Implicit audio output device selection for getUserMedia.

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 <vector>
 
 #include "base/callback.h"
@@ skipping 30 matching lines @@
 // Tracking this at http://crbug.com/164813.
 class CONTENT_EXPORT WebRtcLocalAudioRenderer
     : NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
       NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) {
  public:
   // Creates a local renderer and registers a capturing |source| object.
   // The |source| is owned by the WebRtcAudioDeviceImpl.
   // Called on the main thread.
   WebRtcLocalAudioRenderer(WebRtcLocalAudioTrack* audio_track,
-                           int source_render_view_id);
+                           int source_render_view_id,
+                           int session_id,
+                           int sample_rate,
+                           int frames_per_buffer);
 
   // MediaStreamAudioRenderer implementation.
   // Called on the main thread.
   virtual void Start() OVERRIDE;
   virtual void Stop() OVERRIDE;
   virtual void Play() OVERRIDE;
   virtual void Pause() OVERRIDE;
   virtual void SetVolume(float volume) OVERRIDE;
   virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
   virtual bool IsLocalRenderer() const OVERRIDE;
@@ skipping 32 matching lines @@
   // The audio track which provides data to render. Given that this class
   // implements local loopback, the audio track is getting data from a capture
   // instance like a selected microphone and forwards the recorded data to its
   // sinks. The recorded data is stored in a FIFO and consumed
   // by this class when the sink asks for new data.
   // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
   scoped_refptr<WebRtcLocalAudioTrack> audio_track_;
 
   // 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_;
 
   // 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.
   base::Time last_render_time_;
 
   // Keeps track of total time audio has been rendered.
   base::TimeDelta total_render_time_;
 
   // The audio parameters used by the renderer.
   media::AudioParameters audio_params_;
 
   // Set when playing, cleared when paused.
   bool playing_;
 
   // Protects |loopback_fifo_|, |playing_| and |sink_|.
   mutable base::Lock thread_lock_;
 
+  // The preferred sample rate and buffer sizes provided via the ctor.
+  const int sample_rate_;
+  const int frames_per_buffer_;
+
+  // The preferred device id of the output device or empty for the default
+  // output device.
+  const std::string output_device_id_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_