commit | 53eae87bf8f576b8fe5563fd7924769ea3f2c9e4 | [log] [tgz] |
---|---|---|
author | Jakob Ivarsson <jakobi@webrtc.org> | Thu Jan 10 15:58:36 2019 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Jan 10 16:28:43 2019 +0000 |
tree | 7c9588ac3f10c198a7e5d6c97d3926e83b99a993 | |
parent | 43f3982d6fcc21b27222bb941eed2e5bc4885225 [diff] |
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.