| commit | 9410b51037ab6b2497b03f3888a3e1defd9dcbeb | [log] [tgz] |
|---|---|---|
| author | ehmaldonado <ehmaldonado@webrtc.org> | Thu Jan 19 19:57:56 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Jan 19 19:57:56 2017 |
| tree | 3e396f5c668d1f77d00c00abf2001b8208ea5eab | |
| parent | d74886350e432547e7a70095170fe1a9a0dab0c8 [diff] |
GN: Add audio_conference_mixer_unittests to modules_unittests. Was removed by accident in https://codereview.webrtc.org/2629923002/ BUG=webrtc:6954 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2641203002 Cr-Commit-Position: refs/heads/master@{#16170}
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.