Add mute state field to AudioFrame.

webrtc/voice_engine/file_player.cc

 /*
  *  Copyright (c) 2012 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 108 matching lines @@
                     << " codec freq = " << _codec.plfreq
                     << ", wanted freq = " << frequencyInHz;
     return -1;
   }
 
   AudioFrame unresampledAudioFrame;
   if (STR_CASE_CMP(_codec.plname, "L16") == 0) {
     unresampledAudioFrame.sample_rate_hz_ = _codec.plfreq;
 
     // L16 is un-encoded data. Just pull 10 ms.
-    size_t lengthInBytes = sizeof(unresampledAudioFrame.data_);
+    size_t lengthInBytes = AudioFrame::kMaxDataSizeBytes;
     if (_fileModule.PlayoutAudioData(
-            reinterpret_cast<int8_t*>(unresampledAudioFrame.data_),
+            reinterpret_cast<int8_t*>(unresampledAudioFrame.mutable_data()),
             lengthInBytes) == -1) {
       // End of file reached.
       return -1;
     }
     if (lengthInBytes == 0) {
       *lengthInSamples = 0;
       return 0;
     }
     // One sample is two bytes.
     unresampledAudioFrame.samples_per_channel_ = lengthInBytes >> 1;
@@ skipping 24 matching lines @@
   size_t outLen = 0;
   if (_resampler.ResetIfNeeded(unresampledAudioFrame.sample_rate_hz_,
                                frequencyInHz, 1)) {
     LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec.";
 
     // New sampling frequency. Update state.
     outLen = static_cast<size_t>(frequencyInHz / 100);
     memset(outBuffer, 0, outLen * sizeof(int16_t));
     return 0;
   }
-  _resampler.Push(unresampledAudioFrame.data_,
+  _resampler.Push(unresampledAudioFrame.data(),
                   unresampledAudioFrame.samples_per_channel_, outBuffer,
                   MAX_AUDIO_BUFFER_IN_SAMPLES, outLen);
 
   *lengthInSamples = outLen;
 
   if (_scaling != 1.0) {
     for (size_t i = 0; i < outLen; i++) {
       outBuffer[i] = (int16_t)(outBuffer[i] * _scaling);
     }
   }
@@ skipping 195 matching lines @@
       // audio formats
       return std::unique_ptr<FilePlayer>(
           new FilePlayerImpl(instanceID, fileFormat));
     default:
       assert(false);
       return nullptr;
   }
 }
 
 }  // namespace webrtc