commit | df527fd6a6af9efe722f7500f16f775bfa59782a | [log] [tgz] |
---|---|---|
author | Steve Anton <steveanton@webrtc.org> | Fri Apr 27 22:52:03 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Apr 27 23:29:23 2018 |
tree | 6631aa9602b493e381cd014fbd1a4b45ea40116b | |
parent | de2ed7dc18d5bed8198a6a3ef8a828c3d7556f0e [diff] |
Add e2e test for multiple video tracks without signaling SSRCs This is intended to exercise end-to-end sending with the MID RTP header extension and demuxing by MID. Bug: webrtc:4050 Change-Id: I81edb3687c65f5efce9591fa34cb03522ad675e5 Reviewed-on: https://webrtc-review.googlesource.com/71601 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23062}
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.