Part 1: Plumb render view ID to render host

content/renderer/media/webrtc_audio_device_impl.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_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 191 matching lines @@
 //    (WebRTC client a media layer). This approach ensures that we can avoid
 //    transferring maximum levels between the renderer and the browser.
 //
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule),
       NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureCallback),
       NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureEventHandler) {
  public:
   // Methods called on main render thread.
-  WebRtcAudioDeviceImpl();
+  WebRtcAudioDeviceImpl(int render_view_id);
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
   // to release the reference and delete this object.
   virtual int32_t AddRef() OVERRIDE;
   virtual int32_t Release() OVERRIDE;
 
   // We need this one to support runnable method tasks.
   static bool ImplementsThreadSafeReferenceCounting() { return true; }
 
@@ skipping 227 matching lines @@
   bool agc_is_enabled_;
 
   // Used for histograms of total recording and playout times.
   base::Time start_capture_time_;
   base::Time start_render_time_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_