commit | 2c41cbae37cac548a1133589b9d2c2e8614fa6cb | [log] [tgz] |
---|---|---|
author | Björn Terelius <terelius@webrtc.org> | Wed Sep 01 16:03:26 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Sep 01 17:32:00 2021 |
tree | 29f4a24302c931b042af2ba28022d1e4dd05b041 | |
parent | 78a8ce0a4c86c55a191ad6fac3cd934a7fc8d845 [diff] |
Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 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}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.