commit | 2d319df955f2105ab3d2a8370d53d14d3288c996 | [log] [tgz] |
---|---|---|
author | Tommi <tommi@webrtc.org> | Thu Dec 23 21:45:50 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Jan 03 15:32:33 2022 |
tree | f0b40eee04716bf8750e835de7fac93c61e5eed2 | |
parent | c3795ff2161e5ecc34db01c50de2bf97ab0531c0 [diff] |
Add a sequence checker and a few checks to RtpVideoSender. Moving the following TODO into a bug for tracking. // TODO(holmer): Remove mutex_ once RtpVideoSender runs on the // transport task queue. Bug: webrtc:13517 Change-Id: Ie3deb1276c2edaf9894001501ce79409f5437dd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242368 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35612}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.