Truncate CSRC list in RtpPacket::SetCsrcs if too large Truncates the CSRC list to a maximum of 15 entries as per the 4-bit RTP header field limitation, and logs a warning when truncation occurs. Also updates the buffer capacity check and allocation to happen prior to writing data to avoid heap buffer overflow. Additionally, ensure that the underlying buffer size is updated before writing the CSRC data. By calling SetSize prior to WriteAt, we prevent potential heap buffer overflows that could occur if the buffer was not sufficiently allocated before the write operation. Bug: chromium:486317116 Change-Id: I2821867f94c0a8d174fb25047ac1f4b94ee5b867 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459943 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47266}
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.