commit | 073ece45b68fc1aecb88692438bcb762e9fd18d5 | [log] [tgz] |
---|---|---|
author | johan <johan@webrtc.org> | Fri Aug 26 09:59:47 2016 |
committer | Commit bot <commit-bot@chromium.org> | Fri Aug 26 09:59:56 2016 |
tree | 5aec4c17e0bbdc643f000b8da4e44bebae3c1637 | |
parent | c76680419e5a6fe5572400f28d55955c6ab24aaf [diff] |
Skip unit test if GYP_DEFINES="rtc_use_h264=1" is not set. Unit test would fail in default configuration (e.g. rtc_use_h264=0), cause it tests instantiating H264 specifics. BUG=webrtc:6194, webrtc:6198 Review-Url: https://codereview.webrtc.org/2228733004 Cr-Commit-Position: refs/heads/master@{#13929}
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.