| commit | 26617bef59b6efed61472f05abb3a5a59c76cf1c | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Thu Jan 23 17:24:56 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Jan 27 16:02:33 2025 |
| tree | 34e7ac17876a7a778edbd953ac624c0ea02ebf55 | |
| parent | 13170bd177dd07330d39db63ea8f3129d17cf45a [diff] |
Make AV1 even payload size default-on when packetizer is used directly This flip default behavior for webrtc users that create packetizers without help of RtpSenderVideo class. Bug: webrtc:42226301 Change-Id: I42fe696039334672b7d0b0ed1f87a52c3f6bf5ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374883 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43807}
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.