| commit | 99a71f49c03a3148e29ba5e815ffd5a64afa6e90 | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Mon Jul 19 13:20:46 2021 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Jul 19 14:27:27 2021 |
| tree | e55729614dfa50c00db82e43ffca4822add992b4 | |
| parent | 4261a73b56ada3608e8bec550ee60697d34fd614 [diff] |
Move helpers to parse base rtp packet fields to rtp_rtcp module rtp_rtcp_format is lighter build target than rtc_media_base and a more natural place to keep rtp parsing functions. Bug: None Change-Id: Ibcb5661cc65edbdc89a63f3e411d7ad1218353cc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226330 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34504}
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.