MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc_local_audio_track_unittest.cc

Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
deleted file mode 100644
index 51af138563639080e6770681332ea88ed4b5508a..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ /dev/null
@@ -1,579 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "base/macros.h"
-#include "base/synchronization/waitable_event.h"
-#include "base/test/test_timeouts.h"
-#include "build/build_config.h"
-#include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/media_stream_audio_source.h"
-#include "content/renderer/media/mock_constraint_factory.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-#include "media/base/audio_bus.h"
-#include "media/base/audio_capturer_source.h"
-#include "media/base/audio_parameters.h"
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
-#include "third_party/WebKit/public/web/WebHeap.h"
-#include "third_party/webrtc/api/mediastreaminterface.h"
-
-using ::testing::_;
-using ::testing::AnyNumber;
-using ::testing::AtLeast;
-using ::testing::Return;
-
-namespace content {
-
-namespace {
-
-ACTION_P(SignalEvent, event) {
-  event->Signal();
-}
-
-// A simple thread that we use to fake the audio thread which provides data to
-// the |WebRtcAudioCapturer|.
-class FakeAudioThread : public base::PlatformThread::Delegate {
- public:
-  FakeAudioThread(WebRtcAudioCapturer* capturer,
-                  const media::AudioParameters& params)
-    : capturer_(capturer),
-      thread_(),
-      closure_(false, false) {
-    DCHECK(capturer);
-    audio_bus_ = media::AudioBus::Create(params);
-  }
-
-  ~FakeAudioThread() override { DCHECK(thread_.is_null()); }
-
-  // base::PlatformThread::Delegate:
-  void ThreadMain() override {
-    while (true) {
-      if (closure_.IsSignaled())
-        return;
-
-      media::AudioCapturerSource::CaptureCallback* callback =
-          static_cast<media::AudioCapturerSource::CaptureCallback*>(
-              capturer_);
-      audio_bus_->Zero();
-      callback->Capture(audio_bus_.get(), 0, 0, false);
-
-      // Sleep 1ms to yield the resource for the main thread.
-      base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
-    }
-  }
-
-  void Start() {
-    base::PlatformThread::CreateWithPriority(
-        0, this, &thread_, base::ThreadPriority::REALTIME_AUDIO);
-    CHECK(!thread_.is_null());
-  }
-
-  void Stop() {
-    closure_.Signal();
-    base::PlatformThread::Join(thread_);
-    thread_ = base::PlatformThreadHandle();
-  }
-
- private:
-  std::unique_ptr<media::AudioBus> audio_bus_;
-  WebRtcAudioCapturer* capturer_;
-  base::PlatformThreadHandle thread_;
-  base::WaitableEvent closure_;
-  DISALLOW_COPY_AND_ASSIGN(FakeAudioThread);
-};
-
-class MockCapturerSource : public media::AudioCapturerSource {
- public:
-  explicit MockCapturerSource(WebRtcAudioCapturer* capturer)
-      : capturer_(capturer) {}
-  MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params,
-                                  CaptureCallback* callback,
-                                  int session_id));
-  MOCK_METHOD0(OnStart, void());
-  MOCK_METHOD0(OnStop, void());
-  void SetVolume(double volume) final {}
-  MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
-
-  void Initialize(const media::AudioParameters& params,
-                  CaptureCallback* callback,
-                  int session_id) override {
-    DCHECK(params.IsValid());
-    params_ = params;
-    OnInitialize(params, callback, session_id);
-  }
-  void Start() override {
-    audio_thread_.reset(new FakeAudioThread(capturer_, params_));
-    audio_thread_->Start();
-    OnStart();
-  }
-  void Stop() override {
-    audio_thread_->Stop();
-    audio_thread_.reset();
-    OnStop();
-  }
-
- protected:
-  ~MockCapturerSource() override {}
-
- private:
-  std::unique_ptr<FakeAudioThread> audio_thread_;
-  WebRtcAudioCapturer* capturer_;
-  media::AudioParameters params_;
-};
-
-class MockMediaStreamAudioSink : public MediaStreamAudioSink {
- public:
-  MockMediaStreamAudioSink() {}
-  ~MockMediaStreamAudioSink() {}
-  void OnData(const media::AudioBus& audio_bus,
-              base::TimeTicks estimated_capture_time) override {
-    EXPECT_EQ(params_.channels(), audio_bus.channels());
-    EXPECT_EQ(params_.frames_per_buffer(), audio_bus.frames());
-    EXPECT_FALSE(estimated_capture_time.is_null());
-    CaptureData();
-  }
-  MOCK_METHOD0(CaptureData, void());
-  void OnSetFormat(const media::AudioParameters& params) {
-    params_ = params;
-    FormatIsSet();
-  }
-  MOCK_METHOD0(FormatIsSet, void());
-
-  const media::AudioParameters& audio_params() const { return params_; }
-
- private:
-  media::AudioParameters params_;
-};
-
-}  // namespace
-
-class WebRtcLocalAudioTrackTest : public ::testing::Test {
- protected:
-  void SetUp() override {
-    params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                  media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480);
-    MockConstraintFactory constraint_factory;
-    blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
-                             "dummy",
-                             false /* remote */);
-    MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
-    blink_source_.setExtraData(audio_source);
-
-    StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
-                            std::string(), std::string());
-    {
-      std::unique_ptr<WebRtcAudioCapturer> capturer =
-          WebRtcAudioCapturer::CreateCapturer(
-              -1, device, constraint_factory.CreateWebMediaConstraints(),
-              nullptr, audio_source);
-      capturer_ = capturer.get();
-      audio_source->SetAudioCapturer(std::move(capturer));
-    }
-    capturer_source_ = new MockCapturerSource(capturer_);
-    EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_, -1))
-        .WillOnce(Return());
-    EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
-    EXPECT_CALL(*capturer_source_.get(), OnStart());
-    capturer_->SetCapturerSource(capturer_source_, params_);
-  }
-
-  void TearDown() override {
-    blink_source_.reset();
-    blink::WebHeap::collectAllGarbageForTesting();
-  }
-
-  media::AudioParameters params_;
-  blink::WebMediaStreamSource blink_source_;
-  WebRtcAudioCapturer* capturer_;  // Owned by |blink_source_|.
-  scoped_refptr<MockCapturerSource> capturer_source_;
-};
-
-// Creates a capturer and audio track, fakes its audio thread, and
-// connect/disconnect the sink to the audio track on the fly, the sink should
-// get data callback when the track is connected to the capturer but not when
-// the track is disconnected from the capturer.
-TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get()));
-  track->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track.get()));
-  capturer_->AddTrack(track.get());
-  EXPECT_TRUE(track->GetAudioAdapter()->enabled());
-
-  std::unique_ptr<MockMediaStreamAudioSink> sink(
-      new MockMediaStreamAudioSink());
-  base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, FormatIsSet());
-  EXPECT_CALL(*sink,
-      CaptureData()).Times(AtLeast(1))
-      .WillRepeatedly(SignalEvent(&event));
-  track->AddSink(sink.get());
-  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-  track->RemoveSink(sink.get());
-
-  EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
-  capturer_->Stop();
-}
-
-// The same setup as ConnectAndDisconnectOneSink, but enable and disable the
-// audio track on the fly. When the audio track is disabled, there is no data
-// callback to the sink; when the audio track is enabled, there comes data
-// callback.
-// TODO(xians): Enable this test after resolving the racing issue that TSAN
-// reports on MediaStreamTrack::enabled();
-TEST_F(WebRtcLocalAudioTrackTest,  DISABLED_DisableEnableAudioTrack) {
-  EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
-  EXPECT_CALL(*capturer_source_.get(), OnStart());
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get()));
-  track->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track.get()));
-  capturer_->AddTrack(track.get());
-  EXPECT_TRUE(track->GetAudioAdapter()->enabled());
-  EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
-  std::unique_ptr<MockMediaStreamAudioSink> sink(
-      new MockMediaStreamAudioSink());
-  const media::AudioParameters params = capturer_->GetInputFormat();
-  base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, FormatIsSet()).Times(1);
-  EXPECT_CALL(*sink, CaptureData()).Times(0);
-  EXPECT_EQ(sink->audio_params().frames_per_buffer(),
-            params.sample_rate() / 100);
-  track->AddSink(sink.get());
-  EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
-
-  event.Reset();
-  EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1))
-      .WillRepeatedly(SignalEvent(&event));
-  EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
-  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-  track->RemoveSink(sink.get());
-
-  EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
-  capturer_->Stop();
-  track.reset();
-}
-
-// Create multiple audio tracks and enable/disable them, verify that the audio
-// callbacks appear/disappear.
-// Flaky due to a data race, see http://crbug.com/295418
-TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get()));
-  track_1->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track_1.get()));
-  capturer_->AddTrack(track_1.get());
-  EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
-  std::unique_ptr<MockMediaStreamAudioSink> sink_1(
-      new MockMediaStreamAudioSink());
-  const media::AudioParameters params = capturer_->GetInputFormat();
-  base::WaitableEvent event_1(false, false);
-  EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
-  EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1))
-      .WillRepeatedly(SignalEvent(&event_1));
-  EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
-            params.sample_rate() / 100);
-  track_1->AddSink(sink_1.get());
-  EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
-
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get()));
-  track_2->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track_2.get()));
-  capturer_->AddTrack(track_2.get());
-  EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
-
-  // Verify both |sink_1| and |sink_2| get data.
-  event_1.Reset();
-  base::WaitableEvent event_2(false, false);
-
-  std::unique_ptr<MockMediaStreamAudioSink> sink_2(
-      new MockMediaStreamAudioSink());
-  EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
-  EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1))
-      .WillRepeatedly(SignalEvent(&event_1));
-  EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
-            params.sample_rate() / 100);
-  EXPECT_CALL(*sink_2, CaptureData()).Times(AtLeast(1))
-      .WillRepeatedly(SignalEvent(&event_2));
-  EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
-            params.sample_rate() / 100);
-  track_2->AddSink(sink_2.get());
-  EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
-  EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
-
-  track_1->RemoveSink(sink_1.get());
-  track_1->Stop();
-  track_1.reset();
-
-  EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
-  track_2->RemoveSink(sink_2.get());
-  track_2->Stop();
-  track_2.reset();
-}
-
-
-// Start one track and verify the capturer is correctly starting its source.
-// And it should be fine to not to call Stop() explicitly.
-TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get()));
-  track->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track.get()));
-  capturer_->AddTrack(track.get());
-
-  // When the track goes away, it will automatically stop the
-  // |capturer_source_|.
-  EXPECT_CALL(*capturer_source_.get(), OnStop());
-  track.reset();
-}
-
-// Start two tracks and verify the capturer is correctly starting its source.
-// When the last track connected to the capturer is stopped, the source is
-// stopped.
-TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) {
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track1(
-      new WebRtcLocalAudioTrack(adapter1.get()));
-  track1->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track1.get()));
-  capturer_->AddTrack(track1.get());
-
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
-        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track2(
-      new WebRtcLocalAudioTrack(adapter2.get()));
-  track2->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track2.get()));
-  capturer_->AddTrack(track2.get());
-
-  track1->Stop();
-  // When the last track is stopped, it will automatically stop the
-  // |capturer_source_|.
-  EXPECT_CALL(*capturer_source_.get(), OnStop());
-  track2->Stop();
-}
-
-// Start/Stop tracks and verify the capturer is correctly starting/stopping
-// its source.
-TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
-  base::WaitableEvent event(false, false);
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get()));
-  track_1->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track_1.get()));
-  capturer_->AddTrack(track_1.get());
-
-  // Verify the data flow by connecting the sink to |track_1|.
-  std::unique_ptr<MockMediaStreamAudioSink> sink(
-      new MockMediaStreamAudioSink());
-  event.Reset();
-  EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
-  EXPECT_CALL(*sink, CaptureData())
-      .Times(AnyNumber()).WillRepeatedly(Return());
-  track_1->AddSink(sink.get());
-  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-
-  // Start the second audio track will not start the |capturer_source_|
-  // since it has been started.
-  EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get()));
-  track_2->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track_2.get()));
-  capturer_->AddTrack(track_2.get());
-
-  // Stop the capturer will clear up the track lists in the capturer.
-  EXPECT_CALL(*capturer_source_.get(), OnStop());
-  capturer_->Stop();
-
-  // Adding a new track to the capturer.
-  track_2->AddSink(sink.get());
-  EXPECT_CALL(*sink, FormatIsSet()).Times(0);
-
-  // Stop the capturer again will not trigger stopping the source of the
-  // capturer again..
-  event.Reset();
-  EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
-  capturer_->Stop();
-}
-
-// Create a new capturer with new source, connect it to a new audio track.
-#if defined(THREAD_SANITIZER)
-// Fails under TSan, see https://crbug.com/576634.
-#define MAYBE_ConnectTracksToDifferentCapturers \
-    DISABLED_ConnectTracksToDifferentCapturers
-#else
-#define MAYBE_ConnectTracksToDifferentCapturers \
-    ConnectTracksToDifferentCapturers
-#endif
-TEST_F(WebRtcLocalAudioTrackTest, MAYBE_ConnectTracksToDifferentCapturers) {
-  // Setup the first audio track and start it.
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get()));
-  track_1->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track_1.get()));
-  capturer_->AddTrack(track_1.get());
-
-  // Verify the data flow by connecting the |sink_1| to |track_1|.
-  std::unique_ptr<MockMediaStreamAudioSink> sink_1(
-      new MockMediaStreamAudioSink());
-  EXPECT_CALL(*sink_1.get(), CaptureData())
-      .Times(AnyNumber()).WillRepeatedly(Return());
-  EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
-  track_1->AddSink(sink_1.get());
-
-  // Create a new capturer with new source with different audio format.
-  MockConstraintFactory constraint_factory;
-  StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
-                          std::string(), std::string());
-  std::unique_ptr<WebRtcAudioCapturer> new_capturer(
-      WebRtcAudioCapturer::CreateCapturer(
-          -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
-          NULL));
-  scoped_refptr<MockCapturerSource> new_source(
-      new MockCapturerSource(new_capturer.get()));
-  EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
-  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
-  EXPECT_CALL(*new_source.get(), OnStart());
-
-  media::AudioParameters new_param(
-      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-      media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
-  new_capturer->SetCapturerSource(new_source, new_param);
-
-  // Setup the second audio track, connect it to the new capturer and start it.
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get()));
-  track_2->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track_2.get()));
-  new_capturer->AddTrack(track_2.get());
-
-  // Verify the data flow by connecting the |sink_2| to |track_2|.
-  std::unique_ptr<MockMediaStreamAudioSink> sink_2(
-      new MockMediaStreamAudioSink());
-  base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink_2, CaptureData())
-      .Times(AnyNumber()).WillRepeatedly(Return());
-  EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
-  track_2->AddSink(sink_2.get());
-  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-
-  // Stopping the new source will stop the second track.
-  event.Reset();
-  EXPECT_CALL(*new_source.get(), OnStop())
-      .Times(1).WillOnce(SignalEvent(&event));
-  new_capturer->Stop();
-  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-
-  // Stop the capturer of the first audio track.
-  EXPECT_CALL(*capturer_source_.get(), OnStop());
-  capturer_->Stop();
-}
-
-// Make sure a audio track can deliver packets with a buffer size smaller than
-// 10ms when it is not connected with a peer connection.
-TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
-  // Setup a capturer which works with a buffer size smaller than 10ms.
-  media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
-
-  // Create a capturer with new source which works with the format above.
-  MockConstraintFactory factory;
-  factory.DisableDefaultAudioConstraints();
-  std::unique_ptr<WebRtcAudioCapturer> capturer(
-      WebRtcAudioCapturer::CreateCapturer(
-          -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "",
-                               params.sample_rate(), params.channel_layout(),
-                               params.frames_per_buffer()),
-          factory.CreateWebMediaConstraints(), NULL, NULL));
-  scoped_refptr<MockCapturerSource> source(
-      new MockCapturerSource(capturer.get()));
-  EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
-  EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
-  EXPECT_CALL(*source.get(), OnStart());
-  capturer->SetCapturerSource(source, params);
-
-  // Setup a audio track, connect it to the capturer and start it.
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  std::unique_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get()));
-  track->Start(
-      base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo,
-                 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(),
-                 track.get()));
-  capturer->AddTrack(track.get());
-
-  // Verify the data flow by connecting the |sink| to |track|.
-  std::unique_ptr<MockMediaStreamAudioSink> sink(
-      new MockMediaStreamAudioSink());
-  base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, FormatIsSet()).Times(1);
-  // Verify the sinks are getting the packets with an expecting buffer size.
-#if defined(OS_ANDROID)
-  const int expected_buffer_size = params.sample_rate() / 100;
-#else
-  const int expected_buffer_size = params.frames_per_buffer();
-#endif
-  EXPECT_CALL(*sink, CaptureData())
-      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
-  track->AddSink(sink.get());
-  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
-  EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
-
-  // Stopping the new source will stop the second track.
-  EXPECT_CALL(*source.get(), OnStop()).Times(1);
-  capturer->Stop();
-
-  // Even though this test don't use |capturer_source_| it will be stopped
-  // during teardown of the test harness.
-  EXPECT_CALL(*capturer_source_.get(), OnStop());
-}
-
-}  // namespace content