| 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;
|
|
|