| Index: webrtc/audio/audio_send_stream.h
|
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
|
| index f46608d59afc5e89812f7cf59b58656486be7797..cf36c1393f3ef3d76ff67062cdba21c31ef331b2 100644
|
| --- a/webrtc/audio/audio_send_stream.h
|
| +++ b/webrtc/audio/audio_send_stream.h
|
| @@ -115,7 +115,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
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|
|
| rtc::CriticalSection packet_loss_tracker_cs_;
|
| TransportFeedbackPacketLossTracker packet_loss_tracker_
|
| - GUARDED_BY(&packet_loss_tracker_cs_);
|
| + RTC_GUARDED_BY(&packet_loss_tracker_cs_);
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|
|
| RtpRtcp* rtp_rtcp_module_;
|
| rtc::Optional<RtpState> const suspended_rtp_state_;
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|
|