Migrate leftovers in media/ and modules/ to webrtc::Mutex.
Bug: webrtc:11567 Change-Id: Id40a53fcec6cba1cd5af70422291ba46b0a6da8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178905 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31694}
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@ -804,7 +804,7 @@ bool ModuleRtpRtcpImpl::LastReceivedNTP(
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void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
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{
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rtc::CritScope cs(&critical_section_rtt_);
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MutexLock lock(&mutex_rtt_);
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rtt_ms_ = rtt_ms;
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}
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if (rtp_sender_) {
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@ -813,7 +813,7 @@ void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
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}
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int64_t ModuleRtpRtcpImpl::rtt_ms() const {
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rtc::CritScope cs(&critical_section_rtt_);
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MutexLock lock(&mutex_rtt_);
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return rtt_ms_;
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}
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