[audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was split up into smaller pieces for reviewability. Spec: https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats In [1], ReportBlockData was implemented and tested. This CL adds the plumbing to make it available in MediaSenderInfo for audio streams, but the code is not wired up to make use of these stats. In follow-up CL [2], ReportBlockData will be used to implement RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration tests exercising the full code path. [1] https://webrtc-review.googlesource.com/c/src/+/136584 [2] https://webrtc-review.googlesource.com/c/src/+/138067 Bug: webrtc:10455 Change-Id: Id8940090cc9c121e8f06ccdb068a22ce24c07092 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138066 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28072}
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@ -589,6 +589,11 @@ int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
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return rtcp_receiver_.StatisticsReceived(receive_blocks);
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}
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std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
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const {
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return rtcp_receiver_.GetLatestReportBlockData();
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}
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// (REMB) Receiver Estimated Max Bitrate.
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void ModuleRtpRtcpImpl::SetRemb(int64_t bitrate_bps,
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std::vector<uint32_t> ssrcs) {
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