| commit | 4f08faae82ccb0a252e83aec2baa9598c9f1716a | [log] [tgz] |
|---|---|---|
| author | Anton Sukhanov <sukhanov@webrtc.org> | Tue May 21 11:12:57 2019 -0700 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue May 21 18:58:33 2019 +0000 |
| tree | 6ac64aa22fd70c96572a25c4f6093b7282320e06 | |
| parent | 4880e157074cdba8d39b2aabb26879df5216da58 [diff] |
Introduce MediaTransportConfig Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files. TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change. Bug: webrtc:9719 Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28016}
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.