commit | 696cea084302ee5127e2718a1feb5dfdf2ac4692 | [log] [tgz] |
---|---|---|
author | Erik Språng <sprang@webrtc.org> | Mon Apr 12 08:47:55 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Apr 13 08:37:14 2021 |
tree | 3af3b03c247f7a63cb944cbe0a85f47febf309a7 | |
parent | 5fe0b372bad19b3584b59250d5fd3ca301f4afd0 [diff] |
Refactor some RtpSender-level tests into RtpRtcp-level tests This prepares for ability to defer sequence number assignment to after the pacing stage - a scenario where the RtpRtcp module rather than than RTPSender class has responsibility for sequence numbering. Bug: webrtc:11340 Change-Id: Ife88f60258b9b7cfd9dbd3326f02ac34da8f7603 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214967 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33702}
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.