| commit | 000d16396e748c7593ac41b24a876a6c20f0d277 | [log] [tgz] |
|---|---|---|
| author | mbonadei <mbonadei@webrtc.org> | Thu Dec 22 14:56:21 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Dec 22 14:56:21 2016 |
| tree | 421c3ed729dcb6a72518083cc9b557c9ef542ff1 | |
| parent | fb78c3e6fcb2c0499034f76d67894fe25b3f74d9 [diff] |
Refactor webrtc/modules/audio_conference_mixer for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_conference_mixer/*" BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2593003002 Cr-Commit-Position: refs/heads/master@{#15759}
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.