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 34 matching lines @@
       NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback),
       NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureEventHandler),
       NON_EXPORTED_BASE(
           public content::WebRtcLocalAudioRenderer::LocalRenderCallback) {
  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.
-  // Called on the main render thread.
-  bool Initialize(media::ChannelLayout channel_layout, int sample_rate);
+  // provided audio parameters.  |render_view_id| specifies the render view
+  // consuming audio for capture.  Called on the main render thread.
+  bool Initialize(int render_view_id,
+                  media::ChannelLayout channel_layout, int sample_rate);
 
   // 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.
   // TODO(henrika): add lock if we extend number of supported sinks.
   // Called on the main render thread.
   void AddCapturerSink(WebRtcAudioCapturerSink* sink);
 
   // Called by the client on the sink side to remove a sink.
@@ skipping 137 matching lines @@
 
   // True when automatic gain control is enabled, false otherwise.
   bool agc_is_enabled_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_