commit | 9554a7b641cfc5d006c6a56e830134e4caa1b61c | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Fri Feb 05 18:30:16 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Sat Feb 06 21:34:08 2021 |
tree | c455ad3712abfaa9c65873353d1939dec4677856 | |
parent | d42413a4b4636422ba5ea66ba40e7d24fb64f9b1 [diff] |
Account for extra capacity rtx packet might need When calculating maximum allowed size for a media packet. In particular take in account that rtx packet might need to include mid and repaired-rsid extensions when media packet can omit them. Bug: webrtc:11031 Change-Id: I3e7bc36437c23e0330316588d2a46978407c8c45 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206060 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33184}
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.