Add RTC_GUARDED_BY to RtpSender::media_channel_ Annotate the `media_channel_` member in `RtpSenderBase` with `RTC_GUARDED_BY(worker_thread_)`. Refactor usage in methods like `GetParameters`, `SetParameters`, `SetAudioSend`, and `SetVideoSend` accordingly. Additionally: * Annotate `last_transaction_id_` and `observer_` with `RTC_GUARDED_BY(signaling_thread_)`. * Convert `CheckCodecParameters` from a virtual method to a private helper function restricted to the worker thread. * Remove the unsafe assignment of `media_channel_` to `nullptr` in `Stop()`, avoiding a write violation from the signaling thread. Bug: none Change-Id: I748cce8d073ce398d653fca98b39838cf1c2cf33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/434580 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46503}
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.