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![]() 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} |
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rtp-hdrext | ||
sdp-ext | ||
logging.md | ||
README.md |
WebRTC native code
The WebRTC Native Code package is meant for browser developers who want to integrate WebRTC. Application developers are encouraged to use the WebRTC API instead.
The WebRTC native code can be found at https://webrtc.googlesource.com/src.
The change log is available at https://webrtc.googlesource.com/src/+log
Please read the License & Rights and FAQ before downloading the source code.
The WebRTC issue tracker can be used for submitting bugs found in native code.