Merge AssociateStreamWithProducer message into CreateStream message for both audio output and input.

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 25 matching lines @@
 // if an alternative AudioCapturerSource has been set.
 class CONTENT_EXPORT WebRtcAudioCapturer
     : public base::RefCountedThreadSafe<WebRtcAudioCapturer>,
       NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
  public:
   // Use to construct the audio capturer.
   // Called on the main render thread.
   static scoped_refptr<WebRtcAudioCapturer> CreateCapturer();
 
   // Creates and configures the default audio capturing source using the
-  // provided audio parameters, |session_id| is passed to the browser to
-  // decide which device to use.
-  // Called on the main render thread.
-  bool Initialize(media::ChannelLayout channel_layout,
+  // provided audio parameters.  |render_view_id| specifies the render view
+  // consuming audio for capture.  |session_id| is passed to the browser to
+  // decide which device to use.  Called on the main render thread.
+  bool Initialize(int render_view_id,
+                  media::ChannelLayout channel_layout,
                   int sample_rate,
                   int session_id);
 
   // Called by the client on the sink side to add a sink.
   // WebRtcAudioDeviceImpl calls this method on the main render thread but
   // other clients may call it from other threads. The current implementation
   // does not support multi-thread calling.
   // Called on the main render thread.
   void AddCapturerSink(WebRtcAudioCapturerSink* sink);
 
@@ skipping 82 matching lines @@
 
   // The media session ID used to identify which input device to be started.
   int session_id_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_