MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc/processed_local_audio_track_unittest.cc

 // 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_media_constraint_factory.h"
+#include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.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/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_capturer_source.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"
@@ skipping 66 matching lines @@
 
 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());
-  MOCK_METHOD1(SetVolume, void(double volume));
+  void SetVolume(double volume) final {}  // Ignored.
   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 {
@@ skipping 38 matching lines @@
 
  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);
-    MockMediaConstraintFactory constraint_factory;
-    blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
-                             "dummy",
-                             false /* remote */, true /* readonly */);
-    MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
-    blink_source_.setExtraData(audio_source);
-
-    StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
-                            std::string(), std::string());
-    capturer_ = WebRtcAudioCapturer::CreateCapturer(
-        -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
-        audio_source);
-    audio_source->SetAudioCapturer(capturer_.get());
-    capturer_source_ = new MockCapturerSource(capturer_.get());
-    EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
-        .WillOnce(Return());
-    EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
-    EXPECT_CALL(*capturer_source_.get(), OnStart());
-    capturer_->SetCapturerSource(capturer_source_, params_);
+    CreateSourceWithParams(media::AudioParameters(
+        media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+        media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480));
   }
 
   void TearDown() override {
     blink_source_.reset();
     blink::WebHeap::collectAllGarbageForTesting();
   }
 
+  void CreateSourceWithParams(const media::AudioParameters& params) {
+    params_ = params;
+    MockMediaConstraintFactory constraint_factory;
+    blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
+                             "dummy",
+                             false /* remote */, true /* readonly */);
+    ProcessedLocalAudioSource* const audio_source =
+        new ProcessedLocalAudioSource(
+            -1,
+            StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, std::string(),
+                             std::string(), params_.sample_rate(),
+                             params_.channel_layout(),
+                             params_.frames_per_buffer()),
+            &mock_dependency_factory_);
+    audio_source->SetAllowInvalidRenderFrameIdForTesting(true);
+    audio_source->SetSourceConstraints(
+        MockMediaConstraintFactory().CreateWebMediaConstraints());
+    blink_source_.setExtraData(audio_source);
+
+    audio_source->StartSourceForTesting();
+    capturer_ = audio_source->audio_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_);
+  }
+
+  MockPeerConnectionDependencyFactory mock_dependency_factory_;
   media::AudioParameters params_;
   blink::WebMediaStreamSource blink_source_;
   scoped_refptr<MockCapturerSource> capturer_source_;
-  scoped_refptr<WebRtcAudioCapturer> capturer_;
+  WebRtcAudioCapturer* capturer_;  // Owned by ProcessedLocalAudioSource.
 };
 
 // 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));
-  scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
-  track->Start();
-  EXPECT_TRUE(track->GetAudioAdapter()->enabled());
+  blink::WebMediaStreamTrack blink_track;
+  blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+                         blink_source_);
+  ASSERT_TRUE(
+      MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+
+  WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+      MediaStreamAudioTrack::Get(blink_track));
+  EXPECT_TRUE(track->adapter()->enabled());
 
   scoped_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));
-  scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
-  track->Start();
-  EXPECT_TRUE(track->GetAudioAdapter()->enabled());
-  EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
+  blink::WebMediaStreamTrack blink_track;
+  blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+                         blink_source_);
+  ASSERT_TRUE(
+      MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+
+  WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+      MediaStreamAudioTrack::Get(blink_track));
+  EXPECT_TRUE(track->adapter()->enabled());
+
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->source_audio_parameters();
   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(track->adapter()->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));
-  scoped_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
-  track_1->Start();
-  EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
+  blink::WebMediaStreamTrack blink_track_1;
+  blink_track_1.initialize(blink::WebString::fromUTF8("dummy_track"),
+                           blink_source_);
+  ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+                  ->ConnectToTrack(blink_track_1));
+
+  WebRtcLocalAudioTrack* const track_1 = WebRtcLocalAudioTrack::From(
+      MediaStreamAudioTrack::Get(blink_track_1));
+  EXPECT_TRUE(track_1->adapter()->enabled());
+
   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->source_audio_parameters();
   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));
-  scoped_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
-  track_2->Start();
-  EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
+  blink::WebMediaStreamTrack blink_track_2;
+  blink_track_2.initialize(blink::WebString::fromUTF8("dummy_track"),
+                           blink_source_);
+  ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+                  ->ConnectToTrack(blink_track_2));
+
+  WebRtcLocalAudioTrack* const track_2 = WebRtcLocalAudioTrack::From(
+      MediaStreamAudioTrack::Get(blink_track_2));
+  EXPECT_TRUE(track_2->adapter()->enabled());
 
   // Verify both |sink_1| and |sink_2| get data.
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
 
   scoped_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();
+  blink_track_1.reset();
 
   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
   track_2->RemoveSink(sink_2.get());
   track_2->Stop();
-  track_2.reset();
+  blink_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));
-  scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
-  track->Start();
+  blink::WebMediaStreamTrack blink_track;
+  blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+                         blink_source_);
+  ASSERT_TRUE(
+      MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+
+  WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+      MediaStreamAudioTrack::Get(blink_track));
+  EXPECT_TRUE(track->adapter()->enabled());
 
   // 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));
-  scoped_ptr<WebRtcLocalAudioTrack> track1(
-      new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL));
-  track1->Start();
+  blink::WebMediaStreamTrack blink_track_1;
+  blink_track_1.initialize(blink::WebString::fromUTF8("dummy_track"),
+                           blink_source_);
+  ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+                  ->ConnectToTrack(blink_track_1));
 
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
-        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  scoped_ptr<WebRtcLocalAudioTrack> track2(
-      new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL));
-  track2->Start();
+  blink::WebMediaStreamTrack blink_track_2;
+  blink_track_2.initialize(blink::WebString::fromUTF8("dummy_track"),
+                           blink_source_);
+  // Starting the second audio track will not start the |capturer_source_|
+  // since it has been started.
+  EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
+  ASSERT_TRUE(MediaStreamAudioSource::Get(blink_source_)
+                  ->ConnectToTrack(blink_track_2));
 
-  track1->Stop();
+  WebRtcLocalAudioTrack::From(MediaStreamAudioTrack::Get(blink_track_1))
+      ->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));
-  scoped_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
-  track_1->Start();
-
-  // Verify the data flow by connecting the sink to |track_1|.
-  scoped_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));
-  scoped_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
-  track_2->Start();
-
-  // 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));
-  scoped_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
-  track_1->Start();
-
-  // Verify the data flow by connecting the |sink_1| to |track_1|.
-  scoped_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.
-  MockMediaConstraintFactory constraint_factory;
-  StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
-                          std::string(), std::string());
-  scoped_refptr<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));
-  scoped_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
-  track_2->Start();
-
-  // Verify the data flow by connecting the |sink_2| to |track_2|.
-  scoped_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();
+  WebRtcLocalAudioTrack::From(MediaStreamAudioTrack::Get(blink_track_2))
+      ->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.
-  MockMediaConstraintFactory factory;
-  factory.DisableDefaultAudioConstraints();
-  scoped_refptr<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);
+  CreateSourceWithParams(media::AudioParameters(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128));
 
   // Setup a audio track, connect it to the capturer and start it.
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-  scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
-  track->Start();
+  blink::WebMediaStreamTrack blink_track;
+  blink_track.initialize(blink::WebString::fromUTF8("dummy_track"),
+                         blink_source_);
+  ASSERT_TRUE(
+      MediaStreamAudioSource::Get(blink_source_)->ConnectToTrack(blink_track));
+  WebRtcLocalAudioTrack* const track = WebRtcLocalAudioTrack::From(
+      MediaStreamAudioTrack::Get(blink_track));
+  EXPECT_TRUE(track->adapter()->enabled());
 
   // Verify the data flow by connecting the |sink| to |track|.
   scoped_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
+  const int expected_buffer_size = params_.sample_rate() / 100;
   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());
+  EXPECT_CALL(*capturer_source_, OnStop()).Times(AtLeast(1));
+  capturer_->Stop();
 }
 
 }  // namespace content