Replace webrtc::test::Packet with RtpPacketReceived To reduce number of classes that represent an RTP packet. Functionality of simulating payload data that is absent in an RTP dump is moved into RtpFileSource that reads from such file. Functionality to parse RED header is moved to the only user - rtp_analyze Bug: webrtc:42225366 Change-Id: I8b66cfa0df67a776d9285dcb05da9c0183d6a400 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400760 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45169}
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.