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Revert "Wire up non-sender RTT for audio, and implement related standardized stats."

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This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.

Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.

Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}

# Not skipping CQ checks because original CL landed > 1 day ago.

TBR=hta,hbos,minyue

Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
This commit is contained in:
Björn Terelius
2021-09-01 16:03:26 +00:00
committed by WebRTC LUCI CQ
parent 78a8ce0a4c
commit 2c41cbae37
30 changed files with 27 additions and 552 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -200,9 +200,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
// which is the SSRC of the corresponding outbound RTP stream, is unique.
std::vector<ReportBlockData> GetLatestReportBlockData() const override;
absl::optional<SenderReportStats> GetSenderReportStats() const override;
// Round trip time statistics computed from the XR block contained in the last
// report.
absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
// (REMB) Receiver Estimated Max Bitrate.
void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;