Avoid hitting the thread check when WebRtcAudioRenderer is going away.

content/renderer/media/webrtc_audio_device_impl.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 221 matching lines @@
 
 // TODO(xians): Merge this interface with WebRtcAudioRendererSource.
 // The reason why we could not do it today is that WebRtcAudioRendererSource
 // gets the data by pulling, while the data is pushed into
 // WebRtcPlayoutDataSource::Sink.
 class WebRtcPlayoutDataSource {
  public:
   class Sink {
    public:
     // Callback to get the playout data.
+    // Called on the render audio thread.
     virtual void OnPlayoutData(media::AudioBus* audio_bus,
                                int sample_rate,
                                int audio_delay_milliseconds) = 0;
+
+    // Callback to notify the sink that the source has changed.
+    // Called on the main render thread.
+    virtual void OnPlayoutDataSourceChanged() = 0;
+
    protected:
     virtual ~Sink() {}
   };
 
   // Adds/Removes the sink of WebRtcAudioRendererSource to the ADM.
   // These methods are used by the MediaStreamAudioProcesssor to get the
   // rendered data for AEC.
   virtual void AddPlayoutSink(Sink* sink) = 0;
   virtual void RemovePlayoutSink(Sink* sink) = 0;
 
@@ skipping 175 matching lines @@
   // Buffer used for temporary storage during render callback.
   // It is only accessed by the audio render thread.
   std::vector<int16> render_buffer_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_