| commit | 5efb02b1c6539e2bf1c9cbaf930c3199a69380fd | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Fri Mar 13 16:29:12 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Mar 16 08:14:36 2020 |
| tree | 84682ca827e20f0d2f68680fdd61f79044f32fc4 | |
| parent | b28f0209f9a102878c5064a7c6a4ba758c345f90 [diff] |
Cleanup AddRtpHeaderExtension for RtpSenderVideo make it a member function which allows to reduce number of parameters and simplify accessing more state in the future. Bug: None Change-Id: Iba35125c0c2cf1d6bb67b106c1f73a33ecb8e44e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170366 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30797}
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.