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, |
rtc::CriticalSection packet_loss_tracker_cs_; |
TransportFeedbackPacketLossTracker packet_loss_tracker_ |
- GUARDED_BY(&packet_loss_tracker_cs_); |
+ RTC_GUARDED_BY(&packet_loss_tracker_cs_); |
RtpRtcp* rtp_rtcp_module_; |
rtc::Optional<RtpState> const suspended_rtp_state_; |