Harden payload capacity and reduction checks in RTP packetizers Ensure packet reduction lengths cannot be negative when computing payload limits in RTPSenderVideo and when determining needed payload sizes during aggregation. Additionally, add a bounds check against available payload capacity before copying aggregated fragments in RtpPacketizerH265. Bug: chromium:567088927 Change-Id: Ibd57aacce9ef79379fa627903d4aed89c9fbf489 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/506260 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48750}
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.