Notify the track before source provider goes away.

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/mock_media_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_audio_device_impl.h"
-#include "content/renderer/media/webrtc_local_audio_source_provider.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/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 using ::testing::_;
@@ skipping 168 matching lines @@
 // 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) {
   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));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track->audio_source_provider())->SetSinkParamsForTesting(params_);
   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());
@@ skipping 20 matching lines @@
 // 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));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track->audio_source_provider())->SetSinkParamsForTesting(params_);
   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,
@@ skipping 19 matching lines @@
 // 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) {
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*capturer_source_.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
     new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   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))
       .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));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   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());
@@ skipping 25 matching lines @@
 
 // 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) {
   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));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
 
   // When the track goes away, it will automatically stop the
   // |capturer_source_|.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
   capturer_->Stop();
   track.reset();
 }
 
 // Start/Stop tracks and verify the capturer is correctly starting/stopping
 // its source.
 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   // Starting the first audio track will start the |capturer_source_|.
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*capturer_source_.get(), OnStart()).WillOnce(SignalEvent(&event));
   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);
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // 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));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   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.
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*capturer_source_.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   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());
@@ skipping 18 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.
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*new_source.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
       new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   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());
@@ skipping 42 matching lines @@
   EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
   capturer->SetCapturerSourceForTesting(source, params);
 
   // Setup a audio track, connect it to the capturer and start it.
   EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*source.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
       new WebRtcLocalAudioTrack(adapter, capturer, NULL));
-  static_cast<WebRtcLocalAudioSourceProvider*>(
-      track->audio_source_provider())->SetSinkParamsForTesting(params);
   track->Start();
 
   // 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(
       0, 0, 0, _, false))
       .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, OnStop()).Times(1);
   capturer->Stop();
 }
 
 }  // namespace content