| commit | a3cac0514abf7e0eddcdab40abd858298cbc4f6c | [log] [tgz] |
|---|---|---|
| author | kjellander <kjellander@webrtc.org> | Mon Oct 24 08:52:39 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Oct 24 09:09:40 2016 |
| tree | 00f22279ee368e5cba8cba852950bff0781a1fad | |
| parent | 43536c3d6a728407adbaabc5b2f2d7a42234234a [diff] |
GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn Move the following targets into webrtc/video/BUILD.gn: * screenshare_loopback * video_quality_test * video_loopback Add new target 'run_tests' in webrtc/test/BUILD.gn, being used by two of the above and make then depend on that instead. BUG=webrtc:6440 NOTRY=True Review-Url: https://codereview.webrtc.org/2438973002 Cr-Commit-Position: refs/heads/master@{#14735}
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.