Fix the race when the WebRtcAudioDeviceImpl goes away before capturers stop

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/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
@@ skipping 190 matching lines @@
 // 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, capturer_, NULL));
   track->Start();
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 
-  // Connect a number of network channels to the audio track.
-  static const int kNumberOfNetworkChannels = 4;
-  for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
-    static_cast<webrtc::AudioTrackInterface*>(
-        adapter.get())->GetRenderer()->AddChannel(i);
-  }
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, FormatIsSet());
   EXPECT_CALL(*sink,
-      CaptureData(kNumberOfNetworkChannels,
+      CaptureData(0,
                   0,
                   0,
                   _,
                   false)).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, capturer_, NULL));
   track->Start();
-  static_cast<webrtc::AudioTrackInterface*>(
-      adapter.get())->GetRenderer()->AddChannel(0);
   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();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
   EXPECT_CALL(*sink,
-              CaptureData(1, 0, 0, _, false)).Times(0);
+              CaptureData(0, 0, 0, _, false)).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(1, 0, 0, _, false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink, CaptureData(0, 0, 0, _, false)).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, capturer_, NULL));
   track_1->Start();
-  static_cast<webrtc::AudioTrackInterface*>(
-      adapter_1.get())->GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track_1->GetAudioAdapter()->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(1, 0, 0, _, false)).Times(AtLeast(1))
+      CaptureData(0, 0, 0, _, false)).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, capturer_, NULL));
   track_2->Start();
-  static_cast<webrtc::AudioTrackInterface*>(
-      adapter_2.get())->GetRenderer()->AddChannel(1);
   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(1, 0, 0, _, false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink_1, CaptureData(0, 0, 0, _, false)).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(1, 0, 0, _, false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false)).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();
@@ skipping 44 matching lines @@
 }
 
 // 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, capturer_, NULL));
-  static_cast<webrtc::AudioTrackInterface*>(
-      adapter_1.get())->GetRenderer()->AddChannel(0);
   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(_, 0, 0, _, false))
       .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, capturer_, NULL));
   track_2->Start();
-  static_cast<webrtc::AudioTrackInterface*>(
-      adapter_2.get())->GetRenderer()->AddChannel(1);
 
   // 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.
 TEST_F(WebRtcLocalAudioTrackTest, 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, capturer_, NULL));
   track_1->Start();
 
-  // Connect a number of network channels to the |track_1|.
-  static const int kNumberOfNetworkChannelsForTrack1 = 2;
-  for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
-    static_cast<webrtc::AudioTrackInterface*>(
-        adapter_1.get())->GetRenderer()->AddChannel(i);
-  }
   // 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(kNumberOfNetworkChannelsForTrack1,
-                          0, 0, _, false))
+  EXPECT_CALL(*sink_1.get(), CaptureData(0, 0, 0, _, false))
       .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(
@@ skipping 10 matching lines @@
       media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
   new_capturer->SetCapturerSourceForTesting(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, new_capturer, NULL));
   track_2->Start();
 
-  // Connect a number of network channels to the |track_2|.
-  static const int kNumberOfNetworkChannelsForTrack2 = 3;
-  for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
-    static_cast<webrtc::AudioTrackInterface*>(
-        adapter_2.get())->GetRenderer()->AddChannel(i);
-  }
   // 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(kNumberOfNetworkChannelsForTrack2, 0, 0, _, false))
+  EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false))
       .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();
@@ skipping 57 matching lines @@
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source, 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