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

content/renderer/media/webrtc_audio_capturer.cc

Index: content/renderer/media/webrtc_audio_capturer.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index ff4ecc71cb1617fb102b6cf6e5c6ceda5844d622..8a48e30b25425485ccf3467f0f347a6364bae339 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -297,7 +297,8 @@ void WebRtcAudioCapturer::SetCapturerSource(
                                 channel_layout, 0, sample_rate,
                                 16, buffer_size, effects);
   scoped_refptr<MediaStreamAudioProcessor> new_audio_processor(
-      new MediaStreamAudioProcessor(params, constraints, effects));
+      new MediaStreamAudioProcessor(params, constraints, effects,
+                                    audio_device_));
   {
     base::AutoLock auto_lock(lock_);
     audio_processor_ = new_audio_processor;
@@ -550,27 +551,6 @@ void WebRtcAudioCapturer::GetAudioProcessingParams(
   *key_pressed = key_pressed_;
 }
 
-void WebRtcAudioCapturer::FeedRenderDataToAudioProcessor(
-    const int16* render_audio,
-    int sample_rate,
-    int number_of_channels,
-    int number_of_frames,
-    base::TimeDelta render_delay) {
-  scoped_refptr<MediaStreamAudioProcessor> audio_processor;
-  {
-    base::AutoLock auto_lock(lock_);
-    if (!running_)
-      return;
-
-    audio_processor = audio_processor_;
-  }
-
-  audio_processor->PushRenderData(render_audio, sample_rate,
-                                  number_of_channels,
-                                  number_of_frames,
-                                  render_delay);
-}
-
 void WebRtcAudioCapturer::SetCapturerSourceForTesting(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {