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

content/renderer/media/webrtc_local_audio_renderer.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_local_audio_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop_proxy.h"
 #include "base/synchronization/lock.h"
 #include "content/renderer/media/audio_device_factory.h"
-#include "content/renderer/media/renderer_audio_output_device.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
+#include "media/audio/audio_output_device.h"
 #include "media/base/audio_bus.h"
 
 namespace content {
 
 // media::AudioRendererSink::RenderCallback implementation
 int WebRtcLocalAudioRenderer::Render(
     media::AudioBus* audio_bus, int audio_delay_milliseconds) {
   base::AutoLock auto_lock(thread_lock_);
 
   if (!playing_) {
@@ skipping 127 matching lines @@
   DCHECK(!loopback_fifo_);
   loopback_fifo_.reset(new media::AudioFifo(
       source_params.channels(),
       10 * source_params.frames_per_buffer()));
 
   media::AudioParameters sink_params(source_params.format(),
                                      source_params.channel_layout(),
                                      source_params.sample_rate(),
                                      source_params.bits_per_sample(),
                                      2 * source_params.frames_per_buffer());
-  sink_ = AudioDeviceFactory::NewOutputDevice();
+  sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_);
   // TODO(henrika): we could utilize the unified audio here instead and do
   // sink_->InitializeIO(sink_params, 2, callback_.get());
   // It would then be possible to avoid using the WebRtcAudioCapturer.
   sink_->Initialize(sink_params, this);
-  sink_->SetSourceRenderView(source_render_view_id_);
 
   // Start the capturer and local rendering. Note that, the capturer is owned
   // by the WebRTC ADM and might already bee running.
   source_->Start();
   sink_->Start();
 
   last_render_time_ = base::Time::Now();
   playing_ = false;
 }
 
@@ skipping 75 matching lines @@
   if (!sink_)
     return base::TimeDelta();
   return total_render_time();
 }
 
 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const {
   return true;
 }
 
 }  // namespace content