MediaStream audio object graph untangling and clean-ups.

content/renderer/media/webrtc_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"
@@ skipping 77 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 {}
   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 49 matching lines @@
                   media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480);
     MockConstraintFactory 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))
+    {
+      scoped_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_;
-  scoped_refptr<WebRtcAudioCapturer> capturer_;
 };
 
 // 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();
+      new WebRtcLocalAudioTrack(adapter.get()));
+  capturer_->AddTrack(track.get());
   EXPECT_TRUE(track->GetAudioAdapter()->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();
+      new WebRtcLocalAudioTrack(adapter.get()));
+  capturer_->AddTrack(track.get());
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
-  const media::AudioParameters params = capturer_->source_audio_parameters();
+  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));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
-  track_1->Start();
+      new WebRtcLocalAudioTrack(adapter_1.get()));
+  capturer_->AddTrack(track_1.get());
   EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
-  const media::AudioParameters params = capturer_->source_audio_parameters();
+  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));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
-  track_2->Start();
+      new WebRtcLocalAudioTrack(adapter_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);
 
   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));
@@ skipping 17 matching lines @@
   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();
+      new WebRtcLocalAudioTrack(adapter.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));
   scoped_ptr<WebRtcLocalAudioTrack> track1(
-      new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL));
-  track1->Start();
+      new WebRtcLocalAudioTrack(adapter1.get()));
+  capturer_->AddTrack(track1.get());
 
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track2(
-      new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL));
-  track2->Start();
+      new WebRtcLocalAudioTrack(adapter2.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));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
-  track_1->Start();
+      new WebRtcLocalAudioTrack(adapter_1.get()));
+  capturer_->AddTrack(track_1.get());
 
   // 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();
+      new WebRtcLocalAudioTrack(adapter_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
@@ skipping 10 matching lines @@
     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();
+      new WebRtcLocalAudioTrack(adapter_1.get()));
+  capturer_->AddTrack(track_1.get());
 
   // 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.
   MockConstraintFactory constraint_factory;
   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
                           std::string(), std::string());
-  scoped_refptr<WebRtcAudioCapturer> new_capturer(
+  scoped_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));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
-  track_2->Start();
+      new WebRtcLocalAudioTrack(adapter_2.get()));
+  new_capturer->AddTrack(track_2.get());
 
   // 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()));
 
@@ skipping 12 matching lines @@
 // 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();
-  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_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));
   scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
-  track->Start();
+      new WebRtcLocalAudioTrack(adapter.get()));
+  capturer->AddTrack(track.get());
 
   // 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
   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