| commit | d3dc33415dba12a7b26bae4b667970468fcb256b | [log] [tgz] |
|---|---|---|
| author | Tony Herre <herre@google.com> | Mon Nov 17 10:06:40 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Nov 18 07:38:36 2025 |
| tree | 33e29a630631f958f8d9d3fe544f650dba47ae8f | |
| parent | 625877737de940ae38d1b43ba7d17b2f5abe02bc [diff] |
Add a match_any RtpDemuxerCriteria Allows sinks like a DatagramConnection to consume all received RTP packets for an RtpTransport - see following cl. Bug: chromium:443019066 Change-Id: Icdb56b86b07f147e0e53bdf33fa9746678a51e14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/421980 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#46213}
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.