| commit | 492ee28b731e33913acbd4bd62dd45432df58af2 | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <stefan@webrtc.org> | Thu Oct 27 17:19:20 2016 +0200 |
| committer | Stefan Holmer <stefan@webrtc.org> | Thu Oct 27 15:19:34 2016 +0000 |
| tree | d4eb765825ec1d91576d0326d7a6bf74b49c8e05 | |
| parent | 9890a5861f6ecf9ba16f95225b88c2aeaf5da2a1 [diff] |
Use bayesian estimate of acked bitrate. This helps a lot to avoid reducing the bitrate too quickly when there's a short period of very few packets delivered, followed by the rate resuming at the regular rate. It specifically avoids the BWE going down to super low values as a response delay spikes. BUG=webrtc:6566 R=terelius@webrtc.org Review URL: https://codereview.webrtc.org/2422063002 . Cr-Commit-Position: refs/heads/master@{#14802}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.