move the APM to chrome.

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 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>
 
@@ skipping 192 matching lines @@
 };
 
 class WebRtcAudioCapturerSink {
  public:
   // Callback to deliver the captured interleaved data.
   // |channels| contains a vector of WebRtc VoE channels.
   // |audio_data| is the pointer to the audio data.
   // |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,
                           int number_of_channels,
                           int number_of_frames,
                           int audio_delay_milliseconds,
                           int current_volume,
                           bool need_audio_processing,
                           bool key_pressed) = 0;
 
   // Set the format for the capture audio parameters.
   virtual void SetCaptureFormat(const media::AudioParameters& params) = 0;
 
  protected:
   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.
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl),
       NON_EXPORTED_BASE(public WebRtcAudioCapturerSink),
       NON_EXPORTED_BASE(public WebRtcAudioRendererSource) {
  public:
   // The maximum volume value WebRtc uses.
   static const int kMaxVolumeLevel = 255;
@@ skipping 52 matching lines @@
   virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
   virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
 
   // Sets the |renderer_|, returns false if |renderer_| already exists.
   // Called on the main renderer thread.
   bool SetAudioRenderer(WebRtcAudioRenderer* renderer);
 
   // 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
   // capturer in the |capturers_| can have a valid |device_id|.
   scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const;
 
   const scoped_refptr<WebRtcAudioRenderer>& renderer() const {
     return renderer_;
   }
   int output_buffer_size() const {
     return output_audio_parameters_.frames_per_buffer();
   }
   int output_channels() const {
     return output_audio_parameters_.channels();
   }
   int output_sample_rate() const {
     return output_audio_parameters_.sample_rate();
   }
 
  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();
 
   // WebRtcAudioCapturerSink implementation.
 
   // Called on the AudioInputDevice worker thread.
   virtual int CaptureData(const std::vector<int>& channels,
                           const int16* audio_data,
                           int sample_rate,
@@ skipping 24 matching lines @@
 
   int ref_count_;
 
   // List of captures which provides access to the native audio input layer
   // in the browser process.
   CapturerList capturers_;
 
   // 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().
   webrtc::AudioTransport* audio_transport_callback_;
 
   // Cached values of used output audio parameters. Platform dependent.
   media::AudioParameters output_audio_parameters_;
 
   // Cached value of the current audio delay on the input/capture side.
   int input_delay_ms_;
@@ skipping 19 matching lines @@
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_