Merge AssociateStreamWithProducer message into CreateStream message for both audio output and input.

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 "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.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 "webkit/media/media_stream_audio_renderer.h"
 
+namespace media {
+class AudioOutputDevice;
+}
+
 namespace content {
 
-class RendererAudioOutputDevice;
 class WebRtcAudioRendererSource;
 
 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
 // for connecting WebRtc MediaStream with the audio pipeline.
 class CONTENT_EXPORT WebRtcAudioRenderer
     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public webkit_media::MediaStreamAudioRenderer) {
  public:
   explicit WebRtcAudioRenderer(int source_render_view_id);
 
@@ skipping 34 matching lines @@
   virtual void OnRenderError() OVERRIDE;
 
   // Called by AudioPullFifo when more data is necessary.
   // This method is called on the AudioOutputDevice worker thread.
   void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus);
 
   // The render view in which the audio is rendered into |sink_|.
   const int source_render_view_id_;
 
   // The sink (destination) for rendered audio.
-  scoped_refptr<RendererAudioOutputDevice> sink_;
+  scoped_refptr<media::AudioOutputDevice> sink_;
 
   // Audio data source from the browser process.
   WebRtcAudioRendererSource* source_;
 
   // Buffers used for temporary storage during render callbacks.
   // Allocated during initialization.
   scoped_ptr<int16[]> buffer_;
 
   // Protects access to |state_|, |source_| and |sink_|.
   base::Lock lock_;
@@ skipping 13 matching lines @@
   double frame_duration_milliseconds_;
 
   double fifo_io_ratio_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_