commit | a86cef7e2ccaa1ff4db0d26754ea70634c94dd70 | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Thu Jan 14 17:35:41 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Jan 15 10:58:20 2021 |
tree | 156833edaf0d0940f7d9a18da19123ec029ec5eb | |
parent | 87e9f6e666b00ecd714434643d7d7eb946d17b49 [diff] |
Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding Bug: webrtc:12336 Change-Id: Icae229b957c2bfcc410788179a504c576cfde151 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201736 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32995}
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.