Switching audio clients to using RestartableAudioRendererSink interface as a sink.

content/renderer/media/webrtc_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_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/non_thread_safe.h"
 #include "base/threading/thread_checker.h"
 #include "content/public/renderer/media_stream_audio_renderer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_decoder.h"
 #include "media/base/audio_pull_fifo.h"
 #include "media/base/audio_renderer_sink.h"
 #include "media/base/channel_layout.h"
 #include "media/base/output_device.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 
-namespace media {
-class AudioOutputDevice;
-}  // namespace media
-
 namespace webrtc {
 class AudioSourceInterface;
 }  // namespace webrtc
 
 namespace content {
 
 class WebRtcAudioRendererSource;
 
 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
 // for connecting WebRtc MediaStream with the audio pipeline.
@@ skipping 165 matching lines @@
   // |sink_|.
   void PrepareSink();
 
   // The RenderFrame in which the audio is rendered into |sink_|.
   const int source_render_frame_id_;
   const int session_id_;
 
   const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_;
 
   // The sink (destination) for rendered audio.
-  scoped_refptr<media::AudioOutputDevice> sink_;
+  scoped_refptr<media::AudioRendererSink> sink_;
 
   // The media stream that holds the audio tracks that this renderer renders.
   const blink::WebMediaStream media_stream_;
 
   // Audio data source from the browser process.
   WebRtcAudioRendererSource* source_;
 
   // Protects access to |state_|, |source_|, |audio_fifo_|,
   // |audio_delay_milliseconds_|, |fifo_delay_milliseconds_|, |current_time_|,
   // |sink_params_| and |render_callback_count_|
@@ skipping 36 matching lines @@
   // Used for triggering new UMA histogram. Counts number of render
   // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|.
   int render_callback_count_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_