Add mute state field to AudioFrame.

webrtc/modules/audio_coding/neteq/tools/audio_sink.h

 /*
  *  Copyright (c) 2014 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.
  */
 
@@ skipping 15 matching lines @@
   virtual ~AudioSink() {}
 
   // Writes |num_samples| from |audio| to the AudioSink. Returns true if
   // successful, otherwise false.
   virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
 
   // Writes |audio_frame| to the AudioSink. Returns true if successful,
   // otherwise false.
   bool WriteAudioFrame(const AudioFrame& audio_frame) {
     return WriteArray(
-        audio_frame.data_,
+        audio_frame.data(),
         audio_frame.samples_per_channel_ * audio_frame.num_channels_);
   }
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
 };
 
 // Forks the output audio to two AudioSink objects.
 class AudioSinkFork : public AudioSink {
  public:
   AudioSinkFork(AudioSink* left, AudioSink* right)
       : left_sink_(left), right_sink_(right) {}
 
   bool WriteArray(const int16_t* audio, size_t num_samples) override;
 
  private:
   AudioSink* left_sink_;
   AudioSink* right_sink_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_