Associate audio streams with their source/destination RenderView.

content/renderer/media/webrtc_audio_capturer.cc

Index: content/renderer/media/webrtc_audio_capturer.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 8340ae827f3c711a3f2e4239ffb147a72e296d3c..403e5d776aca4cb5a414456c4b99cb9024e728d4 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -7,9 +7,11 @@
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
-#include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/audio_hardware.h"
+#include "content/renderer/media/audio_input_message_filter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/render_thread_impl.h"
+#include "media/audio/audio_input_device.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/sample_rates.h"
 
@@ -95,7 +97,7 @@ void WebRtcAudioCapturer::RemoveCapturerSink(WebRtcAudioCapturerSink* sink) {
 }
 
 void WebRtcAudioCapturer::SetCapturerSource(
-    media::AudioCapturerSource* source) {
+    scoped_refptr<media::AudioCapturerSource> source) {
   DVLOG(1) <<  "SetCapturerSource()";
   scoped_refptr<media::AudioCapturerSource> old_source;
   {
@@ -155,7 +157,9 @@ bool WebRtcAudioCapturer::Initialize() {
   // Create and configure the default audio capturing source. The |source_|
   // will be overwritten if the client call the source calls
   // SetCapturerSource().
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice());
+  SetCapturerSource(new media::AudioInputDevice(
+      RenderThreadImpl::current()->audio_input_message_filter(),
+      RenderThreadImpl::current()->GetIOMessageLoopProxy()));
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX);