commit | 6026f05ef1cad2975ee215262c0cd34fc8333a27 | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Tue Oct 16 14:22:33 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Oct 16 15:32:37 2018 |
tree | 60c20b26df3a40aeb1c1a96db6f265e661340196 | |
parent | f5e767dbbce2592754871640fb46bffa48ca5d9a [diff] |
Calculate max payload size for an rtp packet to fit full video frame instead of sometimes incorrectly guessing it Bug: webrtc:9868 Change-Id: I8b15ecca4c660d83ea129dc9df6ec174ad83b4c6 Reviewed-on: https://webrtc-review.googlesource.com/c/106281 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25213}
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.