Remove playout delay lock.
Now update the playout delay and related stats on the worker thread. This was previously reviewed here: https://webrtc-review.googlesource.com/c/src/+/172929/ With the exception of reducing unnecessarily broad lock scope in one function in rtp_rtcp_impl.cc and added comments in rtp_streams_synchronizer.h Bug: webrtc:11489 Change-Id: I77807b5da2accfe774255d9409542d358f288993 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174200 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31193}
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@ -774,8 +774,10 @@ std::vector<rtcp::TmmbItem> ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner) {
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}
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void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
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rtc::CritScope cs(&critical_section_rtt_);
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rtt_ms_ = rtt_ms;
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{
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rtc::CritScope cs(&critical_section_rtt_);
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rtt_ms_ = rtt_ms;
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}
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if (rtp_sender_) {
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rtp_sender_->packet_history.SetRtt(rtt_ms);
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}
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