| commit | a68eb8c1cb201b988adad3baab6f253e92221022 | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Mon Feb 17 15:13:14 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Feb 18 11:03:58 2020 |
| tree | e60954040cc6720511f5c9b039c518694d777244 | |
| parent | 93d9ae8a17f2e7b90641cbac28e740afc67d383a [diff] |
in call RtpVideoSenderTests rely on simulated time instead of waiting on an rtc::Event to make tests faster and potentially less flaky Bug: None Change-Id: I04e8fa79761e782f60838b924d40e6d6a104b14b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168644 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30540}
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.