Move AudioDevice and AudioInputDevice to media.

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_
 
 #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_input_device.h"
+#include "media/audio/audio_input_device.h"
 #include "media/base/audio_renderer_sink.h"
 #include "third_party/webrtc/modules/audio_device/main/interface/audio_device.h"
 
 // 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
@@ skipping 27 matching lines @@
 //      VoiceEngine::Delete(voe);
 //   }
 //
 // webrtc::VoiceEngine::Init() calls these ADM methods (in this order):
 //
 //  RegisterAudioCallback(this)
 //    webrtc::VoiceEngine is an webrtc::AudioTransport implementation and
 //    implements the RecordedDataIsAvailable() and NeedMorePlayData() callbacks.
 //
 //  Init()
-//    Creates and initializes the AudioDevice and AudioInputDevice objects.
+//    Creates and initializes the AudioOutputDevice and AudioInputDevice
+//    objects.
 //
 //  SetAGC(true)
 //    Enables the adaptive analog mode of the AGC which ensures that a
 //    suitable microphone volume level will be set. This scheme will affect
 //    the actual microphone control slider.
 //
 // Media example:
 //
 // When the underlying audio layer wants data samples to be played out, the
-// AudioDevice::RenderCallback() will be called, which in turn uses the
+// AudioOutputDevice::RenderCallback() will be called, which in turn uses the
 // registered webrtc::AudioTransport callback and gets the data to be played
 // out from the webrtc::VoiceEngine.
 //
 // The picture below illustrates the media flow on the capture side where the
 // AudioInputDevice client acts as link between the renderer and browser
 // process:
 //
 //                   .------------------.            .----------------------.
 // (Native audio) => | AudioInputStream |-> OnData ->| AudioInputController |-.
 //                   .------------------.            .----------------------. |
@@ skipping 109 matching lines @@
 //  - The webrtc::AudioDeviceModule is reference counted.
 //  - AGC is only supported in combination with the WASAPI-based audio layer
 //    on Windows, i.e., it is not supported on Windows XP.
 //  - All volume levels required for the AGC scheme are transfered in a
 //    normalized range [0.0, 1.0]. Scaling takes place in both endpoints
 //    (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 {
+      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
+      NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureCallback),
+      NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureEventHandler) {
  public:
   // Methods called on main render thread.
   WebRtcAudioDeviceImpl();
 
   // 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;
 
@@ skipping 177 matching lines @@
   // be executed on that thread.
   void InitOnRenderThread(int32_t* error, base::WaitableEvent* event);
 
   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_;
+  scoped_refptr<media::AudioInputDevice> audio_input_device_;
 
   // Provides access to the native audio output layer in the browser process.
   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.
@@ skipping 32 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_