Storing raw audio sink for default audio track.

webrtc/audio/audio_sink.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_SINK_H_
 #define WEBRTC_AUDIO_AUDIO_SINK_H_
 
 #if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
 // Avoid conflict with format_macros.h.
 #define __STDC_FORMAT_MACROS
 #endif
 
 #include <inttypes.h>
 #include <stddef.h>
 
+#include "webrtc/base/refcount.h"
+
 namespace webrtc {
 
 // Represents a simple push audio sink.
-class AudioSinkInterface {
+class AudioSinkInterface : public rtc::RefCountInterface {
  public:
   virtual ~AudioSinkInterface() {}
 
   struct Data {
     Data(int16_t* data,
          size_t samples_per_channel,
          int sample_rate,
          int channels,
          uint32_t timestamp)
         : data(data),
           samples_per_channel(samples_per_channel),
           sample_rate(sample_rate),
           channels(channels),
           timestamp(timestamp) {}
 
     int16_t* data;               // The actual 16bit audio data.
     size_t samples_per_channel;  // Number of frames in the buffer.
     int sample_rate;             // Sample rate in Hz.
     int channels;                // Number of channels in the audio data.
     uint32_t timestamp;          // The RTP timestamp of the first sample.
   };
 
   virtual void OnData(const Data& audio) = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SINK_H_