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, 175 deletions

diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
deleted file mode 100644
index 8495e0e596..0000000000
--- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ /dev/null
@@ -1,175 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-
-#include <assert.h>
-#include <math.h>
-
-#include <iostream>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-namespace {
-
-double FrameRms(AudioFrame& frame) {
-  size_t samples = frame.num_channels_ * frame.samples_per_channel_;
-  double rms = 0;
-  for (size_t n = 0; n < samples; ++n)
-    rms += frame.data_[n] * frame.data_[n];
-  rms /= samples;
-  rms = sqrt(rms);
-  return rms;
-}
-
-}
-
-class InitialPlayoutDelayTest : public ::testing::Test {
- protected:
-  InitialPlayoutDelayTest()
-      : acm_a_(AudioCodingModule::Create(0)),
-        acm_b_(AudioCodingModule::Create(1)),
-        channel_a2b_(NULL) {}
-
-  ~InitialPlayoutDelayTest() {
-    if (channel_a2b_ != NULL) {
-      delete channel_a2b_;
-      channel_a2b_ = NULL;
-    }
-  }
-
-  void SetUp() {
-    ASSERT_TRUE(acm_a_.get() != NULL);
-    ASSERT_TRUE(acm_b_.get() != NULL);
-
-    EXPECT_EQ(0, acm_b_->InitializeReceiver());
-    EXPECT_EQ(0, acm_a_->InitializeReceiver());
-
-    // Register all L16 codecs in receiver.
-    CodecInst codec;
-    const int kFsHz[3] = { 8000, 16000, 32000 };
-    const int kChannels[2] = { 1, 2 };
-    for (int n = 0; n < 3; ++n) {
-      for (int k = 0; k < 2; ++k) {
-        AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
-        acm_b_->RegisterReceiveCodec(codec);
-      }
-    }
-
-    // Create and connect the channel
-    channel_a2b_ = new Channel;
-    acm_a_->RegisterTransportCallback(channel_a2b_);
-    channel_a2b_->RegisterReceiverACM(acm_b_.get());
-  }
-
-  void NbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 1);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
-  void NbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 2);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
- private:
-  void Run(CodecInst codec, int initial_delay_ms) {
-    AudioFrame in_audio_frame;
-    AudioFrame out_audio_frame;
-    int num_frames = 0;
-    const int kAmp = 10000;
-    in_audio_frame.sample_rate_hz_ = codec.plfreq;
-    in_audio_frame.num_channels_ = codec.channels;
-    in_audio_frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
-    size_t samples = in_audio_frame.num_channels_ *
-        in_audio_frame.samples_per_channel_;
-    for (size_t n = 0; n < samples; ++n) {
-      in_audio_frame.data_[n] = kAmp;
-    }
-
-    uint32_t timestamp = 0;
-    double rms = 0;
-    ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
-    acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
-    while (rms < kAmp / 2) {
-      in_audio_frame.timestamp_ = timestamp;
-      timestamp += static_cast<uint32_t>(in_audio_frame.samples_per_channel_);
-      ASSERT_GE(acm_a_->Add10MsData(in_audio_frame), 0);
-      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
-      rms = FrameRms(out_audio_frame);
-      ++num_frames;
-    }
-
-    ASSERT_GE(num_frames * 10, initial_delay_ms);
-    ASSERT_LE(num_frames * 10, initial_delay_ms + 100);
-  }
-
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
-  Channel* channel_a2b_;
-};
-
-TEST_F(InitialPlayoutDelayTest, NbMono) { NbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, WbMono) { WbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
-
-}  // namespace webrtc