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.
  ...

1 files changed, 222 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
new file mode 100644
index 0000000000..855a39e675
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -0,0 +1,222 @@
+/*
+ *  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.
+ */
+
+#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/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"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+// Returns true if the codec should be registered, otherwise false. Changes
+// the number of channels for the Opus codec to always be 1.
+bool ModifyAndUseThisCodec(CodecInst* codec_param) {
+  if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
+      codec_param->plfreq == 48000)
+    return false;  // Skip 48 kHz comfort noise.
+
+  if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
+    return false;  // Skip DTFM.
+
+  return true;
+}
+
+// Remaps payload types from ACM's default to those used in the resource file
+// neteq_universal_new.rtp. Returns true if the codec should be registered,
+// otherwise false. The payload types are set as follows (all are mono codecs):
+// PCMu = 0;
+// PCMa = 8;
+// Comfort noise 8 kHz = 13
+// Comfort noise 16 kHz = 98
+// Comfort noise 32 kHz = 99
+// iLBC = 102
+// iSAC wideband = 103
+// iSAC super-wideband = 104
+// AVT/DTMF = 106
+// RED = 117
+// PCM16b 8 kHz = 93
+// PCM16b 16 kHz = 94
+// PCM16b 32 kHz = 95
+// G.722 = 94
+bool RemapPltypeAndUseThisCodec(const char* plname,
+                                int plfreq,
+                                size_t channels,
+                                int* pltype) {
+  if (channels != 1)
+    return false;  // Don't use non-mono codecs.
+
+  // Re-map pltypes to those used in the NetEq test files.
+  if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
+    *pltype = 0;
+  } else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
+    *pltype = 8;
+  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
+    *pltype = 13;
+  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
+    *pltype = 98;
+  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
+    *pltype = 99;
+  } else if (STR_CASE_CMP(plname, "ILBC") == 0) {
+    *pltype = 102;
+  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
+    *pltype = 103;
+  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
+    *pltype = 104;
+  } else if (STR_CASE_CMP(plname, "telephone-event") == 0) {
+    *pltype = 106;
+  } else if (STR_CASE_CMP(plname, "red") == 0) {
+    *pltype = 117;
+  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
+    *pltype = 93;
+  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
+    *pltype = 94;
+  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
+    *pltype = 95;
+  } else if (STR_CASE_CMP(plname, "G722") == 0) {
+    *pltype = 9;
+  } else {
+    // Don't use any other codecs.
+    return false;
+  }
+  return true;
+}
+}  // namespace
+
+AcmReceiveTestOldApi::AcmReceiveTestOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz,
+    NumOutputChannels exptected_output_channels)
+    : clock_(0),
+      acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
+      packet_source_(packet_source),
+      audio_sink_(audio_sink),
+      output_freq_hz_(output_freq_hz),
+      exptected_output_channels_(exptected_output_channels) {
+}
+
+void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
+  CodecInst my_codec_param;
+  for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
+    ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
+    if (ModifyAndUseThisCodec(&my_codec_param)) {
+      ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
+          << "Couldn't register receive codec.\n";
+    }
+  }
+}
+
+void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
+  CodecInst my_codec_param;
+  for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
+    ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
+    if (!ModifyAndUseThisCodec(&my_codec_param)) {
+      // Skip this codec.
+      continue;
+    }
+
+    if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
+                                   my_codec_param.plfreq,
+                                   my_codec_param.channels,
+                                   &my_codec_param.pltype)) {
+      ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
+          << "Couldn't register receive codec.\n";
+    }
+  }
+}
+
+int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
+    int rtp_payload_type,
+    AudioDecoder* external_decoder,
+    int sample_rate_hz,
+    int num_channels,
+    const std::string& name) {
+  return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
+                                            sample_rate_hz, num_channels, name);
+}
+
+void AcmReceiveTestOldApi::Run() {
+  for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
+       packet.reset(packet_source_->NextPacket())) {
+    // Pull audio until time to insert packet.
+    while (clock_.TimeInMilliseconds() < packet->time_ms()) {
+      AudioFrame output_frame;
+      EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
+      EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
+      const size_t samples_per_block =
+          static_cast<size_t>(output_freq_hz_ * 10 / 1000);
+      EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
+      if (exptected_output_channels_ != kArbitraryChannels) {
+        if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
+          // Don't check number of channels for PLC output, since each test run
+          // usually starts with a short period of mono PLC before decoding the
+          // first packet.
+        } else {
+          EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
+        }
+      }
+      ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
+      clock_.AdvanceTimeMilliseconds(10);
+      AfterGetAudio();
+    }
+
+    // Insert packet after converting from RTPHeader to WebRtcRTPHeader.
+    WebRtcRTPHeader header;
+    header.header = packet->header();
+    header.frameType = kAudioFrameSpeech;
+    memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
+    EXPECT_EQ(0,
+              acm_->IncomingPacket(
+                  packet->payload(),
+                  static_cast<int32_t>(packet->payload_length_bytes()),
+                  header))
+        << "Failure when inserting packet:" << std::endl
+        << "  PT = " << static_cast<int>(header.header.payloadType) << std::endl
+        << "  TS = " << header.header.timestamp << std::endl
+        << "  SN = " << header.header.sequenceNumber;
+  }
+}
+
+AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz_1,
+    int output_freq_hz_2,
+    int toggle_period_ms,
+    NumOutputChannels exptected_output_channels)
+    : AcmReceiveTestOldApi(packet_source,
+                           audio_sink,
+                           output_freq_hz_1,
+                           exptected_output_channels),
+      output_freq_hz_1_(output_freq_hz_1),
+      output_freq_hz_2_(output_freq_hz_2),
+      toggle_period_ms_(toggle_period_ms),
+      last_toggle_time_ms_(clock_.TimeInMilliseconds()) {
+}
+
+void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
+  if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
+    output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
+                          ? output_freq_hz_2_
+                          : output_freq_hz_1_;
+    last_toggle_time_ms_ = clock_.TimeInMilliseconds();
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc