commit | 4e268edb533bcbc2f25406ee804473fb3ddd7027 | [log] [tgz] |
---|---|---|
author | Henrik Lundin <henrik.lundin@webrtc.org> | Tue May 08 14:36:33 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Tue May 08 16:05:12 2018 |
tree | c0eb2aa72801bcf1eae11a703ce86d07789d2df1 | |
parent | acb4cba5b1a36e226717827036811c49e846499c [diff] |
Add two new RTP header extensions to neteq_rtpplay This change adds flags and default values for two more RTP header extensions: VideoContentType and VideoTiming. This will silence a number of annoying warnings when running with application logs. Bug: none Change-Id: I9bb01ea2519813d3c47553ecff384141fbede23e Reviewed-on: https://webrtc-review.googlesource.com/75300 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23178}
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.