Video timing simulator: Add RTX support. This change adds RTX reception support to the video timing simulator. This is done by introducing `Receiver`, which demuxes the stream of RTP packets (video + RTX), so that video packets are passed through but RTX packets are decapsulated. The resulting recovered stream is then sent to the existing `Assembler` and then the pipeline continues as usual from there. Other related and required changes: - Update `RtpPacketSimulator` to handle logged packets with RTX OSN. - Update the stream factories to feed simulated packets to the `Receiver`, instead of the `Assembler`. - Enable NACKing in `Assembler` by creating a `NackRequester` and corresponding `NackPeriodicProcessor`. (Without this, the jitter buffer adaptation for frames with recovered packets will be incorrectly done.) - Add simulation-level smoke tests for a lossy RtcEventLog scenario. Bug: b/423646186 Change-Id: I2bdccca07107234c148300629dd45e831ee61158 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432961 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46755}
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.