| commit | a166a353fbc3091c5d227b52802ad75b0ebcad23 | [log] [tgz] |
|---|---|---|
| author | Sam Zackrisson <saza@webrtc.org> | Mon Jul 06 15:46:36 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Jul 06 17:05:25 2020 |
| tree | b3740b778907d9af43003f564745198d6505efdf | |
| parent | b3a6816b3ef976a2b127713d590777d6b41fea7e [diff] |
webrtc::AudioSendStream: Add lock annotation to audio_level_ This is a follow-up CL to https://webrtc-review.googlesource.com/c/src/+/176741 Bug: webrtc:11567 Change-Id: Ic64aec56534efc3229a1d9fa61552db4b83cae4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178780 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31638}
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.