commit | 99a20962483581349f8cd1c37ef7b4cc79dd9ca5 | [log] [tgz] |
---|---|---|
author | Ivo Creusen <ivoc@webrtc.org> | Mon Oct 07 11:18:18 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Oct 07 12:26:44 2019 |
tree | 17e9a660e13da223b1deb9e38fc702fc012bd523 | |
parent | 35cf9e76a88fbc60bcebcc204e6f83b3c0bc99bd [diff] |
Added support for skipping get_audio events, adding dummy packets and setting a field trial string. Bug: webrtc:10337 Change-Id: I0507da4d955daa914af774c946be16a4168be21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150780 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29392}
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.