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

content/renderer/media/webrtc_audio_device_impl.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_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 111 matching lines @@
     // Report unexpected sample rates using a unique histogram name.
     if (dir == kAudioOutput) {
       UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputFramesPerBufferUnexpected",
                            param);
     } else {
       UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputFramesPerBufferUnexpected", param);
     }
   }
 }
 
-WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
+WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl(int render_view_id)
     : ref_count_(0),
       render_loop_(base::MessageLoopProxy::current()),
       audio_transport_callback_(NULL),
       input_delay_ms_(0),
       output_delay_ms_(0),
       last_error_(AudioDeviceModule::kAdmErrNone),
       last_process_time_(base::TimeTicks::Now()),
       session_id_(0),
       bytes_per_sample_(0),
       initialized_(false),
       playing_(false),
       recording_(false),
       agc_is_enabled_(false) {
   DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()";
   // TODO(henrika): remove this restriction when factory is used for the
   // 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_);
 }
 
 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() {
   DVLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()";
   if (playing_)
     StopPlayout();
   if (recording_)
     StopRecording();
   if (initialized_)
@@ skipping 239 matching lines @@
         base::Bind(&WebRtcAudioDeviceImpl::InitOnRenderThread,
                    this, &error, &event));
     event.Wait();
     return error;
   }
 
   // Calling Init() multiple times in a row is OK.
   if (initialized_)
     return 0;
 
-  DCHECK(!audio_input_device_);
   DCHECK(!input_buffer_.get());
   DCHECK(!output_buffer_.get());
 
   // TODO(henrika): it could be possible to allow one of the directions (input
   // or output) to use a non-supported rate. As an example: if only the
   // output rate is OK, we could finalize Init() and only set up an
   // AudioOutputDevice.
 
   // Ask the browser for the default audio output hardware sample-rate.
   // This request is based on a synchronous IPC message.
@@ skipping 143 matching lines @@
   // after a successful initialization.
   output_audio_parameters_.Reset(
       AudioParameters::AUDIO_PCM_LOW_LATENCY, out_channel_layout,
       out_sample_rate, 16, out_buffer_size);
 
   input_audio_parameters_.Reset(
       in_format, in_channel_layout, in_sample_rate,
       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",
                             out_channel_layout, media::CHANNEL_LAYOUT_MAX);
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             in_channel_layout, media::CHANNEL_LAYOUT_MAX);
   AddHistogramFramesPerBuffer(kAudioOutput, out_buffer_size);
   AddHistogramFramesPerBuffer(kAudioInput, in_buffer_size);
 
   // 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
   // number of channels.
   input_buffer_.reset(new int16[input_buffer_size() * input_channels()]);
   output_buffer_.reset(new int16[output_buffer_size() * output_channels()]);
 
   DCHECK(input_buffer_.get());
   DCHECK(output_buffer_.get());
 
@@ skipping 17 matching lines @@
   event->Signal();
 }
 
 int32_t WebRtcAudioDeviceImpl::Terminate() {
   DVLOG(1) << "Terminate()";
 
   // Calling Terminate() multiple times in a row is OK.
   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();
 
   initialized_ = false;
   return 0;
 }
 
 bool WebRtcAudioDeviceImpl::Initialized() const {
   return initialized_;
 }
@@ skipping 514 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
   session_id_ = session_id;
 }