Update MediaStreamTrack::Stop to latest draft.

content/renderer/media/webrtc_audio_capturer_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/command_line.h"
 #include "base/logging.h"
 #include "content/public/common/content_switches.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"
@@ skipping 78 matching lines @@
         switches::kEnableAudioTrackProcessing);
   }
 
   void VerifyAudioParams(const blink::WebMediaConstraints& constraints,
                          bool need_audio_processing) {
     capturer_ = WebRtcAudioCapturer::CreateCapturer(
         -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
                              "", "", params_.sample_rate(),
                              params_.channel_layout(),
                              params_.frames_per_buffer()),
-        constraints, NULL);
+        constraints, NULL, NULL);
     capturer_source_ = new MockCapturerSource();
     EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1));
+    EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
+    EXPECT_CALL(*capturer_source_.get(), Start());
     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
 
-    EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
-    EXPECT_CALL(*capturer_source_.get(), Start());
     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
     track_.reset(new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
     track_->Start();
 
     // Connect a mock sink to the track.
     scoped_ptr<MockPeerConnectionAudioSink> sink(
         new MockPeerConnectionAudioSink());
     track_->AddSink(sink.get());
 
@@ skipping 49 matching lines @@
 TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) {
   EnableAudioTrackProcessing();
   // Turn off the default constraints to verify that the sink will get packets
   // with a buffer size smaller than 10ms.
   MockMediaConstraintFactory constraint_factory;
   constraint_factory.DisableDefaultAudioConstraints();
   VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false);
 }
 
 }  // namespace content