commit | 95d12adf379afa9e5bb8d9fe0d89467c29aba0b2 | [log] [tgz] |
---|---|---|
author | Harald Alvestrand <hta@webrtc.org> | Mon Feb 06 12:22:44 2023 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Feb 06 14:00:39 2023 |
tree | 9941aaf3614b900a0e3bb478a1b60d774be0dad0 | |
parent | 4b0d6f908be0f024d2bd6603d77a7e9264601b0b [diff] |
Create unit test for the population of capture_start_ntp_time This verifies that receiving two RTCP SR packets is enough to get a defined capture start time stat. Bug: webrtc:13931 Change-Id: Ib5f7c2954eab6500917f25c44f523d3aedae5e94 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291520 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39261}
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.