Add mute state field to AudioFrame.

webrtc/voice_engine/file_player.cc

Index: webrtc/voice_engine/file_player.cc
diff --git a/webrtc/voice_engine/file_player.cc b/webrtc/voice_engine/file_player.cc
index b581d5235b78dc892018cece8c127b0aaabb7a17..5faf4664137c362a76ebef5a82c23bcdc07c38a5 100644
--- a/webrtc/voice_engine/file_player.cc
+++ b/webrtc/voice_engine/file_player.cc
@@ -127,9 +127,9 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(int16_t* outBuffer,
     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;
@@ -174,7 +174,7 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(int16_t* outBuffer,
     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);