Adds AudioDevice factory for all audio clients in Chrome

content/renderer/media/webrtc_audio_device_impl.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #pragma once
 
 #include <string>
 #include <vector>
 
 #include "base/basictypes.h"
 #include "base/compiler_specific.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop_proxy.h"
 #include "base/time.h"
 #include "content/common/content_export.h"
-#include "content/renderer/media/audio_device.h"
 #include "content/renderer/media/audio_input_device.h"
+#include "media/base/audio_renderer_sink.h"
 #include "third_party/webrtc/modules/audio_device/main/interface/audio_device.h"
 
+class AudioDeviceFactoryInterface;
+
 // A WebRtcAudioDeviceImpl instance implements the abstract interface
 // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc::
 // VoiceEngine) to register this class as an external AudioDeviceModule (ADM).
 // Then WebRtcAudioDeviceImpl::SetSessionId() needs to be called to set the
 // session id that tells which device to use. The user can either get the
 // session id from the MediaStream or use a value of 1 (AudioInputDeviceManager
 // ::kFakeOpenSessionId), the later will open the default device without going
 // through the MediaStream. The user can then call WebRtcAudioDeviceImpl::
 // StartPlayout() and WebRtcAudioDeviceImpl::StartRecording() from the render
 // process to initiate and start audio rendering and capturing in the browser
@@ skipping 169 matching lines @@
 //    (WebRTC client a media layer). This approach ensures that we can avoid
 //    transferring maximum levels between the renderer and the browser.
 //
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule),
       public media::AudioRendererSink::RenderCallback,
       public AudioInputDevice::CaptureCallback,
       public AudioInputDevice::CaptureEventHandler {
  public:
   // Methods called on main render thread.
-  WebRtcAudioDeviceImpl();
+  WebRtcAudioDeviceImpl(AudioDeviceFactoryInterface* audio_device_factory);
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
   // to release the reference and delete this object.
   virtual int32_t AddRef() OVERRIDE;
   virtual int32_t Release() OVERRIDE;
 
   // We need this one to support runnable method tasks.
   static bool ImplementsThreadSafeReferenceCounting() { return true; }
 
-  // AudioDevice::RenderCallback implementation.
+  // media::AudioRendererSink::RenderCallback implementation.
   virtual int Render(const std::vector<float*>& audio_data,
                      int number_of_frames,
                      int audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
   // AudioInputDevice::CaptureCallback implementation.
   virtual void Capture(const std::vector<float*>& audio_data,
                        int number_of_frames,
                        int audio_delay_milliseconds,
                        double volume) OVERRIDE;
@@ skipping 166 matching lines @@
   int ref_count_;
 
   // Gives access to the message loop of the render thread on which this
   // object is created.
   scoped_refptr<base::MessageLoopProxy> render_loop_;
 
   // Provides access to the native audio input layer in the browser process.
   scoped_refptr<AudioInputDevice> audio_input_device_;
 
   // Provides access to the native audio output layer in the browser process.
-  scoped_refptr<AudioDevice> audio_output_device_;
+  scoped_refptr<media::AudioRendererSink> audio_output_device_;
 
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().
   webrtc::AudioTransport* audio_transport_callback_;
 
   // Cached values of utilized audio parameters. Platform dependent.
   media::AudioParameters input_audio_parameters_;
   media::AudioParameters output_audio_parameters_;
 
@@ skipping 29 matching lines @@
   bool agc_is_enabled_;
 
   // Used for histograms of total recording and playout times.
   base::Time start_capture_time_;
   base::Time start_render_time_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_