Implicit audio output device selection for getUserMedia.

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/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"
 
 namespace media {
 class AudioOutputDevice;
 }
 
 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.
 class CONTENT_EXPORT WebRtcAudioRenderer
     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public MediaStreamAudioRenderer) {
  public:
-  explicit WebRtcAudioRenderer(int source_render_view_id);
+  WebRtcAudioRenderer(int source_render_view_id,
+                      int session_id,
+                      int sample_rate,
+                      int frames_per_buffer);
 
   // Initialize function called by clients like WebRtcAudioDeviceImpl.
   // Stop() has to be called before |source| is deleted.
   bool Initialize(WebRtcAudioRendererSource* source);
 
   // Methods called by WebMediaPlayerMS and WebRtcAudioDeviceImpl.
   // MediaStreamAudioRenderer implementation.
   virtual void Start() OVERRIDE;
   virtual void Play() OVERRIDE;
   virtual void Pause() OVERRIDE;
@@ skipping 23 matching lines @@
   virtual int Render(media::AudioBus* audio_bus,
                      int audio_delay_milliseconds) OVERRIDE;
   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_;
+  const int session_id_;
 
   // The sink (destination) for rendered audio.
   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_;
 
   // Ref count for the MediaPlayers which are playing audio.
   int play_ref_count_;
 
   // Used to buffer data between the client and the output device in cases where
   // the client buffer size is not the same as the output device buffer size.
   scoped_ptr<media::AudioPullFifo> audio_fifo_;
 
   // Contains the accumulated delay estimate which is provided to the WebRTC
   // AEC.
   int audio_delay_milliseconds_;
 
   // Delay due to the FIFO in milliseconds.
   int fifo_delay_milliseconds_;
 
+  // The preferred sample rate and buffer sizes provided via the ctor.
+  const int sample_rate_;
+  const int frames_per_buffer_;
+
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_