| commit | 45087cd23ff5df9122ebdacf9e4c50661adcf3cf | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Thu Mar 01 15:56:57 2018 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 01 17:22:28 2018 +0000 |
| tree | f1caee97b51335be265fe5839d4385eb6446b79e | |
| parent | 546d7f98a5f1498c5865c12795b184d289b65668 [diff] |
Moved retransmission rate limiter to Call class. Ownership of the retransmission rate limiter for video is moved from send side congestion controller to Call. This is to reduce the interface on the rtp transport controller send. Bug: webrtc:8415 Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769 Reviewed-on: https://webrtc-review.googlesource.com/58745 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22254}
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.