First step towards moving AudioDevice from content/ to media/audio.

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 129 matching lines @@
       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::Create();
+  audio_output_device_ = AudioDeviceFactory::NewOutputDevice();
   DCHECK(audio_output_device_);
 }
 
 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() {
   DVLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()";
   if (playing_)
     StopPlayout();
   if (recording_)
     StopRecording();
   if (initialized_)
@@ skipping 230 matching lines @@
     LOG(ERROR) << "Unable to (de)register transport during active media";
     return -1;
   }
   audio_transport_callback_ = audio_callback;
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::Init() {
   DVLOG(1) << "Init()";
 
+  // TODO(henrika): After switching to using the AudioDeviceFactory for
+  // instantiating the input device, maybe this isn't a requirement anymore?
   if (!render_loop_->BelongsToCurrentThread()) {
     int32_t error = 0;
     base::WaitableEvent event(false, false);
     // Ensure that we call Init() from the main render thread since
     // the audio clients can only be created on this thread.
     render_loop_->PostTask(
         FROM_HERE,
         base::Bind(&WebRtcAudioDeviceImpl::InitOnRenderThread,
                    this, &error, &event));
     event.Wait();
@@ skipping 160 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_ = new AudioInputDevice(
-      input_audio_parameters_, this, this);
+  audio_input_device_ = AudioDeviceFactory::NewInputDevice();
+  audio_input_device_->Initialize(input_audio_parameters_, this, this);
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputChannelLayout",
                             out_channel_layout, CHANNEL_LAYOUT_MAX);
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             in_channel_layout, 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);
@@ skipping 572 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
   session_id_ = session_id;
 }