move the APM to chrome.

content/renderer/media/webrtc_audio_device_impl.h

Index: content/renderer/media/webrtc_audio_device_impl.h
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index b644178030b5009677935ad1a45bbf80033479b4..039175550dfb8c66ce29c6534084a9a6f3a37617 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -210,12 +210,6 @@ class WebRtcAudioCapturerSink {
   // |sample_rate| is the sample frequency of audio data.
   // |number_of_channels| is the number of channels reflecting the order of
   // surround sound channels.
-  // |audio_delay_milliseconds| is recording delay value.
-  // |current_volume| is current microphone volume, in range of |0, 255].
-  // |need_audio_processing| indicates if the audio needs WebRtc AEC/NS/AGC
-  // audio processing.
-  // The return value is the new microphone volume, in the range of |0, 255].
-  // When the volume does not need to be updated, it returns 0.
   virtual int CaptureData(const std::vector<int>& channels,

[Henrik Grunell, 2013/10/31 11:56:12]

Why remove the description of some input parameters and the return value?

                           const int16* audio_data,
                           int sample_rate,
@@ -233,6 +227,22 @@ class WebRtcAudioCapturerSink {
   virtual ~WebRtcAudioCapturerSink() {}
 };
 
+class WebRtcAudioRenderDataObserver {

[Henrik Grunell, 2013/10/31 11:56:12]

Add comment. See also next comment.

+ public:
+  // Callback to get the mixed rendered data.
+  virtual void OnRenderData(const int16* render_audio,
+                            int sample_rate,
+                            int number_of_channels,
+                            int number_of_frames,
+                            int render_delay_ms) = 0;
+
+  // Callback to notify the observer the webrtc render object is closing.
+  virtual void OnRenderClosing() = 0;
+
+ protected:
+  virtual ~WebRtcAudioRenderDataObserver() {}
+};
+
 // Note that this class inherits from webrtc::AudioDeviceModule but due to
 // the high number of non-implemented methods, we move the cruft over to the
 // WebRtcAudioDeviceNotImpl.
@@ -305,6 +315,10 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // Adds the capturer to the ADM.
   void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
 
+  // Add/Remove the observer to get the mixed render data for AEC.

[Henrik Grunell, 2013/10/31 11:56:12]

I think the description "to get the mixed render data for AEC" should be moved
to a comment for WebRtcAudioRenderDataObserver above. Also change the text so
that it's clear that using the data for AEC is one example. There may be other
uses in the future.

+  void AddRenderDataObserver(WebRtcAudioRenderDataObserver* observer);
+  void RemoveRenderDataObserver(WebRtcAudioRenderDataObserver* observer);
+
   // Gets the default capturer, which is the capturer in the list with
   // a valid |device_id|. Microphones are represented by capturers with a valid
   // |device_id|, since only one microphone is supported today, only one
@@ -326,6 +340,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
+  typedef std::list<WebRtcAudioRenderDataObserver* > RenderDataObservers;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
@@ -370,6 +385,8 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // Provides access to the audio renderer in the browser process.
   scoped_refptr<WebRtcAudioRenderer> renderer_;
 
+  RenderDataObservers render_data_observers_;
+
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().