commit | 7f354f860699375d96da5999b8664e47aa30f4fa | [log] [tgz] |
---|---|---|
author | Åsa Persson <asapersson@webrtc.org> | Thu Feb 04 14:52:15 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 04 17:26:21 2021 |
tree | e001f8026e0647797ffcf0c36f3d3233310a8c60 | |
parent | 3ba7beba29c4e542c4a9bffcc5a47d5e911865be [diff] |
Use bandwidth allocation in DropDueToSize when incoming resolution increases. Use bandwidth allocation instead of encoder target bitrate in DropDueToSize when incoming resolution increases to avoid downgrades due to target bitrate being limited by the max bitrate at low resolutions. Bug: none Change-Id: Ic41b31c1a86911d4e97b61b0cbc41ce0da739bd4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205622 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33168}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.