Move audio and video packet demuxing to the network thread Move RTP packet demuxing in Call to the network thread. By resolving the correct receiver sink earlier in the pipeline, we encapsulate the asynchronous demuxing process while maintaining the autonomy of individual receive streams. To safely handle the cross-thread delivery of packets—where a sink might be resolved on the network thread but destroyed before the worker thread can process the packet—an `RtpSinkValidator` interface is introduced. `Call` implements this validator to track active sinks and verifies their existence prior to packet dispatch, preventing use-after-free bugs. Additionally, a `DisconnectFromNetworkThread` method is added to ensure safe teardown during connection closure. Bug: webrtc:42222117 Change-Id: Ieddee38399d5d8098ae9e03a1a8a6ed3f86783d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/472240 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47733}
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.