Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/webrtc_audio_capturer.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_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
@@ skipping 40 matching lines @@
   // Called by the client on the sink side to add a sink.
   void AddCapturerSink(WebRtcAudioCapturerSink* sink);
 
   // Called by the client on the sink side to remove a sink.
   void RemoveCapturerSink(WebRtcAudioCapturerSink* sink);
 
   // SetCapturerSource() is called if the client on the source side desires to
   // provide their own captured audio data. Client is responsible for calling
   // Start() on its own source to have the ball rolling.
   void SetCapturerSource(
-      const scoped_refptr<media::AudioCapturerSource>& source);
+      const scoped_refptr<media::AudioCapturerSource>& source,
+      media::ChannelLayout channel_layout,
+      float sample_rate);
 
   // The |on_device_stopped_cb| callback will be called in OnDeviceStopped().
   void SetStopCallback(const base::Closure& on_device_stopped_cb);
 
   // Informs this class that a local sink shall be used in addition to the
   // registered WebRtcAudioCapturerSink sink(s). The capturer will enter a
   // buffering mode and store all incoming audio frames in a local FIFO.
   // The renderer will read data from this buffer using the ProvideInput()
   // method. Called on the main render thread.
   void PrepareLoopback();
@@ skipping 99 matching lines @@
 
   // True when FIFO is utilized, false otherwise.
   bool buffering_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_