commit | 648d28ad62573c7b1a1e5b05a6122227f7272950 | [log] [tgz] |
---|---|---|
author | Tim Haloun <thaloun@google.com> | Thu Oct 18 16:52:22 2018 -0700 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Oct 19 21:47:55 2018 +0000 |
tree | eef56b6be43e69766df38b10314be89436aff775 | |
parent | 51cc30c12411ef8e3a64c91c17f748f3e83ae491 [diff] |
Media engine and channel support for per-channel dscp values, specified by RtpParameter - Similar to network priority - Still requires MediaConfig.enable_dscp = true (i.e. googDscp == true to peerconnection) - Needs followups 1) Specify value in chrome renderer js idl 2) disable audio bwe when value differs from video 3)remove googDscp guard Bug: webrtc:5008 Change-Id: Ibdcbb1183f0ca2ae85e3bced6d0aedbccae3ced4 Reviewed-on: https://webrtc-review.googlesource.com/c/93560 Commit-Queue: Tim Haloun <thaloun@chromium.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25280}
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.