| commit | 32cd2c4103e7a84a5167a59973054dfca92bae74 | [log] [tgz] |
|---|---|---|
| author | danilchap <danilchap@webrtc.org> | Mon Aug 01 13:58:34 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Aug 01 13:58:41 2016 |
| tree | 23cae9dbfc4eb5be1010ef4d2d6598fbfea84eac | |
| parent | 95e756035eb03971b8c66caa74b444094306562a [diff] |
Fix issues with RestartingSendStreamPreservesRtpStatesWithRtx double check rtp_sender in sending mode when altering sequence_number adjust test to skip validating timestamp on rtx streams fix test by waiting for all 3 media streams instead of 3 out 6 media and rtx streams. BUG=webrtc:4332 Review-Url: https://codereview.webrtc.org/2177523002 Cr-Commit-Position: refs/heads/master@{#13587}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.