| commit | 83119dd83387a2a6abdbca2c3ee98b4353e11bc9 | [log] [tgz] |
|---|---|---|
| author | Steve Anton <steveanton@webrtc.org> | Fri Nov 10 16:19:52 2017 -0800 |
| committer | Commit Bot <commit-bot@chromium.org> | Sat Nov 11 00:50:35 2017 +0000 |
| tree | 976b687ad0affae501843a221b1fea4c50dba3fb | |
| parent | cda2562b066d7be65229da17156a960bf4e7ed8a [diff] |
Fix and re-enable flaky PeerConnectionIntegrationTests PeerConnectionIntegrationTest.AddMediaToConnectedBundleDoesNotRestartIce --> Fixed by an earlier CL (https://webrtc-review.googlesource.com/c/src/+/16261) so re-enabled. PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates --> An existing bug causes this to by flaky when using a fake clock. Fake clock removed with a TODO to change it back when the bug is fixed. PeerConnectionIntegrationTest.TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges --> The heuristic that >25% concealed audio samples is abnormal is unfortunately not reliable enough on certain slow trybots. Bump the threshold to 50% in hopes that is enough. Bug: webrtc:8496 Change-Id: I17cfdf956a8a72ac399212c3c7efcdd2236be00d Reviewed-on: https://webrtc-review.googlesource.com/20963 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20638}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.