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

content/renderer/media/webrtc_audio_capturer.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_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
@@ skipping 64 matching lines @@
   // up the native buffer size for the stream in order to optimize the
   // performance for peer connection.
   void EnablePeerConnectionMode();
 
   // Volume APIs used by WebRtcAudioDeviceImpl.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(int volume);
   int Volume() const;
   int MaxVolume() const;
 
-
   // Audio parameters utilized by the source of the audio capturer.
   // TODO(phoglund): Think over the implications of this accessor and if we can
   // remove it.
   media::AudioParameters source_audio_parameters() const;
 
   // Gets information about the paired output device. Returns true if such a
   // device exists.
   bool GetPairedOutputParameters(int* session_id,
                                  int* output_sample_rate,
                                  int* output_frames_per_buffer) const;
 
   const std::string& device_id() const { return device_info_.device.id; }
   int session_id() const { return device_info_.session_id; }
 
   // Stops recording audio. This method will empty its track lists since
   // stopping the capturer will implicitly invalidate all its tracks.
   // This method is exposed to the public because the media stream track can
   // call Stop() on its source.
   void Stop();
 
   // Called by the WebAudioCapturerSource to get the audio processing params.
   // This function is triggered by provideInput() on the WebAudio audio thread,
   // TODO(xians): Remove after moving APM from WebRtc to Chrome.
   void GetAudioProcessingParams(base::TimeDelta* delay, int* volume,
                                 bool* key_pressed);
 
-  // Called by the WebRtcAudioDeviceImpl to push the render audio to
-  // audio processor for echo cancellation analysis.
-  void FeedRenderDataToAudioProcessor(const int16* render_audio,
-                                      int sample_rate,
-                                      int number_of_channels,
-                                      int number_of_frames,
-                                      base::TimeDelta render_delay);
-
   // Use by the unittests to inject their own source to the capturer.
   void SetCapturerSourceForTesting(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::AudioParameters params);
 
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   virtual ~WebRtcAudioCapturer();
 
  private:
@@ skipping 83 matching lines @@
   // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
   // of RenderThread.
   WebRtcAudioDeviceImpl* audio_device_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_