Update MediaStreamTrack::Stop to latest draft.

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"
 #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"
@@ skipping 147 matching lines @@
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
     blink::WebMediaConstraints constraints;
+    blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
+                             "dummy");
+    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,
-                                                    constraints, NULL);
+                                                    constraints, NULL,
+                                                    audio_source);
+    audio_source->SetAudioCapturer(capturer_);
     capturer_source_ = new MockCapturerSource(capturer_.get());
     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
         .WillOnce(Return());
     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
   }
 
   media::AudioParameters params_;
+  blink::WebMediaStreamSource 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) {
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
@@ skipping 126 matching lines @@
             params.sample_rate() / 100);
   EXPECT_CALL(*sink_2, CaptureData(1, 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->StopTrack();
   track_1.reset();
 
   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
   track_2->RemoveSink(sink_2.get());
-  track_2->Stop();
+  track_2->StopTrack();
   track_2.reset();
-
-  capturer_->Stop();
 }
 
 
 // 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 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) {
+  EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
+  EXPECT_CALL(*capturer_source_.get(), OnStart());
+  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
+      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+  scoped_ptr<WebRtcLocalAudioTrack> track1(
+      new WebRtcLocalAudioTrack(adapter1, capturer_, NULL));
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track1->audio_source_provider())->SetSinkParamsForTesting(params_);
+  track1->Start();
+
+  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
+        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+  scoped_ptr<WebRtcLocalAudioTrack> track2(
+      new WebRtcLocalAudioTrack(adapter2, capturer_, NULL));
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track2->audio_source_provider())->SetSinkParamsForTesting(params_);
+  track2->Start();
+
+  track1->StopTrack();
+  // When the last track is stopped, it will automatically stop the
+  // |capturer_source_|.
+  EXPECT_CALL(*capturer_source_.get(), OnStop());
+  track2->StopTrack();
+}
+
+
 // 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(
@@ skipping 68 matching lines @@
                           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.
   blink::WebMediaConstraints constraints;
   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
                           std::string(), std::string());
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
-      WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL));
+      WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL, NULL));
   scoped_refptr<MockCapturerSource> new_source(
       new MockCapturerSource(new_capturer.get()));
   EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
   media::AudioParameters new_param(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
       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));
@@ skipping 46 matching lines @@
   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, NULL));
   scoped_refptr<MockCapturerSource> source(
       new MockCapturerSource(capturer.get()));
   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));
@@ skipping 19 matching lines @@
   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