1 files changed, 4 insertions, 1 deletions
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 25c5e4dd88a..4bc0266b7d2 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -12,6 +12,8 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 
 #include "webrtc/common_types.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
@@ -72,7 +74,7 @@ class RTPSenderAudio : public DTMFqueue {
   Clock* const _clock;
   RTPSender* const _rtpSender;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+  rtc::CriticalSection _sendAudioCritsect;
 
   uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
 
@@ -100,6 +102,7 @@ class RTPSenderAudio : public DTMFqueue {
   // Audio level indication
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
   uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
+  OneTimeEvent first_packet_sent_;
 };
 }  // namespace webrtc