Associate audio streams with their source/destination RenderView.

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/audio_hardware.h"
+#include "content/renderer/media/audio_message_filter.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_util.h"
 #include "media/audio/sample_rates.h"
 #if defined(OS_WIN)
 #include "media/audio/win/core_audio_util_win.h"
 #endif
 
 namespace content {
 
 namespace {
 
@@ skipping 50 matching lines @@
     UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
                               afpb, kUnexpectedAudioBufferSize);
   } else {
     // Report unexpected sample rates using a unique histogram name.
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputFramesPerBufferUnexpected", param);
   }
 }
 
 }  // namespace
 
-WebRtcAudioRenderer::WebRtcAudioRenderer()
+WebRtcAudioRenderer::WebRtcAudioRenderer(int source_render_view_id)
     : state_(UNINITIALIZED),
+      source_render_view_id_(source_render_view_id),
       source_(NULL) {
 }
 
 WebRtcAudioRenderer::~WebRtcAudioRenderer() {
   DCHECK_EQ(state_, UNINITIALIZED);
   buffer_.reset();
 }
 
 bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
   base::AutoLock auto_lock(lock_);
   DCHECK_EQ(state_, UNINITIALIZED);
   DCHECK(source);
   DCHECK(!sink_);
   DCHECK(!source_);
 
-  sink_ = AudioDeviceFactory::NewOutputDevice();
-  DCHECK(sink_);
-
   // Ask the browser for the default audio output hardware sample-rate.
   // This request is based on a synchronous IPC message.
   int sample_rate = GetAudioOutputSampleRate();
   DVLOG(1) << "Audio output hardware sample rate: " << sample_rate;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputSampleRate",
                             sample_rate, media::kUnexpectedAudioSampleRate);
 
   // Verify that the reported output hardware sample rate is supported
   // on the current platform.
   if (std::find(&kValidOutputRates[0],
@@ skipping 70 matching lines @@
   // Allocate local audio buffers based on the parameters above.
   // 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[params_.frames_per_buffer() * params_.channels()]);
 
   source_ = source;
   source->SetRenderFormat(params_);
 
   // Configure the audio rendering client and start the rendering.
+  scoped_refptr<content::AudioMessageFilter> audio_message_filter =
+      RenderThreadImpl::current()->audio_message_filter();
+  sink_ = new media::AudioOutputDevice(
+      audio_message_filter,
+      RenderThreadImpl::current()->GetIOMessageLoopProxy());
   sink_->Initialize(params_, this);
-
+  audio_message_filter->AssociateStreamWithProducer(sink_->stream_id(),
+                                                    source_render_view_id_);
   sink_->Start();
 
   state_ = PAUSED;
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX);
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
                             buffer_size, kUnexpectedAudioBufferSize);
   AddHistogramFramesPerBuffer(buffer_size);
 
@@ skipping 58 matching lines @@
                              params_.bits_per_sample() / 8);
   return audio_bus->frames();
 }
 
 void WebRtcAudioRenderer::OnRenderError() {
   NOTIMPLEMENTED();
   LOG(ERROR) << "OnRenderError()";
 }
 
 }  // namespace content