commit | ce79f873e72b30c26ef63cfbc54a71f9afbec6e9 | [log] [tgz] |
---|---|---|
author | Per Kjellander <perkj@webrtc.org> | Fri Dec 02 15:07:09 2022 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Dec 06 14:35:39 2022 |
tree | 20437f320e540c2a7375d9e9b8baceb716936b47 | |
parent | fcbf3724ebb8d6805f1dbc147077d0a3c26b90bd [diff] |
Update Call Scenario test framwork to use defaults from Chrome Default send transport wide sequence numbers on audio Use 32kbit/s audio. Pace in bursts 40ms, See chromium:1354491 Bug: none Change-Id: I40b1305ce71478749723a53f6cc84669ddf930e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285883 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38827}
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.