commit | 10403ae87c73e99c5d2b6be74709263bc4e1f32d | [log] [tgz] |
---|---|---|
author | Jakob Ivarsson <jakobi@webrtc.org> | Tue Nov 27 15:45:20 2018 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Nov 27 19:49:48 2018 +0000 |
tree | e21a547bd770eb8c71d3289c2c15761778d12c63 | |
parent | c7f1a0af92e10a2f3b0d65b4b7c73001329ee1b5 [diff] |
Add PeerConnection option to configure minimum audio jitter buffer delay. Note that this value will override the minimum delay that is used for audio/video sync. Bug: webrtc:10053 Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33 Reviewed-on: https://webrtc-review.googlesource.com/c/112121 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25805}
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.