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/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"
@@ skipping 156 matching lines @@
 
 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;
     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());
     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
   }
 
   media::AudioParameters params_;
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
 };
@@ skipping 133 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();

[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.

 }
 
 
 // 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));
@@ skipping 91 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