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, 0 insertions, 271 deletions

diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
deleted file mode 100644
index bd0335a5f3..0000000000
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ /dev/null
@@ -1,271 +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/audio_coding/main/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/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-#ifdef WEBRTC_CODEC_ISAC
-const CodecInst kIsacWb = {103, "ISAC", 16000, 480, 1, 32000};
-const CodecInst kIsacSwb = {104, "ISAC", 32000, 960, 1, 56000};
-#endif
-
-#ifdef WEBRTC_CODEC_ILBC
-const CodecInst kIlbc = {102, "ILBC", 8000, 240, 1, 13300};
-#endif
-
-#ifdef WEBRTC_CODEC_OPUS
-const CodecInst kOpus = {120, "opus", 48000, 960, 1, 64000};
-const CodecInst kOpusStereo = {120, "opus", 48000, 960, 2, 64000};
-#endif
-
-ActivityMonitor::ActivityMonitor() {
-  ResetStatistics();
-}
-
-int32_t ActivityMonitor::InFrameType(FrameType frame_type) {
-  counter_[frame_type]++;
-  return 0;
-}
-
-void ActivityMonitor::PrintStatistics() {
-  printf("\n");
-  printf("kEmptyFrame       %u\n", counter_[kEmptyFrame]);
-  printf("kAudioFrameSpeech %u\n", counter_[kAudioFrameSpeech]);
-  printf("kAudioFrameCN     %u\n", counter_[kAudioFrameCN]);
-  printf("kVideoFrameKey    %u\n", counter_[kVideoFrameKey]);
-  printf("kVideoFrameDelta  %u\n", counter_[kVideoFrameDelta]);
-  printf("\n\n");
-}
-
-void ActivityMonitor::ResetStatistics() {
-  memset(counter_, 0, sizeof(counter_));
-}
-
-void ActivityMonitor::GetStatistics(uint32_t* counter) {
-  memcpy(counter, counter_, sizeof(counter_));
-}
-
-TestVadDtx::TestVadDtx()
-    : acm_send_(AudioCodingModule::Create(0)),
-      acm_receive_(AudioCodingModule::Create(1)),
-      channel_(new Channel),
-      monitor_(new ActivityMonitor) {
-  EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
-  channel_->RegisterReceiverACM(acm_receive_.get());
-  EXPECT_EQ(0, acm_send_->RegisterVADCallback(monitor_.get()));
-}
-
-void TestVadDtx::RegisterCodec(CodecInst codec_param) {
-  // Set the codec for sending and receiving.
-  EXPECT_EQ(0, acm_send_->RegisterSendCodec(codec_param));
-  EXPECT_EQ(0, acm_receive_->RegisterReceiveCodec(codec_param));
-  channel_->SetIsStereo(codec_param.channels > 1);
-}
-
-// Encoding a file and see if the numbers that various packets occur follow
-// the expectation.
-void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
-                     std::string out_filename, bool append,
-                     const int* expects) {
-  monitor_->ResetStatistics();
-
-  PCMFile in_file;
-  in_file.Open(in_filename, frequency, "rb");
-  in_file.ReadStereo(channels > 1);
-
-  PCMFile out_file;
-  if (append) {
-    out_file.Open(out_filename, kOutputFreqHz, "ab");
-  } else {
-    out_file.Open(out_filename, kOutputFreqHz, "wb");
-  }
-
-  uint16_t frame_size_samples = in_file.PayloadLength10Ms();
-  uint32_t time_stamp = 0x12345678;
-  AudioFrame audio_frame;
-  while (!in_file.EndOfFile()) {
-    in_file.Read10MsData(audio_frame);
-    audio_frame.timestamp_ = time_stamp;
-    time_stamp += frame_size_samples;
-    EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
-    acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame);
-    out_file.Write10MsData(audio_frame);
-  }
-
-  in_file.Close();
-  out_file.Close();
-
-#ifdef PRINT_STAT
-  monitor_->PrintStatistics();
-#endif
-
-  uint32_t stats[5];
-  monitor_->GetStatistics(stats);
-  monitor_->ResetStatistics();
-
-  for (const auto& st : stats) {
-    int i = &st - stats;  // Calculate the current position in stats.
-    switch (expects[i]) {
-      case 0: {
-        EXPECT_EQ(0u, st) << "stats[" << i << "] error.";
-        break;
-      }
-      case 1: {
-        EXPECT_GT(st, 0u) << "stats[" << i << "] error.";
-        break;
-      }
-    }
-  }
-}
-
-// Following is the implementation of TestWebRtcVadDtx.
-TestWebRtcVadDtx::TestWebRtcVadDtx()
-    : vad_enabled_(false),
-      dtx_enabled_(false),
-      output_file_num_(0) {
-}
-
-void TestWebRtcVadDtx::Perform() {
-  // Go through various test cases.
-#ifdef WEBRTC_CODEC_ISAC
-  // Register iSAC WB as send codec
-  RegisterCodec(kIsacWb);
-  RunTestCases();
-
-  // Register iSAC SWB as send codec
-  RegisterCodec(kIsacSwb);
-  RunTestCases();
-#endif
-
-#ifdef WEBRTC_CODEC_ILBC
-  // Register iLBC as send codec
-  RegisterCodec(kIlbc);
-  RunTestCases();
-#endif
-
-#ifdef WEBRTC_CODEC_OPUS
-  // Register Opus as send codec
-  RegisterCodec(kOpus);
-  RunTestCases();
-#endif
-}
-
-// Test various configurations on VAD/DTX.
-void TestWebRtcVadDtx::RunTestCases() {
-  // #1 DTX = OFF, VAD = OFF, VADNormal
-  SetVAD(false, false, VADNormal);
-  Test(true);
-
-  // #2 DTX = ON, VAD = ON, VADAggr
-  SetVAD(true, true, VADAggr);
-  Test(false);
-
-  // #3 DTX = ON, VAD = ON, VADLowBitrate
-  SetVAD(true, true, VADLowBitrate);
-  Test(false);
-
-  // #4 DTX = ON, VAD = ON, VADVeryAggr
-  SetVAD(true, true, VADVeryAggr);
-  Test(false);
-
-  // #5 DTX = ON, VAD = ON, VADNormal
-  SetVAD(true, true, VADNormal);
-  Test(false);
-}
-
-// Set the expectation and run the test.
-void TestWebRtcVadDtx::Test(bool new_outfile) {
-  int expects[] = {-1, 1, dtx_enabled_, 0, 0};
-  if (new_outfile) {
-    output_file_num_++;
-  }
-  std::stringstream out_filename;
-  out_filename << webrtc::test::OutputPath()
-               << "testWebRtcVadDtx_outFile_"
-               << output_file_num_
-               << ".pcm";
-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
-      32000, 1, out_filename.str(), !new_outfile, expects);
-}
-
-void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
-                              ACMVADMode vad_mode) {
-  ACMVADMode mode;
-  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) {
-    // If send codec is Opus, WebRTC VAD/DTX cannot be used.
-    enable_dtx = enable_vad = false;
-  }
-
-  EXPECT_EQ(dtx_enabled_ , enable_dtx); // DTX should be set as expected.
-
-  if (dtx_enabled_) {
-    EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
-  } else {
-    // Using no DTX should not affect setting of VAD.
-    EXPECT_EQ(enable_vad, vad_enabled_);
-  }
-}
-
-// Following is the implementation of TestOpusDtx.
-void TestOpusDtx::Perform() {
-#ifdef WEBRTC_CODEC_ISAC
-  // If we set other codec than Opus, DTX cannot be toggled.
-  RegisterCodec(kIsacWb);
-  EXPECT_EQ(-1, acm_send_->EnableOpusDtx());
-  EXPECT_EQ(-1, acm_send_->DisableOpusDtx());
-#endif
-
-#ifdef WEBRTC_CODEC_OPUS
-  int expects[] = {0, 1, 0, 0, 0};
-
-  // Register Opus as send codec
-  std::string out_filename = webrtc::test::OutputPath() +
-      "testOpusDtx_outFile_mono.pcm";
-  RegisterCodec(kOpus);
-  EXPECT_EQ(0, acm_send_->DisableOpusDtx());
-
-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
-      32000, 1, out_filename, false, expects);
-
-  EXPECT_EQ(0, acm_send_->EnableOpusDtx());
-  expects[kEmptyFrame] = 1;
-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
-      32000, 1, out_filename, true, expects);
-
-  // Register stereo Opus as send codec
-  out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
-  RegisterCodec(kOpusStereo);
-  EXPECT_EQ(0, acm_send_->DisableOpusDtx());
-  expects[kEmptyFrame] = 0;
-  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
-      32000, 2, out_filename, false, expects);
-
-  EXPECT_EQ(0, acm_send_->EnableOpusDtx());
-
-  expects[kEmptyFrame] = 1;
-  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
-      32000, 2, out_filename, true, expects);
-#endif
-}
-
-}  // namespace webrtc