commit | aa1e7c284e469274ef6f1100be405443364be6a1 | [log] [tgz] |
---|---|---|
author | Piotr (Peter) Slatala <psla@webrtc.org> | Tue Oct 16 17:04:45 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Oct 16 18:33:47 2018 |
tree | f32b0272d216700d42dec83645d32acd500ecaf7 | |
parent | da65ed2adcfa57ff3288ce01c1602c973fcab00d [diff] |
Allow 'use_media_transport' to be modified on PeerConnection before local/remote description are set. Downstream clients will be able to use GetConfiguration() and SetConfiguration() to enable MediaTransport. Bug: webrtc:9719 Change-Id: Ica77b25222732df211dc492dac848342d3f90ff2 Reviewed-on: https://webrtc-review.googlesource.com/c/106423 Commit-Queue: Peter Slatala <psla@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25221}
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.