Update MediaStreamTrack::Stop to latest draft.

content/renderer/media/webrtc_local_audio_track_unittest.cc

Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 8c387bbe3e6c35d9d98515bf8b8b6ec8d2db6467..06e16d89aa3fbae0406c549665b183aa4f0515af 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -4,6 +4,7 @@
 
 #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"
@@ -171,10 +172,17 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
     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());
@@ -182,6 +190,7 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
   }
 
   media::AudioParameters params_;
+  blink::WebMediaStreamSource blink_source_;
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
 };
@@ -328,15 +337,13 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
   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();
 }
 
 
@@ -356,10 +363,39 @@ TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
   // 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) {
@@ -448,7 +484,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   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));
@@ -515,7 +551,7 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
                            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));