Add PeerConnection option to enable RTX handling in the audio jitter buffer. Bug: webrtc:10178 Change-Id: I70abce0c7b74124d2b1978d9a5eb8216b6233d1a Reviewed-on: https://webrtc-review.googlesource.com/c/116784 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26203}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.