commit | 0429f78992f7395e7785c46bf29dafd575d4c113 | [log] [tgz] |
---|---|---|
author | Sebastian Jansson <srte@webrtc.org> | Thu Oct 03 16:32:45 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Oct 03 17:38:22 2019 |
tree | 94ecc179049a85fd3e44a045b538069fa382e305 | |
parent | 78c82a40409dc147cec33241b8af48115bfdb76f [diff] |
Base overhead calculation for audio priority rate on available data. This improves the accuracy of the priority bitrate on IPv6 networks and when the min frame length is longer than 20 ms. Unless either of those are true, there's no significant change in behavior. Bug: webrtc:11001 Change-Id: I29530655cb607a8e7e8186431cd9362ca397910b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155521 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29375}
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.