Index: webrtc/modules/audio_coding/test/Channel.h |
diff --git a/webrtc/modules/audio_coding/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h |
index 3dcd499c03cdc6bfe5cf5bb1cf27239bdc10f38c..5910fade25cba02990e2787073df4066a45fdce3 100644 |
--- a/webrtc/modules/audio_coding/test/Channel.h |
+++ b/webrtc/modules/audio_coding/test/Channel.h |
@@ -100,7 +100,7 @@ class Channel : public AudioPacketizationCallback { |
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample |
uint8_t _payloadData[60 * 32 * 2 * 2]; |
- mutable rtc::CriticalSection _channelCritSect; |
+ rtc::CriticalSection _channelCritSect; |
FILE* _bitStreamFile; |
bool _saveBitStream; |
int16_t _lastPayloadType; |