| commit | 3589135d09261187663183ee3f01cd14852d8efa | [log] [tgz] |
|---|---|---|
| author | Philipp Hancke <phancke@meta.com> | Mon Jun 23 20:06:06 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Jun 25 15:55:29 2025 |
| tree | 21716482c70defca72e0e20da649dcea1c5eb7d8 | |
| parent | cb88340bacc1292532a52a5713c01df85d8517fd [diff] |
sdp munging: detect rtcp-mux and rtcp-rsize changes Changing rtcp-mux is sufficiently hard that is should be unused. rtcp-rsize is newer for audio than for video so measurements are split up per-type. Bug: chromium:40567530 Change-Id: I66068f9ac51fc18b652ab65a88222ed739df8230 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/397661 Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45023}
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.