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Specify in which RTP packet corruption score will be sent on.

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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}
This commit is contained in:
Emil Vardar 2024-09-12 11:44:37 +00:00 committed by WebRTC LUCI CQ
parent fb0da3a2aa
commit 185910953a
1 changed files with 2 additions and 0 deletions
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docs/native-code/rtp-hdrext/corruption-detection/README.md
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@ -350,3 +350,5 @@ Regardless of method, the implementation at the send side SHOULD strive to set
the filter size and error thresholds such that 99.5% of filtered samples end up
with a delta <= the error threshold for that plane, based on a representative
set of test clips and bandwidth constraints.
Notes: The extension must not be present in more than 1 packet per video frame.