In RtpVideoStreamReceiver do not rely on RTP sequence number unwrap to be stable Currently this class assumed that if the same RTP sequence number is unwrapped again result would be the same. That might not be true when several packets were inserted in between these two calls and unwrapper changed its state This CL propose instead to unwrap once, and save the result in the intermediate struct. To minimize the change and the risk, only redundant unwrapping is replaced to use unwrapped sequence number Bug: webrtc:353565743 Change-Id: I8a18c8c206a0e16010951cabcf81dd9cb1588eda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357660 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42662}
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.