commit | fa5ec8d20d8dbf0f2e9ebe1ad4da970dd7397807 | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Fri Sep 07 08:57:26 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Sep 07 09:24:18 2018 |
tree | ef73da6469685b3db9aee99518961c795f3ebf64 | |
parent | 5e2e66d8a0fd5e1bf9b3efc54a94cba3e7088b00 [diff] |
Use signed integers for limiting packet size in video packetizers Using signed integers allows to centralize checking of edge cases in RtpPacketizer::SplitAboutEqually and reduce chance of hitting issues with size_t underflow Bug: webrtc:9680 Change-Id: Ic05bf0a9565a277c4608f43061ca46cf44e82d08 Reviewed-on: https://webrtc-review.googlesource.com/98602 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24618}
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.