MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/webrtc_audio_sink_adapter.h

 // Copyright 2014 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_WEBRTC_AUDIO_SINK_ADAPTER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_ADAPTER_H_
 
 #include <stdint.h>
 
 #include "base/macros.h"
 #include "base/memory/scoped_ptr.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "media/audio/audio_parameters.h"
+#include "media/base/audio_push_fifo.h"
 
 namespace webrtc {
 class AudioTrackSinkInterface;
 }  // namespace webrtc
 
 namespace content {
 
 // Adapter to the webrtc::AudioTrackSinkInterface of the audio track.
 // This class is used in between the MediaStreamAudioSink and
 // webrtc::AudioTrackSinkInterface. It gets data callback via the
-// MediaStreamAudioSink::OnData() interface and pass the data to
+// MediaStreamAudioSink::OnData() interface, re-chunks the data into the
+// required 10ms chunks if needed, and passes the data to WebRTC via the
 // webrtc::AudioTrackSinkInterface.
 class WebRtcAudioSinkAdapter : public MediaStreamAudioSink {
  public:
   explicit WebRtcAudioSinkAdapter(
       webrtc::AudioTrackSinkInterface* sink);
   ~WebRtcAudioSinkAdapter() override;
 
   bool IsEqual(const webrtc::AudioTrackSinkInterface* other) const;
 
  private:
   // MediaStreamAudioSink implementation.
   void OnData(const media::AudioBus& audio_bus,
               base::TimeTicks estimated_capture_time) override;
   void OnSetFormat(const media::AudioParameters& params) override;
 
+  // Called by AudioPushFifo zero or more times during the call to OnData().
+  // Delivers audio data with the required 10ms buffer size to |sink_|.
+  void DeliverRebufferedAudio(const media::AudioBus& audio_bus,
+                              int frame_delay);
+
   webrtc::AudioTrackSinkInterface* const sink_;
-
+  media::AudioPushFifo fifo_;
   media::AudioParameters params_;
   scoped_ptr<int16_t[]> interleaved_data_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioSinkAdapter);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_ADAPTER_H_