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..2dcce6c275c4f467a7428c31347517b1b1067f37 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -174,7 +174,7 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
     StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
                             std::string(), std::string());
     capturer_ = WebRtcAudioCapturer::CreateCapturer(-1, device,
-                                                    constraints, NULL);
+                                                    constraints, NULL, NULL);
     capturer_source_ = new MockCapturerSource(capturer_.get());
     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
         .WillOnce(Return());
@@ -328,12 +328,12 @@ 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();

[no longer working on chromium, 2014/04/01 18:29:58]

I think you should skip this capturer_->Stop(); but try to verify that the
source has been stopped by the tracks' stop()

[perkj_chrome, 2014/04/02 13:35:49]

Do you have any plans to fix http://crbug.com/295418 and enable this?
This test fails long before you reach here.

I added StartTwoAudioTracks and modified StartOneAudioTrack to test this.

@@ -448,7 +448,7 @@ 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 +515,7 @@ 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));