commit | d19513f3ffbb939fd56b5377b678bb31d3154e14 | [log] [tgz] |
---|---|---|
author | Artem Titov <titovartem@google.com> | Wed Mar 25 10:53:41 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Mar 25 11:38:47 2020 |
tree | dcbe953ba56bc45aa1837067d3538adfd9eda496 | |
parent | c8fbd899bdd716903a09a9ce8922c47f1517584b [diff] |
Move calculation of target_encode_bitrate to DefaultVideoQualityAnalyzer To migrate on new GetStats API and properly support target encode bitrate for regular, simulcast and svc cases we need to calculate it inside video quality analyzer getting values from SetRates in VideoEncoder. Bug: webrtc:11381 Change-Id: Ia37acac764ed3c30f64cdbfda8906d543fa03ae2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171501 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30881}
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.