Plumb render view ID from audio-related code in renderer through IPCs to AudioRendererHost in brows…

content/renderer/media/webrtc_audio_device_impl.cc

Index: content/renderer/media/webrtc_audio_device_impl.cc
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index 9fea501836b0b970ea83de8ac7072e3b01e9e134..55ee36eb7dd64f1514c389f1fec20d111a566f4e 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -129,7 +129,7 @@ static void AddHistogramFramesPerBuffer(HistogramDirection dir, int param) {
   }
 }
 
-WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
+WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl(int render_view_id)
     : ref_count_(0),
       render_loop_(base::MessageLoopProxy::current()),
       audio_transport_callback_(NULL),
@@ -148,7 +148,9 @@ WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
   // input side as well.
   DCHECK(RenderThreadImpl::current()) <<
       "WebRtcAudioDeviceImpl must be constructed on the render thread";
-  audio_output_device_ = AudioDeviceFactory::NewOutputDevice();
+  audio_input_device_ = AudioDeviceFactory::NewInputDevice(render_view_id);
+  DCHECK(audio_input_device_);
+  audio_output_device_ = AudioDeviceFactory::NewOutputDevice(render_view_id);
   DCHECK(audio_output_device_);
 }
 
@@ -408,7 +410,6 @@ int32_t WebRtcAudioDeviceImpl::Init() {
   if (initialized_)
     return 0;
 
-  DCHECK(!audio_input_device_);
   DCHECK(!input_buffer_.get());
   DCHECK(!output_buffer_.get());
 
@@ -572,7 +573,6 @@ int32_t WebRtcAudioDeviceImpl::Init() {
       16, in_buffer_size);
 
   // Create and configure the audio capturing client.
-  audio_input_device_ = AudioDeviceFactory::NewInputDevice();
   audio_input_device_->Initialize(input_audio_parameters_, this, this);
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputChannelLayout",
@@ -585,8 +585,6 @@ int32_t WebRtcAudioDeviceImpl::Init() {
   // Configure the audio rendering client.
   audio_output_device_->Initialize(output_audio_parameters_, this);
 
-  DCHECK(audio_input_device_);
-
   // 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
@@ -624,12 +622,10 @@ int32_t WebRtcAudioDeviceImpl::Terminate() {
   if (!initialized_)
     return 0;
 
-  DCHECK(audio_input_device_);
   DCHECK(input_buffer_.get());
   DCHECK(output_buffer_.get());
 
   // Release all resources allocated in Init().
-  audio_input_device_ = NULL;
   input_buffer_.reset();
   output_buffer_.reset();