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

content/renderer/media/webrtc_local_audio_renderer.cc

 // 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.
 
 #include "content/renderer/media/webrtc_local_audio_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/metrics/histogram.h"
@@ skipping 38 matching lines @@
     DVLOG(2) << "loopback FIFO is empty";
   }
 
   return audio_bus->frames();
 }
 
 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 <=
       loopback_fifo_->max_frames()) {
     scoped_ptr<media::AudioBus> audio_source = media::AudioBus::Create(
         number_of_channels, number_of_frames);
     audio_source->FromInterleaved(audio_data,
                                   audio_source->frames(),
                                   sizeof(audio_data[0]));
     loopback_fifo_->Push(audio_source.get());
 
     const base::TimeTicks now = base::TimeTicks::Now();
     total_render_time_ += now - last_render_time_;
     last_render_time_ = now;
   } 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();
   DCHECK(capture_thread_checker_.CalledOnValidThread());
 
   // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
   // the new format.
   {
     base::AutoLock auto_lock(thread_lock_);
     if (source_params_ == params)
@@ skipping 212 matching lines @@
   // parameters.  Then, invoke MaybeStartSink() to restart everything again.
   if (sink_started_) {
     sink_->Stop();
     sink_started_ = false;
   }
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_);
   MaybeStartSink();
 }
 
 }  // namespace content