Associate audio streams with their source/destination RenderView.

content/renderer/media/webrtc_audio_capturer.cc

 // 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.
 
 #include "content/renderer/media/webrtc_audio_capturer.h"
 
 #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"
 
 namespace content {
 
 // Supported hardware sample rates for input and output sides.
 #if defined(OS_WIN) || defined(OS_MACOSX)
 // media::GetAudioInputHardwareSampleRate() asks the audio layer
 // for its current sample rate (set by the user) on Windows and Mac OS X.
 // The listed rates below adds restrictions and WebRtcAudioDeviceImpl::Init()
@@ skipping 65 matching lines @@
   base::AutoLock auto_lock(lock_);
   for (SinkList::iterator it = sinks_.begin(); it != sinks_.end(); ++it) {
     if (sink == *it) {
       sinks_.erase(it);
       break;
     }
   }
 }
 
 void WebRtcAudioCapturer::SetCapturerSource(
-    media::AudioCapturerSource* source) {
+    scoped_refptr<media::AudioCapturerSource> source) {
   DVLOG(1) <<  "SetCapturerSource()";
   scoped_refptr<media::AudioCapturerSource> old_source;
   {
     base::AutoLock auto_lock(lock_);
     if (source_ == source)
       return;
 
     source_.swap(old_source);
     source_ = source;
   }
@@ skipping 39 matching lines @@
     return false;
   }
 
   params_.Reset(format, channel_layout, sample_rate, 16, buffer_size);
 
   buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]);
 
   // 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);
 
   return true;
 }
 
 void WebRtcAudioCapturer::Start() {
   base::AutoLock auto_lock(lock_);
   if (running_)
@@ skipping 79 matching lines @@
 
 void WebRtcAudioCapturer::OnDeviceStarted(const std::string& device_id) {
   device_id_ = device_id;
 }
 
 void WebRtcAudioCapturer::OnDeviceStopped() {
   NOTIMPLEMENTED();
 }
 
 }  // namespace content