Configure RTX at construction of the RTP module The RTX mode and payload types were being set right after the RTP module had been created. That kept RTPSender's RTX state mutable, which requires a lock for every read of the RTX mode. * Add rtx mode and payload types to RtpRtcpInterface::Configuration * Apply them when constructing RTPSender. * RtpVideoSender now passes the RTX configuration at creation. * Remove SetRtxSendStatus() and SetRtxSendPayloadType() from RtpRtcpInterface and ModuleRtpRtcpImpl2. * Deprecate RTPSender::SetRtxStatus() and SetRtxPayloadType(). Once direct users of RTPSender have migrated, they can be removed and the RTX state made const. Invalid configurations still log an error and leave RTX off. Since all RTX payload types are now applied at once, a single invalid one drops the whole mapping. This keeps RTX off for a simulcast layer whose codec has no RTX payload type, as before. Keeping just the RED mapping would enable RTX without a way to retransmit media packets. Bug: webrtc:42223727 Change-Id: I34be4f765293599fc2358c4ab52cebe1fcdb98d0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507340 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48836}
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.