commit | f0adf38d51a18abbbe3a6a2041ef21861cc3ea96 | [log] [tgz] |
---|---|---|
author | Alessio Bazzica <alessiob@webrtc.org> | Tue Mar 23 08:36:51 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Mar 23 10:53:59 2021 |
tree | a718f7444fa795cb1374ac4674801a4269877f20 | |
parent | 3889de1c4c7ae56ec742fb9ee0ad89657f638169 [diff] |
Fix timestamps for the remote outbound audio stream stats The timestamps must correspond to the time elapsed since the Unix epoch and not since Jan 1 1900 (which is used by the RTCP SRs). Bug: webrtc:12529,webrtc:12605 Change-Id: I6013cf3d9bf9915b5f5db8661f7b2b84231cca57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212606 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33538}
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.