| commit | 185910953ae50c474c5f27d0eaefe48cfa46ae59 | [log] [tgz] |
|---|---|---|
| author | Emil Vardar <vardar@google.com> | Thu Sep 12 11:44:37 2024 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Sep 12 13:31:04 2024 |
| tree | 4aaf1f925d3e9bbad5e276fac50040507f36739b | |
| parent | fb0da3a2aa9c6819d81028b1174a81d89c725087 [diff] |
Specify in which RTP packet corruption score will be sent on. See e.g. this: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc;l=304;bpv=1;bpt=0?q=webrtc%2Fmodules%2Frtp_rtcp%2Fsource%2Frtp_sender_video.cc&ss=chromium%2Fchromium%2Fsrc:third_party%2Fwebrtc%2F. One needs to know if the extension will be added to the first or last packet. Furthermore, one can see that other extensions add it as a note at the bottom, which I follow here. See e.g. http://www.webrtc.org/experiments/rtp-hdrext/video-content-type Bug: webrtc:358039777 Change-Id: I7523f5e6b267528a1389bcbde6ee6fa22fb3233a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362060 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Fanny Linderborg <linderborg@webrtc.org> Commit-Queue: Emil Vardar (xWF) <vardar@google.com> Cr-Commit-Position: refs/heads/main@{#43013}
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.