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

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 "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "webkit/media/media_stream_audio_renderer.h"
 
 namespace media {
 class AudioBus;
+class AudioOutputDevice;
 class AudioParameters;
 }
 
 namespace webrtc {
 class AudioTrackInterface;
 }
 
 namespace content {
 
-class RendererAudioOutputDevice;
 class WebRtcAudioCapturer;
 
 // WebRtcLocalAudioRenderer is a webkit_media::MediaStreamAudioRenderer
 // designed for rendering local audio media stream tracks,
 // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack
 // It also implements media::AudioRendererSink::RenderCallback to render audio
 // data provided from a WebRtcAudioCapturer source which is set at construction.
 // When the audio layer in the browser process asks for data to render, this
 // class provides the data by implementing the WebRtcAudioCapturerSink
 // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective.
@@ skipping 59 matching lines @@
   // by this class when the sink asks for new data.
   // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
   scoped_refptr<WebRtcAudioCapturer> source_;
 
   scoped_refptr<webrtc::AudioTrackInterface> audio_track_;
 
   // 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_;
 
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   // Contains copies of captured audio frames.
   scoped_ptr<media::AudioFifo> loopback_fifo_;
 
   // 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.
@@ skipping 10 matching lines @@
 
   // Protects |loopback_fifo_|, |playing_| and |sink_|.
   mutable base::Lock thread_lock_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_