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

content/renderer/media/webrtc_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_audio_renderer.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/renderer_audio_output_device.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_thread_impl.h"
+#include "media/audio/audio_output_device.h"
 #include "media/audio/audio_parameters.h"
 #include "media/audio/sample_rates.h"
 #include "media/base/audio_hardware_config.h"
 
 #if defined(OS_WIN)
 #include "base/win/windows_version.h"
 #include "media/audio/win/core_audio_util_win.h"
 #endif
 
 namespace content {
@@ skipping 80 matching lines @@
 
 bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
   DVLOG(1) << "WebRtcAudioRenderer::Initialize()";
   DCHECK(thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   DCHECK_EQ(state_, UNINITIALIZED);
   DCHECK(source);
   DCHECK(!sink_);
   DCHECK(!source_);
 
-  sink_ = AudioDeviceFactory::NewOutputDevice();
-  DCHECK(sink_);
-
   // Use mono on all platforms but Windows for now.
   // TODO(henrika): Tracking at http://crbug.com/166771.
   media::ChannelLayout channel_layout = media::CHANNEL_LAYOUT_MONO;
 #if defined(OS_WIN)
   channel_layout = media::CHANNEL_LAYOUT_STEREO;
 #endif
 
   // Ask the renderer for the default audio output hardware sample-rate.
   media::AudioHardwareConfig* hardware_config =
       RenderThreadImpl::current()->GetAudioHardwareConfig();
@@ skipping 72 matching lines @@
   // It is assumed that each audio sample contains 16 bits and each
   // audio frame contains one or two audio samples depending on the
   // number of channels.
   buffer_.reset(
       new int16[source_params.frames_per_buffer() * source_params.channels()]);
 
   source_ = source;
   source->SetRenderFormat(source_params);
 
   // Configure the audio rendering client and start rendering.
+  sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_);
   sink_->Initialize(sink_params, this);
-  sink_->SetSourceRenderView(source_render_view_id_);
   sink_->Start();
 
   // User must call Play() before any audio can be heard.
   state_ = PAUSED;
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputChannelLayout",
                             source_params.channel_layout(),
                             media::CHANNEL_LAYOUT_MAX);
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
                             source_params.frames_per_buffer(),
@@ skipping 123 matching lines @@
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(buffer_.get(),
                              audio_bus->frames(),
                              sizeof(buffer_[0]));
 }
 
 }  // namespace content