Feed the render data to MediaStreamAudioProcessor and used AudioBus in render callback

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 168 matching lines @@
 namespace content {
 
 class WebRtcAudioCapturer;
 class WebRtcAudioRenderer;
 
 // TODO(xians): Move the following two interfaces to webrtc so that
 // libjingle can own references to the renderer and capturer.
 class WebRtcAudioRendererSource {
  public:
   // Callback to get the rendered interleaved data.
-  // TODO(xians): Change uint8* to int16*.
-  virtual void RenderData(uint8* audio_data,
-                          int number_of_channels,
-                          int number_of_frames,
+  virtual void RenderData(media::AudioBus* audio_bus,
+                          int sample_rate,
                           int audio_delay_milliseconds) = 0;
 
-  // Set the format for the capture audio parameters.
-  virtual void SetRenderFormat(const media::AudioParameters& params) = 0;
-
   // Callback to notify the client that the renderer is going away.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
 
  protected:
   virtual ~WebRtcAudioRendererSource() {}
 };
 
 class PeerConnectionAudioSink {
  public:
   // Callback to deliver the captured interleaved data.
@@ skipping 20 matching lines @@
 
   // Set the format for the capture audio parameters.
   // This is called when the capture format has changed, and it must be called
   // on the same thread as calling CaptureData().
   virtual void OnSetFormat(const media::AudioParameters& params) = 0;
 
  protected:
  virtual ~PeerConnectionAudioSink() {}
 };
 
-// Note that this class inherits from webrtc::AudioDeviceModule but due to
+//Note that this class inherits from webrtc::AudioDeviceModule but due to

[perkj_chrome, 2014/01/20 12:29:35]

nit indentation

[no longer working on chromium, 2014/01/23 13:02:30]

Done.

 // the high number of non-implemented methods, we move the cruft over to the
 // WebRtcAudioDeviceNotImpl.
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public PeerConnectionAudioSink),
       NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl),
       NON_EXPORTED_BASE(public WebRtcAudioRendererSource) {
  public:
   // The maximum volume value WebRtc uses.
   static const int kMaxVolumeLevel = 255;
 
@@ skipping 49 matching lines @@
   // Sets the |renderer_|, returns false if |renderer_| already exists.
   // Called on the main renderer thread.
   bool SetAudioRenderer(WebRtcAudioRenderer* renderer);
 
   // Adds/Removes the capturer to the ADM.
   // TODO(xians): Remove these two methods once the ADM does not need to pass
   // hardware information up to WebRtc.
   void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
   void RemoveAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
 
+  // Adds/Removes the observer of WebRtcAudioRendererSource to the ADM.
+  // These methods are used by the MediaStreamAudioProcesssor to get the
+  // render data for AEC.
+  void AddRenderDataObserver(WebRtcAudioRendererSource* observer);

[perkj_chrome, 2014/01/20 12:29:35]

I am a bit confused about the audio naming convention. What is a
RenderDataObserver and why is a WebRtcAudioRendererSource and observer of it? 

[no longer working on chromium, 2014/01/23 13:02:30]

Do you have any other suggestion on the naming here?

APM is the observer to the render data. And I am re-using
WebRtcAudioRendererSource::RenderData to pass the render data to the APMs here.
Another alternative for me is to add an interface like:
WebRtcRenderDataObserver {
public:
  virtual void RenderData(media::AudioBus* audio_bus, int sample_rate, int
delay_ms) = 0;
}
which is very similar to WebRtcAudioRendererSource

+  void RemoveRenderDataObserver(WebRtcAudioRendererSource* observer);
+
   // Gets paired device information of the capture device for the audio
   // renderer. This is used to pass on a session id, sample rate and buffer
   // size to a webrtc audio renderer (either local or remote), so that audio
   // will be rendered to a matching output device.
   // Returns true if the capture device has a paired output device, otherwise
   // false. Note that if there are more than one open capture device the
   // function will not be able to pick an appropriate device and return false.
   bool GetAuthorizedDeviceInfoForAudioRenderer(
       int* session_id, int* output_sample_rate, int* output_buffer_size);
 
   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<WebRtcAudioRendererSource* > RenderDataObservers;
+  class RenderBuffer;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
 
   // PeerConnectionAudioSink implementation.
 
   // Called on the AudioInputDevice worker thread.
   virtual int OnData(const int16* audio_data,
                      int sample_rate,
                      int number_of_channels,
                      int number_of_frames,
                      const std::vector<int>& channels,
                      int audio_delay_milliseconds,
                      int current_volume,
                      bool need_audio_processing,
                      bool key_pressed) OVERRIDE;
 
   // Called on the AudioInputDevice worker thread.
   virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
 
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioInputDevice worker thread.
-  virtual void RenderData(uint8* audio_data,
-                          int number_of_channels,
-                          int number_of_frames,
+  virtual void RenderData(media::AudioBus* audio_bus,
+                          int sample_rate,
                           int audio_delay_milliseconds) OVERRIDE;
 
   // Called on the main render thread.
-  virtual void SetRenderFormat(const media::AudioParameters& params) OVERRIDE;
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE;
 
   // Helper to get the default capturer, which is the last capturer in
   // |capturers_|.
   scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const;
 
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   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_;
 
+  // List of observers which requires access to the render data.

[perkj_chrome, 2014/01/20 12:29:35]

render data? Is this the audio that will be played out to a speaker normally by
for ex an audio tag?

[no longer working on chromium, 2014/01/23 13:02:30]

Yes.

+  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_;
 
   // Cached value of the current audio delay on the output/renderer side.
   int output_delay_ms_;
 
   // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|, |renderer_|
   // |recording_| and |microphone_volume_|.
   mutable base::Lock lock_;
 
   // Used to protect the racing of calling OnData() since there can be more
   // than one input stream calling OnData().
   mutable base::Lock capture_callback_lock_;
 
+  //

[perkj_chrome, 2014/01/20 12:29:35]

?

   bool initialized_;
   bool playing_;
   bool recording_;
 
   // Used for histograms of total recording and playout times.
   base::Time start_capture_time_;
   base::Time start_render_time_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
+  // Buffer used for temporary storage during render callback.
+  // It is only accessed by the audio render thread.
+  scoped_ptr<int16[]> render_buffer_;
+  int render_buffer_size_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_