83119dd833
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct