Associate audio streams with their source/destination RenderView.

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 "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_decoder.h"
 #include "media/base/audio_renderer_sink.h"
 #include "webkit/media/media_stream_audio_renderer.h"
 
 namespace content {
 
+class RendererAudioOutputDevice;
 class WebRtcAudioRendererSource;
 
 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
 // for connecting WebRtc MediaStream with pipeline.
 class CONTENT_EXPORT WebRtcAudioRenderer
     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public webkit_media::MediaStreamAudioRenderer) {
  public:
-  WebRtcAudioRenderer();
+  explicit WebRtcAudioRenderer(int source_render_view_id);
 
   // Initialize function called by clients like WebRtcAudioDeviceImpl. Note,
   // Stop() has to be called before |source| is deleted.
   // Returns false if Initialize() fails.
   bool Initialize(WebRtcAudioRendererSource* source);
 
   // Methods called by WebMediaPlayerMS and WebRtcAudioDeviceImpl.
   // MediaStreamAudioRenderer implementation.
   virtual void Play() OVERRIDE;
   virtual void Pause() OVERRIDE;
@@ skipping 10 matching lines @@
     PAUSED,
   };
   // Flag to keep track the state of the renderer.
   State state_;
 
   // media::AudioRendererSink::RenderCallback implementation.
   virtual int Render(media::AudioBus* audio_bus,
                      int audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
+  // 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<media::AudioRendererSink> sink_;
+  scoped_refptr<RendererAudioOutputDevice> sink_;
 
   // Audio data source from the browser process.
   WebRtcAudioRendererSource* source_;
 
   // Cached values of utilized audio parameters. Platform dependent.
   media::AudioParameters params_;
 
   // Buffers used for temporary storage during render callbacks.
   // Allocated during initialization.
   scoped_array<int16> buffer_;
 
   // Protect access to |state_|.
   base::Lock lock_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioRenderer);

[scherkus (not reviewing), 2012/11/30 22:09:43]

ditto

[miu, 2012/12/01 00:40:17]

Done.

 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_