Associate audio streams with their source/destination RenderView.

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 22 matching lines @@
   // Called by the client on the sink side. Return false if the capturer has
   // not been initialized successfully.
   void AddCapturerSink(WebRtcAudioCapturerSink* sink);
 
   // Called by the client on the sink side to remove
   void RemoveCapturerSink(WebRtcAudioCapturerSink* sink);
 
   // SetCapturerSource() is called if 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(media::AudioCapturerSource* source);
+  void SetCapturerSource(
+      const scoped_refptr<media::AudioCapturerSource>& source);
 
   // Starts recording audio.
   void Start();
 
   // Stops recording audio.
   void Stop();
 
   // Sets the microphone volume.
   void SetVolume(double volume);
 
@@ skipping 45 matching lines @@
 
   // Protect access to |source_|, |sinks_|, |running_|.
   base::Lock lock_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_