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/atomicops.h"
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/command_line.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/bind_to_loop.h"
 #include "media/base/media_switches.h"
 
 namespace content {
 
 // WebRtcLocalAudioRenderer::AudioCallback wraps the AudioOutputDevice thread,
 // receives callbacks on that thread and proxies these requests to the capture
 // sink.
 class WebRtcLocalAudioRenderer::AudioCallback
     : public media::AudioRendererSink::RenderCallback {
@@ skipping 134 matching lines @@
   // tests have shown that it resolves issues with audio glitches for some
   // cases where resampling is needed on the output side.
   // TODO(henrika): verify this scheme on as many different devices and
   // combinations of sample rates as possible
   media::AudioParameters source_params = source_->audio_parameters();
   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_);
   if (CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kEnableWebAudioInput)) {
     // 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.
     DVLOG(1) << "enable-webaudio-input command-line flag is enabled";
   }
   sink_->Initialize(sink_params, callback_.get());
-  sink_->SetSourceRenderView(source_render_view_id_);
 
   // Start local rendering and the capturer. Note that, the capturer is owned
   // by the WebRTC ADM and might already bee running.
   source_->Start();
   sink_->Start();
   callback_->Start();
 }
 
 void WebRtcLocalAudioRenderer::Stop() {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 69 matching lines @@
   DVLOG(1) << "WebRtcLocalAudioRenderer::OnSourceCaptureDeviceStopped()";
   if (!Started())
     return;
 
   // The capture device has stopped and we should therefore stop all activity
   // as well to save resources.
   Stop();
 }
 
 }  // namespace content