34886f10f
--- /dev/null
+++ b/webrtc/modules/video_processing/video_processing_impl.cc
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_processing/video_processing_impl.h"
+
+#include <assert.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+namespace {
+
+int GetSubSamplingFactor(int width, int height) {
+  if (width * height >= 640 * 480) {
+    return 3;
+  } else if (width * height >= 352 * 288) {
+    return 2;
+  } else if (width * height >= 176 * 144) {
+    return 1;
+  } else {
+    return 0;
+  }
+}
+}  // namespace
+
+VideoProcessing* VideoProcessing::Create() {
+  return new VideoProcessingImpl();
+}
+
+VideoProcessingImpl::VideoProcessingImpl() {}
+VideoProcessingImpl::~VideoProcessingImpl() {}
+
+void VideoProcessing::GetFrameStats(const VideoFrame& frame,
+                                    FrameStats* stats) {
+  ClearFrameStats(stats);  // The histogram needs to be zeroed out.
+  if (frame.IsZeroSize()) {
+    return;
+  }
+
+  int width = frame.width();
+  int height = frame.height();
+  stats->sub_sampling_factor = GetSubSamplingFactor(width, height);
+
+  const uint8_t* buffer = frame.buffer(kYPlane);
+  // Compute histogram and sum of frame
+  for (int i = 0; i < height; i += (1 << stats->sub_sampling_factor)) {
+    int k = i * width;
+    for (int j = 0; j < width; j += (1 << stats->sub_sampling_factor)) {
+      stats->hist[buffer[k + j]]++;
+      stats->sum += buffer[k + j];
+    }
+  }
+
+  stats->num_pixels = (width * height) / ((1 << stats->sub_sampling_factor) *
+                                          (1 << stats->sub_sampling_factor));
+  assert(stats->num_pixels > 0);
+
+  // Compute mean value of frame
+  stats->mean = stats->sum / stats->num_pixels;
+}
+
+bool VideoProcessing::ValidFrameStats(const FrameStats& stats) {
+  if (stats.num_pixels == 0) {
+    LOG(LS_WARNING) << "Invalid frame stats.";
+    return false;
+  }
+  return true;
+}
+
+void VideoProcessing::ClearFrameStats(FrameStats* stats) {
+  stats->mean = 0;
+  stats->sum = 0;
+  stats->num_pixels = 0;
+  stats->sub_sampling_factor = 0;
+  memset(stats->hist, 0, sizeof(stats->hist));
+}
+
+void VideoProcessing::Brighten(int delta, VideoFrame* frame) {
+  RTC_DCHECK(!frame->IsZeroSize());
+  RTC_DCHECK(frame->width() > 0);
+  RTC_DCHECK(frame->height() > 0);
+
+  int num_pixels = frame->width() * frame->height();
+
+  int look_up[256];
+  for (int i = 0; i < 256; i++) {
+    int val = i + delta;
+    look_up[i] = ((((val < 0) ? 0 : val) > 255) ? 255 : val);
+  }
+
+  uint8_t* temp_ptr = frame->buffer(kYPlane);
+  for (int i = 0; i < num_pixels; i++) {
+    *temp_ptr = static_cast<uint8_t>(look_up[*temp_ptr]);
+    temp_ptr++;
+  }
+}
+
+int32_t VideoProcessingImpl::Deflickering(VideoFrame* frame,
+                                          FrameStats* stats) {
+  rtc::CritScope mutex(&mutex_);
+  return deflickering_.ProcessFrame(frame, stats);
+}
+
+int32_t VideoProcessingImpl::BrightnessDetection(const VideoFrame& frame,
+                                                 const FrameStats& stats) {
+  rtc::CritScope mutex(&mutex_);
+  return brightness_detection_.ProcessFrame(frame, stats);
+}
+
+void VideoProcessingImpl::EnableTemporalDecimation(bool enable) {
+  rtc::CritScope mutex(&mutex_);
+  frame_pre_processor_.EnableTemporalDecimation(enable);
+}
+
+void VideoProcessingImpl::SetInputFrameResampleMode(
+    VideoFrameResampling resampling_mode) {
+  rtc::CritScope cs(&mutex_);
+  frame_pre_processor_.SetInputFrameResampleMode(resampling_mode);
+}
+
+int32_t VideoProcessingImpl::SetTargetResolution(uint32_t width,
+                                                 uint32_t height,
+                                                 uint32_t frame_rate) {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.SetTargetResolution(width, height, frame_rate);
+}
+
+void VideoProcessingImpl::SetTargetFramerate(int frame_rate) {
+  rtc::CritScope cs(&mutex_);
+  frame_pre_processor_.SetTargetFramerate(frame_rate);
+}
+
+uint32_t VideoProcessingImpl::GetDecimatedFrameRate() {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.GetDecimatedFrameRate();
+}
+
+uint32_t VideoProcessingImpl::GetDecimatedWidth() const {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.GetDecimatedWidth();
+}
+
+uint32_t VideoProcessingImpl::GetDecimatedHeight() const {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.GetDecimatedHeight();
+}
+
+void VideoProcessingImpl::EnableDenosing(bool enable) {
+  rtc::CritScope cs(&mutex_);
+  frame_pre_processor_.EnableDenosing(enable);
+}
+
+const VideoFrame* VideoProcessingImpl::PreprocessFrame(
+    const VideoFrame& frame) {
+  rtc::CritScope mutex(&mutex_);
+  return frame_pre_processor_.PreprocessFrame(frame);
+}
+
+VideoContentMetrics* VideoProcessingImpl::GetContentMetrics() const {
+  rtc::CritScope mutex(&mutex_);
+  return frame_pre_processor_.GetContentMetrics();
+}
+
+void VideoProcessingImpl::EnableContentAnalysis(bool enable) {
+  rtc::CritScope mutex(&mutex_);
+  frame_pre_processor_.EnableContentAnalysis(enable);
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_processing/video_processing_impl.h b/webrtc/modules/video_processing/video_processing_impl.h
new file mode 100644
index 0000000000..edbaba12fa
--- /dev/null
+++ b/webrtc/modules/video_processing/video_processing_impl.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_PROCESSING_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_PROCESSING_IMPL_H_
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/brightness_detection.h"
+#include "webrtc/modules/video_processing/deflickering.h"
+#include "webrtc/modules/video_processing/frame_preprocessor.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class VideoProcessingImpl : public VideoProcessing {
+ public:
+  VideoProcessingImpl();
+  ~VideoProcessingImpl() override;
+
+  // Implements VideoProcessing.
+  int32_t Deflickering(VideoFrame* frame, FrameStats* stats) override;
+  int32_t BrightnessDetection(const VideoFrame& frame,
+                              const FrameStats& stats) override;
+  void EnableTemporalDecimation(bool enable) override;
+  void SetInputFrameResampleMode(VideoFrameResampling resampling_mode) override;
+  void EnableContentAnalysis(bool enable) override;
+  int32_t SetTargetResolution(uint32_t width,
+                              uint32_t height,
+                              uint32_t frame_rate) override;
+  void SetTargetFramerate(int frame_rate) override;
+  uint32_t GetDecimatedFrameRate() override;
+  uint32_t GetDecimatedWidth() const override;
+  uint32_t GetDecimatedHeight() const override;
+  void EnableDenosing(bool enable) override;
+  const VideoFrame* PreprocessFrame(const VideoFrame& frame) override;
+  VideoContentMetrics* GetContentMetrics() const override;
+
+ private:
+  mutable rtc::CriticalSection mutex_;
+  VPMDeflickering deflickering_ GUARDED_BY(mutex_);
+  VPMBrightnessDetection brightness_detection_;
+  VPMFramePreprocessor frame_pre_processor_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_PROCESSING_IMPL_H_
diff --git a/webrtc/modules/video_render/BUILD.gn b/webrtc/modules/video_render/BUILD.gn
index 80f23870aa..0771bd7080 100644
--- a/webrtc/modules/video_render/BUILD.gn
+++ b/webrtc/modules/video_render/BUILD.gn
@@ -13,8 +13,8 @@ source_set("video_render_module") {
     "external/video_render_external_impl.cc",
     "external/video_render_external_impl.h",
     "i_video_render.h",
-    "include/video_render.h",
-    "include/video_render_defines.h",
+    "video_render.h",
+    "video_render_defines.h",
     "video_render_impl.h",
   ]
 
diff --git a/webrtc/modules/video_render/android/video_render_android_impl.cc b/webrtc/modules/video_render/android/video_render_android_impl.cc
index c647501963..9affb23d99 100644
--- a/webrtc/modules/video_render/android/video_render_android_impl.cc
+++ b/webrtc/modules/video_render/android/video_render_android_impl.cc
@@ -141,18 +141,13 @@ int32_t VideoRenderAndroid::StartRender() {
     return 0;
   }
 
-  _javaRenderThread = ThreadWrapper::CreateThread(JavaRenderThreadFun, this,
-                                                  "AndroidRenderThread");
+  _javaRenderThread.reset(new rtc::PlatformThread(JavaRenderThreadFun, this,
+                                                  "AndroidRenderThread"));
 
-  if (_javaRenderThread->Start())
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-                 "%s: thread started", __FUNCTION__);
-  else {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Could not start send thread", __FUNCTION__);
-    return -1;
-  }
-  _javaRenderThread->SetPriority(kRealtimePriority);
+  _javaRenderThread->Start();
+  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: thread started",
+               __FUNCTION__);
+  _javaRenderThread->SetPriority(rtc::kRealtimePriority);
   return 0;
 }
 
diff --git a/webrtc/modules/video_render/android/video_render_android_impl.h b/webrtc/modules/video_render/android/video_render_android_impl.h
index 34950db7d1..e5b7de4643 100644
--- a/webrtc/modules/video_render/android/video_render_android_impl.h
+++ b/webrtc/modules/video_render/android/video_render_android_impl.h
@@ -15,8 +15,8 @@
 
 #include <map>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 
 namespace webrtc {
@@ -144,7 +144,8 @@ class VideoRenderAndroid: IVideoRender {
   EventWrapper& _javaRenderEvent;
   int64_t _lastJavaRenderEvent;
   JNIEnv* _javaRenderJniEnv; // JNIEnv for the java render thread.
-  rtc::scoped_ptr<ThreadWrapper> _javaRenderThread;
+  // TODO(pbos): Remove scoped_ptr and use the member directly.
+  rtc::scoped_ptr<rtc::PlatformThread> _javaRenderThread;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/video_render/android/video_render_android_native_opengl2.h b/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
index b748b2dd47..8be247b834 100644
--- a/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
+++ b/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
@@ -15,7 +15,7 @@
 
 #include "webrtc/modules/video_render/android/video_render_android_impl.h"
 #include "webrtc/modules/video_render/android/video_render_opengles20.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_render/android/video_render_android_surface_view.h b/webrtc/modules/video_render/android/video_render_android_surface_view.h
index 480e8b5106..0f029b54f3 100644
--- a/webrtc/modules/video_render/android/video_render_android_surface_view.h
+++ b/webrtc/modules/video_render/android/video_render_android_surface_view.h
@@ -14,7 +14,7 @@
 #include <jni.h>
 
 #include "webrtc/modules/video_render/android/video_render_android_impl.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_render/android/video_render_opengles20.h b/webrtc/modules/video_render/android/video_render_opengles20.h
index 44ab4c04bc..57e2a10d42 100644
--- a/webrtc/modules/video_render/android/video_render_opengles20.h
+++ b/webrtc/modules/video_render/android/video_render_opengles20.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #include <GLES2/gl2.h>
 #include <GLES2/gl2ext.h>
diff --git a/webrtc/modules/video_render/external/video_render_external_impl.h b/webrtc/modules/video_render/external/video_render_external_impl.h
index 9230e60acc..a8b663fff7 100644
--- a/webrtc/modules/video_render/external/video_render_external_impl.h
+++ b/webrtc/modules/video_render/external/video_render_external_impl.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_render/i_video_render.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
diff --git a/webrtc/modules/video_render/i_video_render.h b/webrtc/modules/video_render/i_video_render.h
index ff1cce782e..e6ec7a4680 100644
--- a/webrtc/modules/video_render/i_video_render.h
+++ b/webrtc/modules/video_render/i_video_render.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
 
-#include "webrtc/modules/video_render/include/video_render.h"
+#include "webrtc/modules/video_render/video_render.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_render/ios/open_gles20.h b/webrtc/modules/video_render/ios/open_gles20.h
index f74955fee4..880ddb5231 100644
--- a/webrtc/modules/video_render/ios/open_gles20.h
+++ b/webrtc/modules/video_render/ios/open_gles20.h
@@ -13,7 +13,7 @@
 
 #include <OpenGLES/ES2/glext.h>
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 /*
  * This OpenGles20 is the class of renderer for VideoFrame into a GLES 2.0
diff --git a/webrtc/modules/video_render/ios/video_render_ios_channel.h b/webrtc/modules/video_render/ios/video_render_ios_channel.h
index 375a5ee719..a15ba393dc 100644
--- a/webrtc/modules/video_render/ios/video_render_ios_channel.h
+++ b/webrtc/modules/video_render/ios/video_render_ios_channel.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_CHANNEL_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_CHANNEL_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/modules/video_render/ios/video_render_ios_view.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/video_render/ios/video_render_ios_gles20.h b/webrtc/modules/video_render/ios/video_render_ios_gles20.h
index b6da12aab8..d703630d92 100644
--- a/webrtc/modules/video_render/ios/video_render_ios_gles20.h
+++ b/webrtc/modules/video_render/ios/video_render_ios_gles20.h
@@ -14,10 +14,10 @@
 #include <list>
 #include <map>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/video_render/ios/video_render_ios_channel.h"
 #include "webrtc/modules/video_render/ios/video_render_ios_view.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 namespace webrtc {
 
@@ -64,7 +64,8 @@ class VideoRenderIosGles20 {
  private:
   rtc::scoped_ptr<CriticalSectionWrapper> gles_crit_sec_;
   EventTimerWrapper* screen_update_event_;
-  rtc::scoped_ptr<ThreadWrapper> screen_update_thread_;
+  // TODO(pbos): Remove scoped_ptr and use member directly.
+  rtc::scoped_ptr<rtc::PlatformThread> screen_update_thread_;
 
   VideoRenderIosView* view_;
   Rect window_rect_;
diff --git a/webrtc/modules/video_render/ios/video_render_ios_gles20.mm b/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
index 3a276d6030..6ad5db8b8c 100644
--- a/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
+++ b/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
@@ -32,15 +32,15 @@ VideoRenderIosGles20::VideoRenderIosGles20(VideoRenderIosView* view,
       z_order_to_channel_(),
       gles_context_([view context]),
       is_rendering_(true) {
-  screen_update_thread_ = ThreadWrapper::CreateThread(
-      ScreenUpdateThreadProc, this, "ScreenUpdateGles20");
+  screen_update_thread_.reset(new rtc::PlatformThread(
+      ScreenUpdateThreadProc, this, "ScreenUpdateGles20"));
   screen_update_event_ = EventTimerWrapper::Create();
   GetWindowRect(window_rect_);
 }
 
 VideoRenderIosGles20::~VideoRenderIosGles20() {
   // Signal event to exit thread, then delete it
-  ThreadWrapper* thread_wrapper = screen_update_thread_.release();
+  rtc::PlatformThread* thread_wrapper = screen_update_thread_.release();
 
   if (thread_wrapper) {
     screen_update_event_->Set();
@@ -83,7 +83,7 @@ int VideoRenderIosGles20::Init() {
   }
 
   screen_update_thread_->Start();
-  screen_update_thread_->SetPriority(kRealtimePriority);
+  screen_update_thread_->SetPriority(rtc::kRealtimePriority);
 
   // Start the event triggering the render process
   unsigned int monitor_freq = 60;
diff --git a/webrtc/modules/video_render/linux/video_x11_channel.h b/webrtc/modules/video_render/linux/video_x11_channel.h
index 8cbd2a27d1..6eb402e12e 100644
--- a/webrtc/modules/video_render/linux/video_x11_channel.h
+++ b/webrtc/modules/video_render/linux/video_x11_channel.h
@@ -13,7 +13,7 @@
 
 #include <sys/shm.h>
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #include <X11/Xlib.h>
 #include <X11/Xutil.h>
diff --git a/webrtc/modules/video_render/linux/video_x11_render.h b/webrtc/modules/video_render/linux/video_x11_render.h
index 265ef7cfab..23b83bd67b 100644
--- a/webrtc/modules/video_render/linux/video_x11_render.h
+++ b/webrtc/modules/video_render/linux/video_x11_render.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #include <X11/Xlib.h>
 #include <map>
diff --git a/webrtc/modules/video_render/mac/video_render_agl.cc b/webrtc/modules/video_render/mac/video_render_agl.cc
index dc157d597b..3243563b2b 100644
--- a/webrtc/modules/video_render/mac/video_render_agl.cc
+++ b/webrtc/modules/video_render/mac/video_render_agl.cc
@@ -395,8 +395,8 @@ _renderingIsPaused( false),
 {
     //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s");
 
-    _screenUpdateThread = ThreadWrapper::CreateThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdate");
+    _screenUpdateThread.reset(
+        new rtc::PlatformThread(ScreenUpdateThreadProc, this, "ScreenUpdate"));
     _screenUpdateEvent = EventWrapper::Create();
 
     if(!IsValidWindowPtr(_windowRef))
@@ -512,8 +512,8 @@ _renderingIsPaused( false),
     //WEBRTC_TRACE(kTraceDebug, "%s:%d Constructor", __FUNCTION__, __LINE__);
     //    _renderCritSec = CriticalSectionWrapper::CreateCriticalSection();
 
-    _screenUpdateThread = ThreadWrapper::CreateThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdateThread");
+    _screenUpdateThread.reset(new rtc::PlatformThread(
+        ScreenUpdateThreadProc, this, "ScreenUpdateThread"));
     _screenUpdateEvent = EventWrapper::Create();
 
     GetWindowRect(_windowRect);
@@ -677,7 +677,7 @@ VideoRenderAGL::~VideoRenderAGL()
 #endif
 
     // Signal event to exit thread, then delete it
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
 
     if (tmpPtr)
     {
@@ -739,7 +739,7 @@ int VideoRenderAGL::Init()
         return -1;
     }
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
     // Start the event triggering the render process
     unsigned int monitorFreq = 60;
@@ -856,7 +856,7 @@ int VideoRenderAGL::DeleteAGLChannel(int channel)
 int VideoRenderAGL::StopThread()
 {
     CriticalSectionScoped cs(&_renderCritSec);
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
 
     if (tmpPtr)
     {
@@ -1880,7 +1880,7 @@ int32_t VideoRenderAGL::StartRender()
             UnlockAGLCntx();
             return -1;
         }
-        _screenUpdateThread->SetPriority(kRealtimePriority);
+        _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
         if(FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
         {
             //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateEvent", __FUNCTION__, __LINE__);
@@ -1891,8 +1891,8 @@ int32_t VideoRenderAGL::StartRender()
         return 0;
     }
 
-    _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc,
-        this, "ScreenUpdate");
+    _screenUpdateThread.reset(
+        new rtc::PlatformThread(ScreenUpdateThreadProc, this, "ScreenUpdate"));
     _screenUpdateEvent = EventWrapper::Create();
 
     if (!_screenUpdateThread)
@@ -1903,14 +1903,13 @@ int32_t VideoRenderAGL::StartRender()
     }
 
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
     _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ);
 
     //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Started screenUpdateThread", __FUNCTION__, __LINE__);
 
     UnlockAGLCntx();
     return 0;
-
 }
 
 int32_t VideoRenderAGL::StopRender()
diff --git a/webrtc/modules/video_render/mac/video_render_agl.h b/webrtc/modules/video_render/mac/video_render_agl.h
index 8710228754..e1da8faf83 100644
--- a/webrtc/modules/video_render/mac/video_render_agl.h
+++ b/webrtc/modules/video_render/mac/video_render_agl.h
@@ -15,8 +15,8 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #define NEW_HIVIEW_PARENT_EVENT_HANDLER 1
 #define NEW_HIVIEW_EVENT_HANDLER 1
@@ -142,7 +142,8 @@ class VideoRenderAGL {
   bool _fullScreen;
   int _id;
   webrtc::CriticalSectionWrapper& _renderCritSec;
-  rtc::scoped_ptr<webrtc::ThreadWrapper> _screenUpdateThread;
+  // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+  rtc::scoped_ptr<rtc::PlatformThread> _screenUpdateThread;
   webrtc::EventWrapper* _screenUpdateEvent;
   bool _isHIViewRef;
   AGLContext _aglContext;
diff --git a/webrtc/modules/video_render/mac/video_render_nsopengl.h b/webrtc/modules/video_render/mac/video_render_nsopengl.h
index 5dab4d266f..a888b68a97 100644
--- a/webrtc/modules/video_render/mac/video_render_nsopengl.h
+++ b/webrtc/modules/video_render/mac/video_render_nsopengl.h
@@ -23,16 +23,19 @@
 #include <map>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #import "webrtc/modules/video_render/mac/cocoa_full_screen_window.h"
 #import "webrtc/modules/video_render/mac/cocoa_render_view.h"
 
 class Trace;
 
+namespace rtc {
+class PlatformThread;
+}  // namespace rtc
+
 namespace webrtc {
 class EventTimerWrapper;
-class ThreadWrapper;
 class VideoRenderNSOpenGL;
 class CriticalSectionWrapper;
 
@@ -166,7 +169,8 @@ private: // variables
     bool _fullScreen;
     int _id;
     CriticalSectionWrapper& _nsglContextCritSec;
-    rtc::scoped_ptr<ThreadWrapper> _screenUpdateThread;
+    // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+    rtc::scoped_ptr<rtc::PlatformThread> _screenUpdateThread;
     EventTimerWrapper* _screenUpdateEvent;
     NSOpenGLContext* _nsglContext;
     NSOpenGLContext* _nsglFullScreenContext;
diff --git a/webrtc/modules/video_render/mac/video_render_nsopengl.mm b/webrtc/modules/video_render/mac/video_render_nsopengl.mm
index b5150eb668..b7683a96af 100644
--- a/webrtc/modules/video_render/mac/video_render_nsopengl.mm
+++ b/webrtc/modules/video_render/mac/video_render_nsopengl.mm
@@ -11,11 +11,11 @@
 #include "webrtc/engine_configurations.h"
 #if defined(COCOA_RENDERING)
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_render/mac/video_render_nsopengl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -378,8 +378,8 @@ _renderingIsPaused (FALSE),
 _windowRefSuperView(NULL),
 _windowRefSuperViewFrame(NSMakeRect(0,0,0,0))
 {
-    _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc,
-            this, "ScreenUpdateNSOpenGL");
+  _screenUpdateThread.reset(new rtc::PlatformThread(
+      ScreenUpdateThreadProc, this, "ScreenUpdateNSOpenGL"));
 }
 
 int VideoRenderNSOpenGL::ChangeWindow(CocoaRenderView* newWindowRef)
@@ -427,15 +427,15 @@ int32_t VideoRenderNSOpenGL::StartRender()
         WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Restarting screenUpdateThread");
 
         // we already have the thread. Most likely StopRender() was called and they were paused
-        if(FALSE == _screenUpdateThread->Start() ||
-                FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
-        {
+        _screenUpdateThread->Start();
+        if (FALSE ==
+            _screenUpdateEvent->StartTimer(true, 1000 / MONITOR_FREQ)) {
             WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "Failed to restart screenUpdateThread or screenUpdateEvent");
             UnlockAGLCntx();
             return -1;
         }
 
-        _screenUpdateThread->SetPriority(kRealtimePriority);
+        _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
         UnlockAGLCntx();
         return 0;
@@ -471,8 +471,8 @@ int32_t VideoRenderNSOpenGL::StopRender()
         return 0;
     }
 
-    if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
-    {
+    _screenUpdateThread->Stop();
+    if (FALSE == _screenUpdateEvent->StopTimer()) {
         _renderingIsPaused = FALSE;
 
         UnlockAGLCntx();
@@ -657,17 +657,15 @@ VideoRenderNSOpenGL::~VideoRenderNSOpenGL()
     }
 
     // Signal event to exit thread, then delete it
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
 
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
         _screenUpdateEvent->StopTimer();
 
-        if (tmpPtr->Stop())
-        {
-            delete tmpPtr;
-        }
+        tmpPtr->Stop();
+        delete tmpPtr;
         delete _screenUpdateEvent;
         _screenUpdateEvent = NULL;
     }
@@ -716,7 +714,7 @@ int VideoRenderNSOpenGL::Init()
     }
 
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
     // Start the event triggering the render process
     unsigned int monitorFreq = 60;
@@ -864,17 +862,15 @@ int32_t VideoRenderNSOpenGL::GetChannelProperties(const uint16_t streamId,
 int VideoRenderNSOpenGL::StopThread()
 {
 
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
     WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
                  "%s Stopping thread ", __FUNCTION__, tmpPtr);
 
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
-        if (tmpPtr->Stop())
-        {
-            delete tmpPtr;
-        }
+        tmpPtr->Stop();
+        delete tmpPtr;
     }
 
     delete _screenUpdateEvent;
diff --git a/webrtc/modules/video_render/test/testAPI/testAPI.cc b/webrtc/modules/video_render/test/testAPI/testAPI.cc
index 256d031c0c..cea2f6b56f 100644
--- a/webrtc/modules/video_render/test/testAPI/testAPI.cc
+++ b/webrtc/modules/video_render/test/testAPI/testAPI.cc
@@ -32,10 +32,10 @@
 #endif
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_render/video_render.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -244,7 +244,7 @@ int WebRtcCreateWindow(Window *outWindow, Display **outDisplay, int winNum, int
 
     return 0;
 }
-#endif  // LINUX
+#endif  // WEBRTC_LINUX
 
 // Note: Mac code is in testApi_mac.mm.
 
diff --git a/webrtc/modules/video_render/test/testAPI/testAPI.h b/webrtc/modules/video_render/test/testAPI/testAPI.h
index 8b14e84931..0655a5b434 100644
--- a/webrtc/modules/video_render/test/testAPI/testAPI.h
+++ b/webrtc/modules/video_render/test/testAPI/testAPI.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 void RunVideoRenderTests(void* window, webrtc::VideoRenderType windowType);
 
diff --git a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
index dd57397c73..dfee4c7298 100644
--- a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
+++ b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
@@ -20,10 +20,10 @@
 
 #import "webrtc/modules/video_render/mac/cocoa_render_view.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_render/video_render.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
diff --git a/webrtc/modules/video_render/video_render.gypi b/webrtc/modules/video_render/video_render.gypi
index 63f69b0a63..e8cc03a4b0 100644
--- a/webrtc/modules/video_render/video_render.gypi
+++ b/webrtc/modules/video_render/video_render.gypi
@@ -25,8 +25,8 @@
         'external/video_render_external_impl.cc',
         'external/video_render_external_impl.h',
         'i_video_render.h',
-        'include/video_render.h',
-        'include/video_render_defines.h',
+        'video_render.h',
+        'video_render_defines.h',
         'video_render_impl.h',
       ],
     },
@@ -149,13 +149,28 @@
                 '<(directx_sdk_path)/Include',
               ],
             }],
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-comment',
+                    '-Wno-reorder',
+                    '-Wno-unused-value',
+                    '-Wno-unused-private-field',
+                  ],
+                },
+              },
+            }],
           ] # conditions
         },
       ],
     }], # build_with_chromium==0
-    ['include_tests==1', {
+    ['include_tests==1 and OS!="ios"', {
       'targets': [
         {
+          # Does not compile on iOS: webrtc:4755.
           'target_name': 'video_render_tests',
           'type': 'executable',
           'dependencies': [
@@ -197,7 +212,7 @@
           ] # conditions
         }, # video_render_module_test
       ], # targets
-    }], # include_tests==1
+    }], # include_tests==1 and OS!=ios
   ], # conditions
 }
 
diff --git a/webrtc/modules/video_render/include/video_render.h b/webrtc/modules/video_render/video_render.h
index 51fcce10c3..a193a187e7 100644
--- a/webrtc/modules/video_render/include/video_render.h
+++ b/webrtc/modules/video_render/video_render.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
 
 /*
  * video_render.h
@@ -20,8 +20,8 @@
  *
  */
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 namespace webrtc {
 
@@ -265,4 +265,4 @@ public:
                                     const uint32_t timeout) = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
+#endif  // WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
diff --git a/webrtc/modules/video_render/include/video_render_defines.h b/webrtc/modules/video_render/video_render_defines.h
index f8f48035ea..999707cb6e 100644
--- a/webrtc/modules/video_render/include/video_render_defines.h
+++ b/webrtc/modules/video_render/video_render_defines.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/incoming_video_stream.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc
 {
@@ -67,4 +67,4 @@ enum StretchMode
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
+#endif  // WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
diff --git a/webrtc/modules/video_render/video_render_impl.cc b/webrtc/modules/video_render/video_render_impl.cc
index 32ec2b0e75..d2a074b4c4 100644
--- a/webrtc/modules/video_render/video_render_impl.cc
+++ b/webrtc/modules/video_render/video_render_impl.cc
@@ -10,11 +10,11 @@
 
 #include <assert.h>
 
-#include "webrtc/common_video/interface/incoming_video_stream.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/video_render/external/video_render_external_impl.h"
 #include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/modules/video_render/video_render_impl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -197,7 +197,8 @@ ModuleVideoRenderImpl::AddIncomingRenderStream(const uint32_t streamId,
     }
 
     // Create platform independant code
-    IncomingVideoStream* ptrIncomingStream = new IncomingVideoStream(streamId);
+    IncomingVideoStream* ptrIncomingStream =
+        new IncomingVideoStream(streamId, false);
     ptrIncomingStream->SetRenderCallback(ptrRenderCallback);
     VideoRenderCallback* moduleCallback = ptrIncomingStream->ModuleCallback();
 
diff --git a/webrtc/modules/video_render/video_render_impl.h b/webrtc/modules/video_render/video_render_impl.h
index cc00897cc3..ce93cea6b5 100644
--- a/webrtc/modules/video_render/video_render_impl.h
+++ b/webrtc/modules/video_render/video_render_impl.h
@@ -14,7 +14,7 @@
 #include <map>
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/include/video_render.h"
+#include "webrtc/modules/video_render/video_render.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
diff --git a/webrtc/modules/video_render/video_render_internal_impl.cc b/webrtc/modules/video_render/video_render_internal_impl.cc
index 09ebc7e548..1fed26e9c4 100644
--- a/webrtc/modules/video_render/video_render_internal_impl.cc
+++ b/webrtc/modules/video_render/video_render_internal_impl.cc
@@ -10,10 +10,10 @@
 
 #include <assert.h>
 
-#include "webrtc/common_video/interface/incoming_video_stream.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/modules/video_render/video_render_impl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -420,7 +420,8 @@ ModuleVideoRenderImpl::AddIncomingRenderStream(const uint32_t streamId,
     }
 
     // Create platform independant code
-    IncomingVideoStream* ptrIncomingStream = new IncomingVideoStream(streamId);
+    IncomingVideoStream* ptrIncomingStream =
+        new IncomingVideoStream(streamId, false);
     ptrIncomingStream->SetRenderCallback(ptrRenderCallback);
     VideoRenderCallback* moduleCallback = ptrIncomingStream->ModuleCallback();
 
diff --git a/webrtc/modules/video_render/windows/i_video_render_win.h b/webrtc/modules/video_render/windows/i_video_render_win.h
index 56731e3770..6dbb4fd3cb 100644
--- a/webrtc/modules/video_render/windows/i_video_render_win.h
+++ b/webrtc/modules/video_render/windows/i_video_render_win.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
 
-#include "webrtc/modules/video_render/include/video_render.h"
+#include "webrtc/modules/video_render/video_render.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/video_render/windows/video_render_direct3d9.cc b/webrtc/modules/video_render/windows/video_render_direct3d9.cc
index 24dd0efddb..83835aebb8 100644
--- a/webrtc/modules/video_render/windows/video_render_direct3d9.cc
+++ b/webrtc/modules/video_render/windows/video_render_direct3d9.cc
@@ -294,8 +294,8 @@ VideoRenderDirect3D9::VideoRenderDirect3D9(Trace* trace,
     _totalMemory(0),
     _availableMemory(0)
 {
-    _screenUpdateThread = ThreadWrapper::CreateThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdateThread");
+    _screenUpdateThread.reset(new rtc::PlatformThread(
+        ScreenUpdateThreadProc, this, "ScreenUpdateThread"));
     _screenUpdateEvent = EventTimerWrapper::Create();
     SetRect(&_originalHwndRect, 0, 0, 0, 0);
 }
@@ -305,7 +305,7 @@ VideoRenderDirect3D9::~VideoRenderDirect3D9()
     //NOTE: we should not enter CriticalSection in here!
 
     // Signal event to exit thread, then delete it
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
@@ -546,7 +546,7 @@ int32_t VideoRenderDirect3D9::Init()
         return -1;
     }
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
     // Start the event triggering the render process
     unsigned int monitorFreq = 60;
diff --git a/webrtc/modules/video_render/windows/video_render_direct3d9.h b/webrtc/modules/video_render/windows/video_render_direct3d9.h
index c8f6639d9a..5a1f207934 100644
--- a/webrtc/modules/video_render/windows/video_render_direct3d9.h
+++ b/webrtc/modules/video_render/windows/video_render_direct3d9.h
@@ -19,8 +19,8 @@
 #include <Map>
 
 // Added
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #pragma comment(lib, "d3d9.lib")       // located in DirectX SDK
 
@@ -203,7 +203,8 @@ private:
 
     CriticalSectionWrapper& _refD3DCritsect;
     Trace* _trace;
-    rtc::scoped_ptr<ThreadWrapper> _screenUpdateThread;
+    // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+    rtc::scoped_ptr<rtc::PlatformThread> _screenUpdateThread;
     EventTimerWrapper* _screenUpdateEvent;
 
     HWND _hWnd;