| commit | ab86e7ffe2106c086cde631eee1418f2c128b05b | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Mon Feb 12 15:16:02 2018 -0800 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Mon Feb 12 23:17:51 2018 +0000 |
| tree | 511e435ec380ed33f215caf13953125a287bdf7a | |
| parent | ef76e5a06207f4e775fc552c19a33af5e1e96a60 [diff] |
Disable a couple flaky RampUpTest tests on Mac. Specifically, UpDownUpAudioVideoTransportSequenceNumberRtx and AudioTransportSequenceNumber. TBR=stefan@webrtc.org Bug: webrtc:8878 Change-Id: I24ad65dc56068bb01c53d97511faadc22bd13c31 Reviewed-on: https://webrtc-review.googlesource.com/52161 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21992}
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.