| commit | 85904f4ee8b4526ff32dcff31183af1317af86eb | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Feb 16 10:11:49 2018 -0800 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Feb 16 18:12:15 2018 +0000 |
| tree | a765defbb73c9da3a65232f3bc860fe18696c3e3 | |
| parent | ba6b503ae500b0c5ecc1bfa2ab6c5aeba3123dc4 [diff] |
Disabling a couple flaky CallPerfTests on Mac. Specifically, KeepsHighBitrateWhenReconfiguringSender and MinVideoAndAudioBitrate. TBR=stefan@webrtc.org Bug: webrtc:8878 Change-Id: I7c3f110dec199ffc4226b24fea404ee205932940 Reviewed-on: https://webrtc-review.googlesource.com/54620 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22060}
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.
Authoritative list of directories that contain the native API header files.