Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/webrtc_local_audio_renderer.cc

Index: content/renderer/media/webrtc_local_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index 017f63239e275bcb0c4fcac96d48f3a4fe4fb4ed..89effe526736413c4e8697bed5bff3961355bfa5 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -56,21 +56,16 @@ void WebRtcLocalAudioRenderer::OnRenderError() {
   NOTIMPLEMENTED();
 }
 
-// content::WebRtcAudioCapturerSink implementation
-int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
-                                          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) {
+// content::MediaStreamAudioSink implementation
+void WebRtcLocalAudioRenderer::OnData(const int16* audio_data,
+                                      int sample_rate,
+                                      int number_of_channels,
+                                      int number_of_frames) {
   DCHECK(capture_thread_checker_.CalledOnValidThread());
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
   base::AutoLock auto_lock(thread_lock_);
   if (!playing_ || !volume_ || !loopback_fifo_)
-    return 0;
+    return;
 
   // Push captured audio to FIFO so it can be read by a local sink.
   if (loopback_fifo_->frames() + number_of_frames <=
@@ -88,13 +83,11 @@ int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
   } else {
     DVLOG(1) << "FIFO is full";
   }
-
-  return 0;
 }
 
-void WebRtcLocalAudioRenderer::SetCaptureFormat(
+void WebRtcLocalAudioRenderer::OnSetFormat(
     const media::AudioParameters& params) {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::SetCaptureFormat()";
+  DVLOG(1) << "WebRtcLocalAudioRenderer::OnSetFormat()";
   // If the source is restarted, we might have changed to another capture
   // thread.
   capture_thread_checker_.DetachFromThread();