commit | e7b9e6b17dd866323671bbf7f22e3bfcfd4ea2db | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Wed Feb 06 18:23:44 2019 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Feb 06 18:00:39 2019 +0000 |
tree | a99dcf6a75d38177bd1d7e06c0f32414202f6453 | |
parent | d70a1148aecf1ed6d09571ffe65302025e10f022 [diff] |
Move RtpSenderVideo tests to separate file. Also refactor most of the RtpSender tests to not use the frame-level method RTPSender::SendOutgoingData. Bug: webrtc:7135 Change-Id: I9b0af6aa45e9b908d8197e48b143fede7e2804c7 Reviewed-on: https://webrtc-review.googlesource.com/c/121461 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26577}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.