Add mute state field to AudioFrame.

webrtc/api/call/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_API_CALL_AUDIO_SINK_H_
 #define WEBRTC_API_CALL_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>
 
 namespace webrtc {
 
 // Represents a simple push audio sink.
 class AudioSinkInterface {
  public:
   virtual ~AudioSinkInterface() {}
 
   struct Data {
-    Data(int16_t* data,
+    Data(const int16_t* data,

[hlundin-webrtc, 2017/03/16 14:47:48]

This file is in our set of public API files, which usually requires a PSA, a
waiting period, and then doing the change.

[yujo, 2017/03/16 23:37:21]

Done.

          size_t samples_per_channel,
          int sample_rate,
          size_t 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.
+    const 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.
     size_t 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_API_CALL_AUDIO_SINK_H_