Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c34b9e5d5cd44c31c4f9da649b71d0d3132cf516

This commit was generated by merge_from_chromium.py.

Change-Id: Ibdea97e7e6e800b2b7d7d3122a1d77e467cfbde4

54 files changed, 821 insertions, 168 deletions

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
new file mode 100644
index 00000000..e4ad4cb3
--- /dev/null
+++ b/modules/audio_coding/BUILD.gn
@@ -0,0 +1,13 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("audio_coding") {
+  # TODO(andrew): Implement.
+}
diff --git a/modules/audio_coding/OWNERS b/modules/audio_coding/OWNERS
new file mode 100644
index 00000000..d5ae8473
--- /dev/null
+++ b/modules/audio_coding/OWNERS
@@ -0,0 +1 @@
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/modules/audio_coding/main/acm2/acm_cng.h b/modules/audio_coding/main/acm2/acm_cng.h
index a0c1c554..8b0a3928 100644
--- a/modules/audio_coding/main/acm2/acm_cng.h
+++ b/modules/audio_coding/main/acm2/acm_cng.h
@@ -35,7 +35,8 @@ class ACMCNG: public ACMGenericCodec {
   int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
  protected:
-  void DestructEncoderSafe();
+  void DestructEncoderSafe() OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalCreateEncoder();
 
diff --git a/modules/audio_coding/main/acm2/acm_g722.h b/modules/audio_coding/main/acm2/acm_g722.h
index 6197a9d9..6cdd79d4 100644
--- a/modules/audio_coding/main/acm2/acm_g722.h
+++ b/modules/audio_coding/main/acm2/acm_g722.h
@@ -33,7 +33,9 @@ class ACMG722 : public ACMGenericCodec {
   // For FEC.
   ACMGenericCodec* CreateInstance(void);
 
-  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
 
@@ -44,7 +46,8 @@ class ACMG722 : public ACMGenericCodec {
                           const uint8_t audio_channel)
       EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
-  void DestructEncoderSafe();
+  void DestructEncoderSafe() OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalCreateEncoder();
 
diff --git a/modules/audio_coding/main/acm2/acm_generic_codec.cc b/modules/audio_coding/main/acm2/acm_generic_codec.cc
index a4808c0e..d16d1d38 100644
--- a/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ b/modules/audio_coding/main/acm2/acm_generic_codec.cc
@@ -60,7 +60,6 @@ ACMGenericCodec::ACMGenericCodec()
       sent_cn_previous_(false),
       prev_frame_cng_(0),
       has_internal_fec_(false),
-      neteq_decode_lock_(NULL),
       codec_wrapper_lock_(*RWLockWrapper::CreateRWLock()),
       last_timestamp_(0xD87F3F9F),
       unique_id_(0) {
@@ -210,7 +209,6 @@ int16_t ACMGenericCodec::Encode(uint8_t* bitstream,
     return 0;
   }
   WriteLockScoped lockCodec(codec_wrapper_lock_);
-  ReadLockScoped lockNetEq(*neteq_decode_lock_);
 
   // Not all codecs accept the whole frame to be pushed into encoder at once.
   // Some codecs needs to be feed with a specific number of samples different
@@ -394,7 +392,6 @@ int16_t ACMGenericCodec::EncoderParamsSafe(WebRtcACMCodecParams* enc_params) {
 
 int16_t ACMGenericCodec::ResetEncoder() {
   WriteLockScoped lockCodec(codec_wrapper_lock_);
-  ReadLockScoped lockNetEq(*neteq_decode_lock_);
   return ResetEncoderSafe();
 }
 
@@ -443,7 +440,6 @@ int16_t ACMGenericCodec::InternalResetEncoder() {
 int16_t ACMGenericCodec::InitEncoder(WebRtcACMCodecParams* codec_params,
                                      bool force_initialization) {
   WriteLockScoped lockCodec(codec_wrapper_lock_);
-  ReadLockScoped lockNetEq(*neteq_decode_lock_);
   return InitEncoderSafe(codec_params, force_initialization);
 }
 
@@ -629,7 +625,6 @@ int16_t ACMGenericCodec::CreateEncoder() {
 void ACMGenericCodec::DestructEncoderInst(void* ptr_inst) {
   if (ptr_inst != NULL) {
     WriteLockScoped lockCodec(codec_wrapper_lock_);
-    ReadLockScoped lockNetEq(*neteq_decode_lock_);
     InternalDestructEncoderInst(ptr_inst);
   }
 }
diff --git a/modules/audio_coding/main/acm2/acm_generic_codec.h b/modules/audio_coding/main/acm2/acm_generic_codec.h
index fa21ca01..80f239ac 100644
--- a/modules/audio_coding/main/acm2/acm_generic_codec.h
+++ b/modules/audio_coding/main/acm2/acm_generic_codec.h
@@ -294,17 +294,6 @@ class ACMGenericCodec {
   int32_t IsInternalDTXReplaced(bool* internal_dtx_replaced);
 
   ///////////////////////////////////////////////////////////////////////////
-  // void SetNetEqDecodeLock()
-  // Passes the NetEq lock to the codec.
-  //
-  // Input:
-  //   -neteq_decode_lock  : pointer to the lock associated with NetEQ of ACM.
-  //
-  void SetNetEqDecodeLock(RWLockWrapper* neteq_decode_lock) {
-    neteq_decode_lock_ = neteq_decode_lock;
-  }
-
-  ///////////////////////////////////////////////////////////////////////////
   // bool HasInternalDTX()
   // Used to check if the codec has internal DTX.
   //
@@ -312,7 +301,10 @@ class ACMGenericCodec {
   //   true if the codec has an internal DTX, e.g. G729,
   //   false otherwise.
   //
-  bool HasInternalDTX() const { return has_internal_dtx_; }
+  bool HasInternalDTX() const {
+    ReadLockScoped rl(codec_wrapper_lock_);
+    return has_internal_dtx_;
+  }
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t GetEstimatedBandwidth()
@@ -436,7 +428,8 @@ class ACMGenericCodec {
   //   -1 if failed, or if this is meaningless for the given codec.
   //    0 if succeeded.
   //
-  virtual int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz);
+  virtual int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz)
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   ///////////////////////////////////////////////////////////////////////////
   // EncoderSampFreq()
@@ -450,7 +443,8 @@ class ACMGenericCodec {
   //   -1 if failed to output sampling rate.
   //    0 if the sample rate is returned successfully.
   //
-  virtual int16_t EncoderSampFreq(uint16_t* samp_freq_hz);
+  virtual int16_t EncoderSampFreq(uint16_t* samp_freq_hz)
+      SHARED_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t ConfigISACBandwidthEstimator()
@@ -515,8 +509,6 @@ class ACMGenericCodec {
   //
   virtual int32_t SetISACMaxRate(const uint32_t max_rate_bps);
 
-  int32_t FrameSize() { return frame_len_smpl_; }
-
   ///////////////////////////////////////////////////////////////////////////
   // REDPayloadISAC()
   // This is an iSAC-specific function. The function is called to get RED
@@ -569,7 +561,10 @@ class ACMGenericCodec {
   //   true if the codec has an internal FEC, e.g. Opus.
   //   false otherwise.
   //
-  bool HasInternalFEC() const { return has_internal_fec_; }
+  bool HasInternalFEC() const {
+    ReadLockScoped rl(codec_wrapper_lock_);
+    return has_internal_fec_;
+  }
 
   ///////////////////////////////////////////////////////////////////////////
   // int SetFEC();
@@ -624,7 +619,8 @@ class ACMGenericCodec {
   // See EncoderParam() for the description of function, input(s)/output(s)
   // and return value.
   //
-  int16_t EncoderParamsSafe(WebRtcACMCodecParams* enc_params);
+  int16_t EncoderParamsSafe(WebRtcACMCodecParams* enc_params)
+      SHARED_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   ///////////////////////////////////////////////////////////////////////////
   // See ResetEncoder() for the description of function, input(s)/output(s)
@@ -651,7 +647,8 @@ class ACMGenericCodec {
   // See DestructEncoder() for the description of function,
   // input(s)/output(s) and return value.
   //
-  virtual void DestructEncoderSafe() = 0;
+  virtual void DestructEncoderSafe()
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // See SetBitRate() for the description of function, input(s)/output(s)
@@ -659,7 +656,8 @@ class ACMGenericCodec {
   //
   // Any codec that can change the bit-rate has to implement this.
   //
-  virtual int16_t SetBitRateSafe(const int32_t bitrate_bps);
+  virtual int16_t SetBitRateSafe(const int32_t bitrate_bps)
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   ///////////////////////////////////////////////////////////////////////////
   // See GetEstimatedBandwidth() for the description of function,
@@ -707,7 +705,7 @@ class ACMGenericCodec {
   //   -1 if failed,
   //    0 if succeeded.
   //
-  int16_t CreateEncoder();
+  int16_t CreateEncoder() EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   ///////////////////////////////////////////////////////////////////////////
   // int16_t EnableVAD();
@@ -773,7 +771,8 @@ class ACMGenericCodec {
   //   otherwise the length of the bit-stream is returned.
   //
   virtual int16_t InternalEncode(uint8_t* bitstream,
-                                 int16_t* bitstream_len_byte) = 0;
+                                 int16_t* bitstream_len_byte)
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int16_t InternalInitEncoder()
@@ -794,7 +793,8 @@ class ACMGenericCodec {
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params) = 0;
+  virtual int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params)
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // void IncreaseNoMissedSamples()
@@ -805,7 +805,8 @@ class ACMGenericCodec {
   //   -num_samples        : the number of overwritten samples is incremented
   //                         by this value.
   //
-  void IncreaseNoMissedSamples(const int16_t num_samples);
+  void IncreaseNoMissedSamples(const int16_t num_samples)
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   ///////////////////////////////////////////////////////////////////////////
   // int16_t InternalCreateEncoder()
@@ -849,7 +850,8 @@ class ACMGenericCodec {
   //   -1 if failed,
   //    0 if succeeded.
   //
-  virtual int16_t InternalResetEncoder();
+  virtual int16_t InternalResetEncoder()
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   ///////////////////////////////////////////////////////////////////////////
   // int16_t ProcessFrameVADDTX()
@@ -900,41 +902,42 @@ class ACMGenericCodec {
 
   // &in_audio_[in_audio_ix_write_] always point to where new audio can be
   // written to
-  int16_t in_audio_ix_write_;
+  int16_t in_audio_ix_write_ GUARDED_BY(codec_wrapper_lock_);
 
   // &in_audio_[in_audio_ix_read_] points to where audio has to be read from
-  int16_t in_audio_ix_read_;
+  int16_t in_audio_ix_read_ GUARDED_BY(codec_wrapper_lock_);
 
-  int16_t in_timestamp_ix_write_;
+  int16_t in_timestamp_ix_write_ GUARDED_BY(codec_wrapper_lock_);
 
   // Where the audio is stored before encoding,
   // To save memory the following buffer can be allocated
   // dynamically for 80 ms depending on the sampling frequency
   // of the codec.
-  int16_t* in_audio_;
-  uint32_t* in_timestamp_;
+  int16_t* in_audio_ GUARDED_BY(codec_wrapper_lock_);
+  uint32_t* in_timestamp_ GUARDED_BY(codec_wrapper_lock_);
 
-  int16_t frame_len_smpl_;
-  uint16_t num_channels_;
+  int16_t frame_len_smpl_ GUARDED_BY(codec_wrapper_lock_);
+  uint16_t num_channels_ GUARDED_BY(codec_wrapper_lock_);
 
   // This will point to a static database of the supported codecs
-  int16_t codec_id_;
+  int16_t codec_id_ GUARDED_BY(codec_wrapper_lock_);
 
   // This will account for the number of samples  were not encoded
   // the case is rare, either samples are missed due to overwrite
   // at input buffer or due to encoding error
-  uint32_t num_missed_samples_;
+  uint32_t num_missed_samples_ GUARDED_BY(codec_wrapper_lock_);
 
   // True if the encoder instance created
-  bool encoder_exist_;
+  bool encoder_exist_ GUARDED_BY(codec_wrapper_lock_);
 
   // True if the encoder instance initialized
-  bool encoder_initialized_;
+  bool encoder_initialized_ GUARDED_BY(codec_wrapper_lock_);
 
-  const bool registered_in_neteq_;  // TODO(henrik.lundin) Remove?
+  const bool registered_in_neteq_
+      GUARDED_BY(codec_wrapper_lock_);  // TODO(henrik.lundin) Remove?
 
   // VAD/DTX
-  bool has_internal_dtx_;
+  bool has_internal_dtx_ GUARDED_BY(codec_wrapper_lock_);
   WebRtcVadInst* ptr_vad_inst_ GUARDED_BY(codec_wrapper_lock_);
   bool vad_enabled_ GUARDED_BY(codec_wrapper_lock_);
   ACMVADMode vad_mode_ GUARDED_BY(codec_wrapper_lock_);
@@ -947,13 +950,9 @@ class ACMGenericCodec {
   int16_t prev_frame_cng_ GUARDED_BY(codec_wrapper_lock_);
 
   // FEC.
-  bool has_internal_fec_;
-
-  WebRtcACMCodecParams encoder_params_;
+  bool has_internal_fec_ GUARDED_BY(codec_wrapper_lock_);
 
-  // Used as a global lock for all available decoders
-  // so that no decoder is used when NetEQ decodes.
-  RWLockWrapper* neteq_decode_lock_;
+  WebRtcACMCodecParams encoder_params_ GUARDED_BY(codec_wrapper_lock_);
 
   // Used to lock wrapper internal data
   // such as buffers and state variables.
diff --git a/modules/audio_coding/main/acm2/acm_ilbc.h b/modules/audio_coding/main/acm2/acm_ilbc.h
index fd6e8537..714c9004 100644
--- a/modules/audio_coding/main/acm2/acm_ilbc.h
+++ b/modules/audio_coding/main/acm2/acm_ilbc.h
@@ -29,14 +29,18 @@ class ACMILBC : public ACMGenericCodec {
   // for FEC
   ACMGenericCodec* CreateInstance(void);
 
-  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
 
  protected:
-  int16_t SetBitRateSafe(const int32_t rate);
+  int16_t SetBitRateSafe(const int32_t rate) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
-  void DestructEncoderSafe();
+  void DestructEncoderSafe() OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalCreateEncoder();
 
diff --git a/modules/audio_coding/main/acm2/acm_isac.cc b/modules/audio_coding/main/acm2/acm_isac.cc
index 9fbcdd4c..2adae861 100644
--- a/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/modules/audio_coding/main/acm2/acm_isac.cc
@@ -694,7 +694,10 @@ int32_t ACMISAC::ConfigISACBandwidthEstimator(
                  "Couldn't config iSAC BWE.");
     return -1;
   }
-  UpdateFrameLen();
+  {
+    WriteLockScoped wl(codec_wrapper_lock_);
+    UpdateFrameLen();
+  }
   CriticalSectionScoped lock(codec_inst_crit_sect_.get());
   ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_);
   return 0;
@@ -792,6 +795,7 @@ int ACMISAC::ErrorCode() {
 
 AudioDecoder* ACMISAC::Decoder(int codec_id) {
   // Create iSAC instance if it does not exist.
+  WriteLockScoped wl(codec_wrapper_lock_);
   if (!encoder_exist_) {
     CriticalSectionScoped lock(codec_inst_crit_sect_.get());
     assert(codec_inst_ptr_->inst == NULL);
diff --git a/modules/audio_coding/main/acm2/acm_isac.h b/modules/audio_coding/main/acm2/acm_isac.h
index 3249526f..31263c70 100644
--- a/modules/audio_coding/main/acm2/acm_isac.h
+++ b/modules/audio_coding/main/acm2/acm_isac.h
@@ -34,19 +34,23 @@ class ACMISAC : public ACMGenericCodec, AudioDecoder {
   explicit ACMISAC(int16_t codec_id);
   ~ACMISAC();
 
-  int16_t InternalInitDecoder(WebRtcACMCodecParams* codec_params);
+  int16_t InternalInitDecoder(WebRtcACMCodecParams* codec_params)
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   // Methods below are inherited from ACMGenericCodec.
   ACMGenericCodec* CreateInstance(void) OVERRIDE;
 
   int16_t InternalEncode(uint8_t* bitstream,
-                         int16_t* bitstream_len_byte) OVERRIDE;
+                         int16_t* bitstream_len_byte) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
-  int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params) OVERRIDE;
+  int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t UpdateDecoderSampFreq(int16_t codec_id) OVERRIDE;
 
-  int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz) OVERRIDE;
+  int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t EncoderSampFreq(uint16_t* samp_freq_hz) OVERRIDE;
 
@@ -95,12 +99,14 @@ class ACMISAC : public ACMGenericCodec, AudioDecoder {
                     int32_t rate,
                     bool is_red);
 
-  void UpdateFrameLen();
+  void UpdateFrameLen() EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   // Methods below are inherited from ACMGenericCodec.
-  void DestructEncoderSafe() OVERRIDE;
+  void DestructEncoderSafe() OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
-  int16_t SetBitRateSafe(const int32_t bit_rate) OVERRIDE;
+  int16_t SetBitRateSafe(const int32_t bit_rate) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int32_t GetEstimatedBandwidthSafe() OVERRIDE;
 
diff --git a/modules/audio_coding/main/acm2/acm_opus.h b/modules/audio_coding/main/acm2/acm_opus.h
index 07ce0721..b94adc47 100644
--- a/modules/audio_coding/main/acm2/acm_opus.h
+++ b/modules/audio_coding/main/acm2/acm_opus.h
@@ -28,7 +28,9 @@ class ACMOpus : public ACMGenericCodec {
 
   ACMGenericCodec* CreateInstance(void);
 
-  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
 
@@ -43,7 +45,8 @@ class ACMOpus : public ACMGenericCodec {
 
   void InternalDestructEncoderInst(void* ptr_inst);
 
-  int16_t SetBitRateSafe(const int32_t rate);
+  int16_t SetBitRateSafe(const int32_t rate) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   WebRtcOpusEncInst* encoder_inst_ptr_;
   uint16_t sample_freq_;
diff --git a/modules/audio_coding/main/acm2/acm_pcm16b.h b/modules/audio_coding/main/acm2/acm_pcm16b.h
index 23b8c121..de4330d5 100644
--- a/modules/audio_coding/main/acm2/acm_pcm16b.h
+++ b/modules/audio_coding/main/acm2/acm_pcm16b.h
@@ -25,12 +25,15 @@ class ACMPCM16B : public ACMGenericCodec {
   // For FEC.
   ACMGenericCodec* CreateInstance(void);
 
-  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
 
  protected:
-  void DestructEncoderSafe();
+  void DestructEncoderSafe() OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalCreateEncoder();
 
diff --git a/modules/audio_coding/main/acm2/acm_pcma.h b/modules/audio_coding/main/acm2/acm_pcma.h
index 2da873cb..a2d8874a 100644
--- a/modules/audio_coding/main/acm2/acm_pcma.h
+++ b/modules/audio_coding/main/acm2/acm_pcma.h
@@ -25,7 +25,9 @@ class ACMPCMA : public ACMGenericCodec {
   // For FEC.
   ACMGenericCodec* CreateInstance(void);
 
-  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
 
diff --git a/modules/audio_coding/main/acm2/acm_pcmu.h b/modules/audio_coding/main/acm2/acm_pcmu.h
index 18d8279d..7aa83b75 100644
--- a/modules/audio_coding/main/acm2/acm_pcmu.h
+++ b/modules/audio_coding/main/acm2/acm_pcmu.h
@@ -25,12 +25,15 @@ class ACMPCMU : public ACMGenericCodec {
   // For FEC.
   ACMGenericCodec* CreateInstance(void);
 
-  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
+  int16_t InternalEncode(uint8_t* bitstream,
+                         int16_t* bitstream_len_byte) OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
 
  protected:
-  void DestructEncoderSafe();
+  void DestructEncoderSafe() OVERRIDE
+      EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
 
   int16_t InternalCreateEncoder();
 
diff --git a/modules/audio_coding/main/acm2/acm_receiver.cc b/modules/audio_coding/main/acm2/acm_receiver.cc
index cb7c4184..f40250f2 100644
--- a/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -26,7 +26,6 @@
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
@@ -126,7 +125,6 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
       nack_(),
       nack_enabled_(false),
       neteq_(NetEq::Create(config.neteq_config)),
-      decode_lock_(RWLockWrapper::CreateRWLock()),
       vad_enabled_(true),
       clock_(config.clock),
       av_sync_(false),
@@ -149,7 +147,6 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
 
 AcmReceiver::~AcmReceiver() {
   delete neteq_;
-  delete decode_lock_;
 }
 
 int AcmReceiver::SetMinimumDelay(int delay_ms) {
@@ -331,22 +328,18 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
     }
   }  // |crit_sect_| is released.
 
-  {
-    WriteLockScoped lock_codecs(*decode_lock_);  // Lock to prevent an encoding.
-
-    // 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 |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) {
-      LOG_FERR1(LS_ERROR, "AcmReceiver::InsertPacket", header->payloadType) <<
-          " Failed to insert packet";
-      return -1;
-    }
+  if (neteq_->InsertPacket(rtp_header, incoming_payload, length_payload,
+                           receive_timestamp) < 0) {
+    LOG_FERR1(LS_ERROR, "AcmReceiver::InsertPacket", header->payloadType) <<
+        " Failed to insert packet";
+    return -1;
   }
   return 0;
 }
@@ -384,24 +377,20 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
     }
   }
 
-  {
-    WriteLockScoped lock_codecs(*decode_lock_);  // Lock to prevent an encoding.
-
-    // 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;
-    }
+  // 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;
+  }
 
-    if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
-                         ptr_audio_buffer,
-                         &samples_per_channel,
-                         &num_channels, &type) != NetEq::kOK) {
-      LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
-      return -1;
-    }
+  if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
+                       ptr_audio_buffer,
+                       &samples_per_channel,
+                       &num_channels, &type) != NetEq::kOK) {
+    LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
+    return -1;
   }
 
   // Accessing members, take the lock.
diff --git a/modules/audio_coding/main/acm2/acm_receiver.h b/modules/audio_coding/main/acm2/acm_receiver.h
index b6898f73..dd6de528 100644
--- a/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/modules/audio_coding/main/acm2/acm_receiver.h
@@ -30,7 +30,6 @@ namespace webrtc {
 
 struct CodecInst;
 class CriticalSectionWrapper;
-class RWLockWrapper;
 class NetEq;
 
 namespace acm2 {
@@ -210,13 +209,6 @@ class AcmReceiver {
   bool vad_enabled() const { return vad_enabled_; }
 
   //
-  // Get the decode lock used to protect decoder instances while decoding.
-  //
-  // Return value             : Pointer to the decode lock.
-  //
-  RWLockWrapper* DecodeLock() const { return decode_lock_; }
-
-  //
   // Flushes the NetEq packet and speech buffers.
   //
   void FlushBuffers();
@@ -354,7 +346,6 @@ class AcmReceiver {
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;
   Decoder decoders_[ACMCodecDB::kMaxNumCodecs];
-  RWLockWrapper* decode_lock_;
   bool vad_enabled_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
 
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index a07e8543..f2410b7d 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -749,7 +749,6 @@ ACMGenericCodec* AudioCodingModuleImpl::CreateCodec(const CodecInst& codec) {
     return my_codec;
   }
   my_codec->SetUniqueID(id_);
-  my_codec->SetNetEqDecodeLock(receiver_.DecodeLock());
 
   return my_codec;
 }
@@ -1203,6 +1202,7 @@ int AudioCodingModuleImpl::SendBitrate() const {
 // Set available bandwidth, inform the encoder about the estimated bandwidth
 // received from the remote party.
 int AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(int bw) {
+  CriticalSectionScoped lock(acm_crit_sect_);
   return codecs_[current_send_codec_idx_]->SetEstimatedBandwidth(bw);
 }
 
@@ -1452,6 +1452,7 @@ int AudioCodingModuleImpl::SetREDStatus(
 //
 
 bool AudioCodingModuleImpl::CodecFEC() const {
+  CriticalSectionScoped lock(acm_crit_sect_);
   return codec_fec_enabled_;
 }
 
@@ -1476,6 +1477,7 @@ int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
 }
 
 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
+  CriticalSectionScoped lock(acm_crit_sect_);
   if (HaveValidEncoder("SetPacketLossRate") &&
       codecs_[current_send_codec_idx_]->SetPacketLossRate(loss_rate) < 0) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
@@ -1950,6 +1952,7 @@ int AudioCodingModuleImpl::REDPayloadISAC(int isac_rate,
                                           int isac_bw_estimate,
                                           uint8_t* payload,
                                           int16_t* length_bytes) {
+  CriticalSectionScoped lock(acm_crit_sect_);
   if (!HaveValidEncoder("EncodeData")) {
     return -1;
   }
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index e54202bf..02290da0 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -214,6 +214,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
 
   // GET RED payload for iSAC. The method id called when 'this' ACM is
   // the default ACM.
+  // TODO(henrik.lundin) Not used. Remove?
   int REDPayloadISAC(int isac_rate,
                      int isac_bw_estimate,
                      uint8_t* payload,
@@ -248,7 +249,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
 
   int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
-  bool HaveValidEncoder(const char* caller_name) const;
+  bool HaveValidEncoder(const char* caller_name) const
+      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   // Set VAD/DTX status. This function does not acquire a lock, and it is
   // created to be called only from inside a critical section.
@@ -303,7 +305,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   // codec owns the decoder-instance. For such codecs |*decoder| should be a
   // valid pointer, otherwise it will be NULL.
   int GetAudioDecoder(const CodecInst& codec, int codec_id,
-                      int mirror_id, AudioDecoder** decoder);
+                      int mirror_id, AudioDecoder** decoder)
+      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   CriticalSectionWrapper* acm_crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
@@ -320,13 +323,14 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   bool vad_enabled_ GUARDED_BY(acm_crit_sect_);
   bool dtx_enabled_ GUARDED_BY(acm_crit_sect_);
   ACMVADMode vad_mode_ GUARDED_BY(acm_crit_sect_);
-  ACMGenericCodec* codecs_[ACMCodecDB::kMaxNumCodecs];
-  int mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs];
+  ACMGenericCodec* codecs_[ACMCodecDB::kMaxNumCodecs]
+      GUARDED_BY(acm_crit_sect_);
+  int mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs] GUARDED_BY(acm_crit_sect_);
   bool stereo_send_ GUARDED_BY(acm_crit_sect_);
-  int current_send_codec_idx_;
-  bool send_codec_registered_;
+  int current_send_codec_idx_ GUARDED_BY(acm_crit_sect_);
+  bool send_codec_registered_ GUARDED_BY(acm_crit_sect_);
   ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
-  AcmReceiver receiver_;
+  AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
 
   // RED.
   bool is_first_red_ GUARDED_BY(acm_crit_sect_);
@@ -345,7 +349,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   uint32_t last_red_timestamp_ GUARDED_BY(acm_crit_sect_);
 
   // Codec internal FEC
-  bool codec_fec_enabled_;
+  bool codec_fec_enabled_ 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_);
diff --git a/modules/audio_coding/main/interface/audio_coding_module.h b/modules/audio_coding/main/interface/audio_coding_module.h
index cb0953aa..48270047 100644
--- a/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/modules/audio_coding/main/interface/audio_coding_module.h
@@ -330,6 +330,7 @@ class AudioCodingModule: public Module {
   //   -1 if error occurred in setting the bandwidth,
   //    0 bandwidth is set successfully.
   //
+  // TODO(henrik.lundin) Unused. Remove?
   virtual int32_t SetReceivedEstimatedBandwidth(
       const int32_t bw) = 0;
 
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 3c5f6b0b..3e3540a8 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -575,7 +575,6 @@ size_t ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file,
             " not supported or unknown." << std::endl;
         webrtc::Trace::ReturnTrace();
         exit(1);
-        assert(false);
     }
   }
   return payload_len;
diff --git a/modules/audio_conference_mixer/BUILD.gn b/modules/audio_conference_mixer/BUILD.gn
new file mode 100644
index 00000000..a27bb84c
--- /dev/null
+++ b/modules/audio_conference_mixer/BUILD.gn
@@ -0,0 +1,41 @@
+# Copyright (c) 2014 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.
+
+config("internal_config") {
+  visibility = ":*"  # Only targets in this file can depend on this.
+  include_dirs = [
+    "interface",
+    "../interface",
+  ]
+}
+
+source_set("audio_conference_mixer") {
+  sources = [
+    "interface/audio_conference_mixer.h",
+    "interface/audio_conference_mixer_defines.h",
+    "source/audio_conference_mixer_impl.cc",
+    "source/audio_conference_mixer_impl.h",
+    "source/audio_frame_manipulator.cc",
+    "source/audio_frame_manipulator.h",
+    "source/level_indicator.cc",
+    "source/level_indicator.h",
+    "source/memory_pool.h",
+    "source/memory_pool_posix.h",
+    "source/memory_pool_win.h",
+    "source/time_scheduler.cc",
+    "source/time_scheduler.h",
+  ]
+
+  direct_dependent_configs = [ ":internal_config" ]
+
+  deps = [
+    "../../system_wrappers",
+    "../audio_processing",
+    "../utility",
+  ]
+}
diff --git a/modules/audio_conference_mixer/OWNERS b/modules/audio_conference_mixer/OWNERS
index 7dc791ef..7d4cc610 100644
--- a/modules/audio_conference_mixer/OWNERS
+++ b/modules/audio_conference_mixer/OWNERS
@@ -1,3 +1,5 @@
 henrike@webrtc.org
 pwestin@webrtc.org
-andrew@webrtc.org
\ No newline at end of file
+andrew@webrtc.org
+
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
new file mode 100644
index 00000000..d6481ad5
--- /dev/null
+++ b/modules/audio_device/BUILD.gn
@@ -0,0 +1,13 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("audio_device") {
+  # TODO(henrike): Implement.
+}
diff --git a/modules/audio_device/OWNERS b/modules/audio_device/OWNERS
index 7bb3cd52..1fe45940 100644
--- a/modules/audio_device/OWNERS
+++ b/modules/audio_device/OWNERS
@@ -9,3 +9,5 @@ per-file *.isolate=kjellander@webrtc.org
 # 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/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
new file mode 100644
index 00000000..a727d54b
--- /dev/null
+++ b/modules/audio_processing/BUILD.gn
@@ -0,0 +1,13 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("audio_processing") {
+  # TODO(andrew): Implement.
+}
diff --git a/modules/audio_processing/OWNERS b/modules/audio_processing/OWNERS
index 7c1f7881..41a82af2 100644
--- a/modules/audio_processing/OWNERS
+++ b/modules/audio_processing/OWNERS
@@ -6,3 +6,5 @@ bjornv@webrtc.org
 # 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/modules/audio_processing/aec/aec_core_neon.c b/modules/audio_processing/aec/aec_core_neon.c
index cec0a7e3..5cce4897 100644
--- a/modules/audio_processing/aec/aec_core_neon.c
+++ b/modules/audio_processing/aec/aec_core_neon.c
@@ -30,6 +30,104 @@ __inline static float MulRe(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bRe - aIm * bIm;
 }
 
+static float32x4_t vdivq_f32(float32x4_t a, float32x4_t b) {
+  int i;
+  float32x4_t x = vrecpeq_f32(b);
+  // from arm documentation
+  // The Newton-Raphson iteration:
+  //     x[n+1] = x[n] * (2 - d * x[n])
+  // converges to (1/d) if x0 is the result of VRECPE applied to d.
+  //
+  // Note: The precision did not improve after 2 iterations.
+  for (i = 0; i < 2; i++) {
+    x = vmulq_f32(vrecpsq_f32(b, x), x);
+  }
+  // a/b = a*(1/b)
+  return vmulq_f32(a, x);
+}
+
+static float32x4_t vsqrtq_f32(float32x4_t s) {
+  int i;
+  float32x4_t x = vrsqrteq_f32(s);
+
+  // Code to handle sqrt(0).
+  // If the input to sqrtf() is zero, a zero will be returned.
+  // If the input to vrsqrteq_f32() is zero, positive infinity is returned.
+  const uint32x4_t vec_p_inf = vdupq_n_u32(0x7F800000);
+  // check for divide by zero
+  const uint32x4_t div_by_zero = vceqq_u32(vec_p_inf, vreinterpretq_u32_f32(x));
+  // zero out the positive infinity results
+  x = vreinterpretq_f32_u32(vandq_u32(vmvnq_u32(div_by_zero),
+                                      vreinterpretq_u32_f32(x)));
+  // from arm documentation
+  // The Newton-Raphson iteration:
+  //     x[n+1] = x[n] * (3 - d * (x[n] * x[n])) / 2)
+  // converges to (1/√d) if x0 is the result of VRSQRTE applied to d.
+  //
+  // Note: The precision did not improve after 2 iterations.
+  for (i = 0; i < 2; i++) {
+    x = vmulq_f32(vrsqrtsq_f32(vmulq_f32(x, x), s), x);
+  }
+  // sqrt(s) = s * 1/sqrt(s)
+  return vmulq_f32(s, x);;
+}
+
+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;
+  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 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);
+    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);
+    const float32x4_t ef_sum2 = vmlaq_f32(ef_re2, ef_im, ef_im);
+    const float32x4_t absEf = vsqrtq_f32(ef_sum2);
+    const uint32x4_t bigger = vcgtq_f32(absEf, kThresh);
+    const float32x4_t absEfPlus = vaddq_f32(absEf, k1e_10f);
+    const float32x4_t absEfInv = vdivq_f32(kThresh, absEfPlus);
+    uint32x4_t ef_re_if = vreinterpretq_u32_f32(vmulq_f32(ef_re, absEfInv));
+    uint32x4_t ef_im_if = vreinterpretq_u32_f32(vmulq_f32(ef_im, absEfInv));
+    uint32x4_t ef_re_u32 = vandq_u32(vmvnq_u32(bigger),
+                                     vreinterpretq_u32_f32(ef_re));
+    uint32x4_t ef_im_u32 = vandq_u32(vmvnq_u32(bigger),
+                                     vreinterpretq_u32_f32(ef_im));
+    ef_re_if = vandq_u32(bigger, ef_re_if);
+    ef_im_if = vandq_u32(bigger, ef_im_if);
+    ef_re_u32 = vorrq_u32(ef_re_u32, ef_re_if);
+    ef_im_u32 = vorrq_u32(ef_im_u32, ef_im_if);
+    ef_re = vmulq_f32(vreinterpretq_f32_u32(ef_re_u32), kMu);
+    ef_im = vmulq_f32(vreinterpretq_f32_u32(ef_im_u32), kMu);
+    vst1q_f32(&ef[0][i], ef_re);
+    vst1q_f32(&ef[1][i], ef_im);
+  }
+  // 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);
+    abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+
+    if (abs_ef > error_threshold) {
+      abs_ef = error_threshold / (abs_ef + 1e-10f);
+      ef[0][i] *= abs_ef;
+      ef[1][i] *= abs_ef;
+    }
+
+    // Stepsize factor
+    ef[0][i] *= mu;
+    ef[1][i] *= mu;
+  }
+}
+
 static void FilterAdaptationNEON(AecCore* aec,
                                  float* fft,
                                  float ef[2][PART_LEN1]) {
@@ -298,6 +396,7 @@ static void OverdriveAndSuppressNEON(AecCore* aec,
 }
 
 void WebRtcAec_InitAec_neon(void) {
+  WebRtcAec_ScaleErrorSignal = ScaleErrorSignalNEON;
   WebRtcAec_FilterAdaptation = FilterAdaptationNEON;
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressNEON;
 }
diff --git a/modules/audio_processing/aec/aec_core_sse2.c b/modules/audio_processing/aec/aec_core_sse2.c
index 8d4afdba..1489d26e 100644
--- a/modules/audio_processing/aec/aec_core_sse2.c
+++ b/modules/audio_processing/aec/aec_core_sse2.c
@@ -354,8 +354,8 @@ static __m128 mm_pow_ps(__m128 a, __m128 b) {
   return a_exp_b;
 }
 
-extern const float WebRtcAec_weightCurve[65];
-extern const float WebRtcAec_overDriveCurve[65];
+extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_weightCurve[65];
+extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_overDriveCurve[65];
 
 static void OverdriveAndSuppressSSE2(AecCore* aec,
                                      float hNl[PART_LEN1],
diff --git a/modules/audio_processing/aec/system_delay_unittest.cc b/modules/audio_processing/aec/system_delay_unittest.cc
index a13d4762..5fbc5600 100644
--- a/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/modules/audio_processing/aec/system_delay_unittest.cc
@@ -248,7 +248,8 @@ TEST_F(SystemDelayTest, CorrectDelayAfterUnstableStartup) {
   }
 }
 
-TEST_F(SystemDelayTest, CorrectDelayAfterStableBufferBuildUp) {
+TEST_F(SystemDelayTest,
+       DISABLED_ON_ANDROID(CorrectDelayAfterStableBufferBuildUp)) {
   // In this test we start by establishing the device buffer size during stable
   // conditions, but with an empty internal far-end buffer. Once that is done we
   // verify that the system delay is increased correctly until we have reach an
@@ -332,7 +333,7 @@ TEST_F(SystemDelayTest, CorrectDelayWhenBufferUnderrun) {
   }
 }
 
-TEST_F(SystemDelayTest, CorrectDelayDuringDrift) {
+TEST_F(SystemDelayTest, DISABLED_ON_ANDROID(CorrectDelayDuringDrift)) {
   // This drift test should verify that the system delay is never exceeding the
   // device buffer. The drift is simulated by decreasing the reported device
   // buffer size by 1 ms every 100 ms. If the device buffer size goes below 30
@@ -366,7 +367,7 @@ TEST_F(SystemDelayTest, CorrectDelayDuringDrift) {
   }
 }
 
-TEST_F(SystemDelayTest, ShouldRecoverAfterGlitch) {
+TEST_F(SystemDelayTest, DISABLED_ON_ANDROID(ShouldRecoverAfterGlitch)) {
   // This glitch test should verify that the system delay recovers if there is
   // a glitch in data. The data glitch is constructed as 200 ms of buffering
   // after which the stable procedure continues. The glitch is never reported by
diff --git a/modules/bitrate_controller/BUILD.gn b/modules/bitrate_controller/BUILD.gn
new file mode 100644
index 00000000..c3ac28ce
--- /dev/null
+++ b/modules/bitrate_controller/BUILD.gn
@@ -0,0 +1,28 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("bitrate_controller") {
+  sources = [
+    "bitrate_controller_impl.cc",
+    "bitrate_controller_impl.h",
+    "include/bitrate_controller.h",
+    "send_side_bandwidth_estimation.cc",
+    "send_side_bandwidth_estimation.h",
+  ]
+
+  if (is_win) {
+    cflags = [
+      # TODO(jschuh): Bug 1348: fix this warning.
+      "/wd4267"  # size_t to int truncations
+    ]
+  }
+
+  deps = [ "../../system_wrappers" ]
+}
diff --git a/modules/bitrate_controller/OWNERS b/modules/bitrate_controller/OWNERS
index cce3a26b..9420ba2e 100644
--- a/modules/bitrate_controller/OWNERS
+++ b/modules/bitrate_controller/OWNERS
@@ -8,3 +8,5 @@ asapersson@webrtc.org
 # 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/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
new file mode 100644
index 00000000..0ccb18a1
--- /dev/null
+++ b/modules/desktop_capture/BUILD.gn
@@ -0,0 +1,19 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("desktop_capture") {
+  # TODO(jiayl): Implement.
+}
+
+if (!is_ios && (cpu_arch == "x86" || cpu_arch == "x64")) {
+  source_set("desktop_capture_differ_sse2") {
+    # TODO(jiayl): Implement.
+  }
+}
diff --git a/modules/desktop_capture/OWNERS b/modules/desktop_capture/OWNERS
index e85861b8..4c0340d6 100644
--- a/modules/desktop_capture/OWNERS
+++ b/modules/desktop_capture/OWNERS
@@ -6,3 +6,5 @@ wez@chromium.org
 # 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/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/modules/desktop_capture/win/screen_capturer_win_gdi.cc
index 9cb3681f..dc10eceb 100644
--- a/modules/desktop_capture/win/screen_capturer_win_gdi.cc
+++ b/modules/desktop_capture/win/screen_capturer_win_gdi.cc
@@ -133,7 +133,8 @@ void ScreenCapturerWinGdi::Capture(const DesktopRegion& region) {
   callback_->OnCaptureCompleted(frame);
 
   // Check for cursor shape update.
-  CaptureCursor();
+  if (mouse_shape_observer_)
+    CaptureCursor();
 }
 
 void ScreenCapturerWinGdi::SetMouseShapeObserver(
@@ -279,6 +280,8 @@ bool ScreenCapturerWinGdi::CaptureImage() {
 }
 
 void ScreenCapturerWinGdi::CaptureCursor() {
+  assert(mouse_shape_observer_);
+
   CURSORINFO cursor_info;
   cursor_info.cbSize = sizeof(CURSORINFO);
   if (!GetCursorInfo(&cursor_info)) {
@@ -317,8 +320,7 @@ void ScreenCapturerWinGdi::CaptureCursor() {
   // Record the last cursor image that we sent to the client.
   last_cursor_ = *cursor;
 
-  if (mouse_shape_observer_)
-    mouse_shape_observer_->OnCursorShapeChanged(cursor.release());
+  mouse_shape_observer_->OnCursorShapeChanged(cursor.release());
 }
 
 }  // namespace webrtc
diff --git a/modules/media_file/BUILD.gn b/modules/media_file/BUILD.gn
new file mode 100644
index 00000000..971a6fa2
--- /dev/null
+++ b/modules/media_file/BUILD.gn
@@ -0,0 +1,41 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+config("internal_config") {
+  visibility = ":*"  # Only targets in this file can depend on this.
+  include_dirs = [
+    "interface",
+    "../interface",
+  ]
+}
+
+source_set("media_file") {
+  sources = [
+    "interface/media_file.h",
+    "interface/media_file_defines.h",
+    "source/avi_file.cc",
+    "source/avi_file.h",
+    "source/media_file_impl.cc",
+    "source/media_file_impl.h",
+    "source/media_file_utility.cc",
+    "source/media_file_utility.h",
+  ]
+
+  if (is_win) {
+    cflags = [
+      # TODO(jschuh): Bug 1348: fix this warning.
+      "/wd4267",  # size_t to int truncations
+    ]
+  }
+
+  direct_dependent_configs = [ ":internal_config" ]
+
+  deps = [ "../../system_wrappers" ]
+}
diff --git a/modules/media_file/OWNERS b/modules/media_file/OWNERS
index 2cc47e48..3387ee9e 100644
--- a/modules/media_file/OWNERS
+++ b/modules/media_file/OWNERS
@@ -1,4 +1,6 @@
 pwestin@webrtc.org

 mflodman@webrtc.org

 perkj@webrtc.org

-niklas.enbom@webrtc.org
\ No newline at end of file
+niklas.enbom@webrtc.org

+

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

diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
new file mode 100644
index 00000000..d3eceb35
--- /dev/null
+++ b/modules/pacing/BUILD.gn
@@ -0,0 +1,16 @@
+# Copyright (c) 2014 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.
+
+source_set("pacing") {
+  sources = [
+    "include/paced_sender.h",
+    "paced_sender.cc",
+  ]
+
+  deps = [ "../../system_wrappers" ]
+}
diff --git a/modules/pacing/OWNERS b/modules/pacing/OWNERS
index 1426abc6..4486a81e 100644
--- a/modules/pacing/OWNERS
+++ b/modules/pacing/OWNERS
@@ -7,3 +7,5 @@ asapersson@webrtc.org
 # 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/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
new file mode 100644
index 00000000..7ee4c8df
--- /dev/null
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -0,0 +1,17 @@
+# Copyright (c) 2014 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.
+
+source_set("remote_bitrate_estimator") {
+  sources = [
+    "overuse_detector.cc",
+    "overuse_detector.h",
+    "remote_bitrate_estimator_single_stream.cc",
+    "remote_rate_control.cc",
+    "remote_rate_control.h",
+  ]
+}
diff --git a/modules/remote_bitrate_estimator/OWNERS b/modules/remote_bitrate_estimator/OWNERS
index eab2b8e0..49d19429 100644
--- a/modules/remote_bitrate_estimator/OWNERS
+++ b/modules/remote_bitrate_estimator/OWNERS
@@ -3,7 +3,10 @@ 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/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
new file mode 100644
index 00000000..e9a2e8d9
--- /dev/null
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -0,0 +1,104 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("rtp_rtcp") {
+  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",
+    "source/bitrate.cc",
+    "source/bitrate.h",
+    "source/byte_io.h",
+    "source/fec_receiver_impl.cc",
+    "source/fec_receiver_impl.h",
+    "source/receive_statistics_impl.cc",
+    "source/receive_statistics_impl.h",
+    "source/remote_ntp_time_estimator.cc",
+    "source/rtp_header_parser.cc",
+    "source/rtp_rtcp_config.h",
+    "source/rtp_rtcp_impl.cc",
+    "source/rtp_rtcp_impl.h",
+    "source/rtcp_packet.cc",
+    "source/rtcp_packet.h",
+    "source/rtcp_receiver.cc",
+    "source/rtcp_receiver.h",
+    "source/rtcp_receiver_help.cc",
+    "source/rtcp_receiver_help.h",
+    "source/rtcp_sender.cc",
+    "source/rtcp_sender.h",
+    "source/rtcp_utility.cc",
+    "source/rtcp_utility.h",
+    "source/rtp_header_extension.cc",
+    "source/rtp_header_extension.h",
+    "source/rtp_receiver_impl.cc",
+    "source/rtp_receiver_impl.h",
+    "source/rtp_sender.cc",
+    "source/rtp_sender.h",
+    "source/rtp_utility.cc",
+    "source/rtp_utility.h",
+    "source/ssrc_database.cc",
+    "source/ssrc_database.h",
+    "source/tmmbr_help.cc",
+    "source/tmmbr_help.h",
+    # Audio Files
+    "source/dtmf_queue.cc",
+    "source/dtmf_queue.h",
+    "source/rtp_receiver_audio.cc",
+    "source/rtp_receiver_audio.h",
+    "source/rtp_sender_audio.cc",
+    "source/rtp_sender_audio.h",
+    # Video Files
+    "source/fec_private_tables_random.h",
+    "source/fec_private_tables_bursty.h",
+    "source/forward_error_correction.cc",
+    "source/forward_error_correction.h",
+    "source/forward_error_correction_internal.cc",
+    "source/forward_error_correction_internal.h",
+    "source/producer_fec.cc",
+    "source/producer_fec.h",
+    "source/rtp_packet_history.cc",
+    "source/rtp_packet_history.h",
+    "source/rtp_payload_registry.cc",
+    "source/rtp_receiver_strategy.cc",
+    "source/rtp_receiver_strategy.h",
+    "source/rtp_receiver_video.cc",
+    "source/rtp_receiver_video.h",
+    "source/rtp_sender_video.cc",
+    "source/rtp_sender_video.h",
+    "source/video_codec_information.h",
+    "source/rtp_format_vp8.cc",
+    "source/rtp_format_vp8.h",
+    "source/rtp_format_video_generic.h",
+    "source/vp8_partition_aggregator.cc",
+    "source/vp8_partition_aggregator.h",
+    # Mocks
+    "mocks/mock_rtp_rtcp.h",
+    "source/mock/mock_rtp_payload_strategy.h",
+  ]
+
+  deps = [
+    "../../system_wrappers",
+    "../pacing",
+    "../remote_bitrate_estimator",
+  ]
+
+  if (is_win) {
+    cflags = [
+      # TODO(jschuh): Bug 1348: fix this warning.
+      "/wd4267",  # size_t to int truncations
+    ]
+  }
+}
diff --git a/modules/rtp_rtcp/OWNERS b/modules/rtp_rtcp/OWNERS
index c9685642..cafe4c19 100644
--- a/modules/rtp_rtcp/OWNERS
+++ b/modules/rtp_rtcp/OWNERS
@@ -2,4 +2,6 @@ pwestin@webrtc.org
 stefan@webrtc.org

 henrik.lundin@webrtc.org

 mflodman@webrtc.org

-asapersson@webrtc.org
\ No newline at end of file
+asapersson@webrtc.org

+

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

diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
new file mode 100644
index 00000000..82512046
--- /dev/null
+++ b/modules/utility/BUILD.gn
@@ -0,0 +1,48 @@
+# Copyright (c) 2014 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.
+
+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/process_thread.h",
+    "interface/rtp_dump.h",
+    "source/audio_frame_operations.cc",
+    "source/coder.cc",
+    "source/coder.h",
+    "source/file_player_impl.cc",
+    "source/file_player_impl.h",
+    "source/file_recorder_impl.cc",
+    "source/file_recorder_impl.h",
+    "source/helpers_android.cc",
+    "source/process_thread_impl.cc",
+    "source/process_thread_impl.h",
+    "source/rtp_dump_impl.cc",
+    "source/rtp_dump_impl.h",
+  ]
+
+  deps = [
+    "../../common_audio",
+    "../../system_wrappers",
+    "../audio_coding",
+    "../media_file",
+  ]
+  if (enable_video) {
+    sources += [
+      "source/frame_scaler.cc",
+      "source/video_coder.cc",
+      "source/video_frames_queue.cc",
+    ]
+
+    deps += [ "../video_coding" ]
+  }
+}
diff --git a/modules/utility/OWNERS b/modules/utility/OWNERS
index 674c7380..dbdb53e3 100644
--- a/modules/utility/OWNERS
+++ b/modules/utility/OWNERS
@@ -1,4 +1,6 @@
 henrike@webrtc.org
 pwestin@webrtc.org
 asapersson@webrtc.org
-perkj@webrtc.org
\ No newline at end of file
+perkj@webrtc.org
+
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
new file mode 100644
index 00000000..fd3fe574
--- /dev/null
+++ b/modules/video_capture/BUILD.gn
@@ -0,0 +1,13 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("video_capture") {
+  # TODO(glaznev): Implement.
+}
diff --git a/modules/video_capture/OWNERS b/modules/video_capture/OWNERS
index fdc2a3ff..ba897650 100644
--- a/modules/video_capture/OWNERS
+++ b/modules/video_capture/OWNERS
@@ -11,3 +11,5 @@ per-file *.isolate=kjellander@webrtc.org
 # 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/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java b/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java
index c8abdbe3..2f571d0c 100644
--- a/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java
+++ b/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java
@@ -11,20 +11,18 @@
 package org.webrtc.videoengine;
 
 import java.io.IOException;
-import java.util.Locale;
 import java.util.concurrent.Exchanger;
-import java.util.concurrent.locks.ReentrantLock;
 
 import android.content.Context;
 import android.graphics.ImageFormat;
-import android.graphics.PixelFormat;
-import android.graphics.Rect;
 import android.graphics.SurfaceTexture;
-import android.graphics.YuvImage;
 import android.hardware.Camera.PreviewCallback;
 import android.hardware.Camera;
+import android.opengl.GLES11Ext;
+import android.opengl.GLES20;
 import android.os.Handler;
 import android.os.Looper;
+import android.os.SystemClock;
 import android.util.Log;
 import android.view.OrientationEventListener;
 import android.view.SurfaceHolder.Callback;
@@ -50,11 +48,15 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
   private final Camera.CameraInfo info;
   private final OrientationEventListener orientationListener;
   private final long native_capturer;  // |VideoCaptureAndroid*| in C++.
-  private SurfaceTexture dummySurfaceTexture;
+  private SurfaceTexture cameraSurfaceTexture;
+  private int[] cameraGlTextures = null;
   // Arbitrary queue depth.  Higher number means more memory allocated & held,
   // lower number means more sensitivity to processing time in the client (and
   // potentially stalling the capturer if it runs out of buffers to write to).
   private final int numCaptureBuffers = 3;
+  private double averageDurationMs;
+  private long lastCaptureTimeMs;
+  private int frameCount;
 
   // Requests future capturers to send their frames to |localPreview| directly.
   public static void setLocalPreview(SurfaceHolder localPreview) {
@@ -114,6 +116,8 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
   private synchronized boolean startCapture(
       final int width, final int height,
       final int min_mfps, final int max_mfps) {
+    Log.d(TAG, "startCapture: " + width + "x" + height + "@" +
+        min_mfps + ":" + max_mfps);
     if (cameraThread != null || cameraThreadHandler != null) {
       throw new RuntimeException("Camera thread already started!");
     }
@@ -121,7 +125,6 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
     cameraThread = new CameraThread(handlerExchanger);
     cameraThread.start();
     cameraThreadHandler = exchange(handlerExchanger, null);
-    orientationListener.enable();
 
     final Exchanger<Boolean> result = new Exchanger<Boolean>();
     cameraThreadHandler.post(new Runnable() {
@@ -129,14 +132,14 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
           startCaptureOnCameraThread(width, height, min_mfps, max_mfps, result);
         }
       });
-    return exchange(result, false);  // |false| is a dummy value here.
+    boolean startResult = exchange(result, false); // |false| is a dummy value.
+    orientationListener.enable();
+    return startResult;
   }
 
   private void startCaptureOnCameraThread(
       int width, int height, int min_mfps, int max_mfps,
       Exchanger<Boolean> result) {
-    Log.d(TAG, "startCapture: " + width + "x" + height + "@" +
-        min_mfps + ":" + max_mfps);
     Throwable error = null;
     try {
       camera = Camera.open(id);
@@ -150,13 +153,27 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
       } else {
         // No local renderer (we only care about onPreviewFrame() buffers, not a
         // directly-displayed UI element).  Camera won't capture without
-        // setPreview{Texture,Display}, so we create a dummy SurfaceTexture and
-        // hand it over to Camera, but never listen for frame-ready callbacks,
+        // setPreview{Texture,Display}, so we create a SurfaceTexture and hand
+        // it over to Camera, but never listen for frame-ready callbacks,
         // and never call updateTexImage on it.
         try {
-          // "42" because http://goo.gl/KaEn8
-          dummySurfaceTexture = new SurfaceTexture(42);
-          camera.setPreviewTexture(dummySurfaceTexture);
+          cameraGlTextures = new int[1];
+          // Generate one texture pointer and bind it as an external texture.
+          GLES20.glGenTextures(1, cameraGlTextures, 0);
+          GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              cameraGlTextures[0]);
+          GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_MIN_FILTER, GLES20.GL_LINEAR);
+          GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_MAG_FILTER, GLES20.GL_LINEAR);
+          GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_WRAP_S, GLES20.GL_CLAMP_TO_EDGE);
+          GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_WRAP_T, GLES20.GL_CLAMP_TO_EDGE);
+
+          cameraSurfaceTexture = new SurfaceTexture(cameraGlTextures[0]);
+          cameraSurfaceTexture.setOnFrameAvailableListener(null);
+          camera.setPreviewTexture(cameraSurfaceTexture);
         } catch (IOException e) {
           throw new RuntimeException(e);
         }
@@ -178,6 +195,8 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
         camera.addCallbackBuffer(new byte[bufSize]);
       }
       camera.setPreviewCallbackWithBuffer(this);
+      frameCount = 0;
+      averageDurationMs = 1000 / max_mfps;
       camera.startPreview();
       exchange(result, true);
       return;
@@ -198,6 +217,8 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
 
   // Called by native code.  Returns true when camera is known to be stopped.
   private synchronized boolean stopCapture() {
+    Log.d(TAG, "stopCapture");
+    orientationListener.disable();
     final Exchanger<Boolean> result = new Exchanger<Boolean>();
     cameraThreadHandler.post(new Runnable() {
         @Override public void run() {
@@ -212,14 +233,12 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
     }
     cameraThreadHandler = null;
     cameraThread = null;
-    orientationListener.disable();
+    Log.d(TAG, "stopCapture done");
     return status;
   }
 
   private void stopCaptureOnCameraThread(
       Exchanger<Boolean> result) {
-    Log.d(TAG, "stopCapture");
-    Looper.myLooper().quit();
     if (camera == null) {
       throw new RuntimeException("Camera is already stopped!");
     }
@@ -232,10 +251,16 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
         camera.setPreviewDisplay(null);
       } else {
         camera.setPreviewTexture(null);
+        cameraSurfaceTexture = null;
+        if (cameraGlTextures != null) {
+          GLES20.glDeleteTextures(1, cameraGlTextures, 0);
+          cameraGlTextures = null;
+        }
       }
       camera.release();
       camera = null;
       exchange(result, true);
+      Looper.myLooper().quit();
       return;
     } catch (IOException e) {
       error = e;
@@ -244,11 +269,12 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
     }
     Log.e(TAG, "Failed to stop camera", error);
     exchange(result, false);
+    Looper.myLooper().quit();
     return;
   }
 
   private native void ProvideCameraFrame(
-      byte[] data, int length, long captureObject);
+      byte[] data, int length, long timeStamp, long captureObject);
 
   // Called on cameraThread so must not "synchronized".
   @Override
@@ -262,7 +288,19 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
     if (camera != callbackCamera) {
       throw new RuntimeException("Unexpected camera in callback!");
     }
-    ProvideCameraFrame(data, data.length, native_capturer);
+    frameCount++;
+    long captureTimeMs = SystemClock.elapsedRealtime();
+    if (frameCount > 1) {
+      double durationMs = captureTimeMs - lastCaptureTimeMs;
+      averageDurationMs = 0.9 * averageDurationMs + 0.1 * durationMs;
+      if ((frameCount % 30) == 0) {
+        Log.d(TAG, "Camera TS " + captureTimeMs +
+            ". Duration: " + (int)durationMs + " ms. FPS: " +
+            (int) (1000 / averageDurationMs + 0.5));
+      }
+    }
+    lastCaptureTimeMs = captureTimeMs;
+    ProvideCameraFrame(data, data.length, captureTimeMs, native_capturer);
     camera.addCallbackBuffer(data);
   }
 
diff --git a/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java b/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
index 1d681c06..fe207ca3 100644
--- a/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
+++ b/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
@@ -10,14 +10,8 @@
 
 package org.webrtc.videoengine;
 
-import java.io.File;
-import java.lang.reflect.InvocationTargetException;
-import java.lang.reflect.Method;
-import java.util.ArrayList;
 import java.util.List;
-import java.util.Locale;
 
-import android.content.Context;
 import android.hardware.Camera.CameraInfo;
 import android.hardware.Camera.Parameters;
 import android.hardware.Camera.Size;
@@ -99,6 +93,7 @@ public class VideoCaptureDeviceInfoAndroid {
             .put("mfpsRanges", mfpsRanges);
       }
       String ret = devices.toString(2);
+      Log.d(TAG, ret);
       return ret;
     } catch (JSONException e) {
       throw new RuntimeException(e);
diff --git a/modules/video_capture/android/video_capture_android.cc b/modules/video_capture/android/video_capture_android.cc
index c9aa52ce..6f0200e6 100644
--- a/modules/video_capture/android/video_capture_android.cc
+++ b/modules/video_capture/android/video_capture_android.cc
@@ -15,6 +15,7 @@
 #include "webrtc/modules/video_capture/android/device_info_android.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logcat_trace_context.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/ref_count.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
@@ -36,6 +37,7 @@ void JNICALL ProvideCameraFrame(
     jobject,
     jbyteArray javaCameraFrame,
     jint length,
+    jlong timeStamp,
     jlong context) {
   webrtc::videocapturemodule::VideoCaptureAndroid* captureModule =
       reinterpret_cast<webrtc::videocapturemodule::VideoCaptureAndroid*>(
@@ -90,7 +92,7 @@ int32_t SetCaptureAndroidVM(JavaVM* javaVM, jobject context) {
          "(JI)V",
          reinterpret_cast<void*>(&OnOrientationChanged)},
         {"ProvideCameraFrame",
-         "([BIJ)V",
+         "([BIJJ)V",
          reinterpret_cast<void*>(&ProvideCameraFrame)}};
     if (ats.env()->RegisterNatives(g_java_capturer_class,
                                    native_methods, 3) != 0)
@@ -146,18 +148,18 @@ int32_t VideoCaptureAndroid::Init(const int32_t id,
     return -1;
 
   // Store the device name
+  LOG(LS_INFO) << "VideoCaptureAndroid::Init: " << deviceUniqueIdUTF8;
+  size_t camera_id = 0;
+  if (!_deviceInfo.FindCameraIndex(deviceUniqueIdUTF8, &camera_id))
+    return -1;
   _deviceUniqueId = new char[nameLength + 1];
   memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
 
   AttachThreadScoped ats(g_jvm);
   JNIEnv* env = ats.env();
-
   jmethodID ctor = env->GetMethodID(g_java_capturer_class, "<init>", "(IJ)V");
   assert(ctor);
   jlong j_this = reinterpret_cast<intptr_t>(this);
-  size_t camera_id = 0;
-  if (!_deviceInfo.FindCameraIndex(deviceUniqueIdUTF8, &camera_id))
-    return -1;
   _jCapturer = env->NewGlobalRef(
       env->NewObject(g_java_capturer_class, ctor, camera_id, j_this));
   assert(_jCapturer);
@@ -234,7 +236,6 @@ int32_t VideoCaptureAndroid::CaptureSettings(
 
 int32_t VideoCaptureAndroid::SetCaptureRotation(
     VideoCaptureRotation rotation) {
-  CriticalSectionScoped cs(&_apiCs);
   int32_t status = VideoCaptureImpl::SetCaptureRotation(rotation);
   if (status != 0)
     return status;
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
new file mode 100644
index 00000000..f5a69b18
--- /dev/null
+++ b/modules/video_coding/BUILD.gn
@@ -0,0 +1,86 @@
+# Copyright (c) 2014 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.
+
+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/er_tables_xor.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_data.h",
+    "main/source/qm_select.cc",
+    "main/source/qm_select.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",
+  ]
+
+  deps = [
+    ":video_coding_utility",
+    ":webrtc_i420",
+    ":webrtc_vp8",
+    "../../common_video",
+    "../../system_wrappers",
+  ]
+}
+
+source_set("video_coding_utility") {
+  # TODO(stefan): Implement.
+}
+
+source_set("webrtc_i420") {
+  # TODO(stefan): Implement.
+}
+
+source_set("webrtc_vp8") {
+  # TODO(stefan): Implement.
+}
diff --git a/modules/video_coding/OWNERS b/modules/video_coding/OWNERS
index 7183cf21..037de93f 100644
--- a/modules/video_coding/OWNERS
+++ b/modules/video_coding/OWNERS
@@ -2,3 +2,5 @@ stefan@webrtc.org
 mikhal@webrtc.org
 marpan@webrtc.org
 henrik.lundin@webrtc.org
+
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/modules/video_processing/BUILD.gn b/modules/video_processing/BUILD.gn
new file mode 100644
index 00000000..40171caf
--- /dev/null
+++ b/modules/video_processing/BUILD.gn
@@ -0,0 +1,13 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("video_processing") {
+  # TODO(stefan): Implement.
+}
diff --git a/modules/video_processing/OWNERS b/modules/video_processing/OWNERS
new file mode 100644
index 00000000..d5ae8473
--- /dev/null
+++ b/modules/video_processing/OWNERS
@@ -0,0 +1 @@
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/modules/video_render/BUILD.gn b/modules/video_render/BUILD.gn
new file mode 100644
index 00000000..c569b7fa
--- /dev/null
+++ b/modules/video_render/BUILD.gn
@@ -0,0 +1,13 @@
+# Copyright (c) 2014 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.
+
+import("../../build/webrtc.gni")
+
+source_set("video_render") {
+  # TODO(wuchengli): Implement.
+}
diff --git a/modules/video_render/OWNERS b/modules/video_render/OWNERS
index 7dd4a362..da941f40 100644
--- a/modules/video_render/OWNERS
+++ b/modules/video_render/OWNERS
@@ -10,3 +10,5 @@ per-file *.isolate=kjellander@webrtc.org
 # 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