Switching audio clients to using RestartableAudioRendererSink interface as a sink.

content/renderer/media/webrtc_local_audio_renderer.h

 // Copyright (c) 2012 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_LOCAL_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <string>
 #include <vector>
 
 #include "base/callback.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
 #include "content/public/renderer/media_stream_audio_renderer.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
+#include "media/base/audio_renderer_sink.h"
 #include "media/base/output_device.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 
 namespace media {
 class AudioBus;
 class AudioShifter;
-class AudioOutputDevice;
 class AudioParameters;
 }
 
 namespace content {
 
 class WebRtcAudioCapturer;
 
 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering
 // local audio media stream tracks,
 // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack
@@ skipping 85 matching lines @@
 
   // The render view and frame in which the audio is rendered into |sink_|.
   const int source_render_frame_id_;
   const int session_id_;
 
   // MessageLoop associated with the single thread that performs all control
   // tasks.  Set to the MessageLoop that invoked the ctor.
   const scoped_refptr<base::SingleThreadTaskRunner> task_runner_;
 
   // The sink (destination) for rendered audio.
-  scoped_refptr<media::AudioOutputDevice> sink_;
+  scoped_refptr<media::AudioRendererSink> sink_;
 
   // This does all the synchronization/resampling/smoothing.
   scoped_ptr<media::AudioShifter> audio_shifter_;
 
   // Stores last time a render callback was received. The time difference
   // between a new time stamp and this value can be used to derive the
   // total render time.
   base::TimeTicks last_render_time_;
 
   // Keeps track of total time audio has been rendered.
@@ skipping 27 matching lines @@
 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_