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df8c29e340
There is generally no requirement to ignore RTCP SR if the RTP time of the SR differs a lot from the last received RTP packet. The mapping between RTP and NTP time stays valid until there was a stream reset, in which case we wouldn't use that information anyway. When using rtcp-sync-send-time=false the default of 1s difference can easily be exceeded, e.g. if encoding of the stream after capture adds more than 1s of latency. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543> |
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| .. | ||
| gstrtpbin.c | ||
| gstrtpbin.h | ||
| gstrtpdtmfmux.c | ||
| gstrtpdtmfmux.h | ||
| gstrtpfunnel.c | ||
| gstrtpfunnel.h | ||
| gstrtphdrext-clientaudiolevel.c | ||
| gstrtphdrext-clientaudiolevel.h | ||
| gstrtphdrext-mid.c | ||
| gstrtphdrext-mid.h | ||
| gstrtphdrext-ntp.c | ||
| gstrtphdrext-ntp.h | ||
| gstrtphdrext-repairedstreamid.c | ||
| gstrtphdrext-repairedstreamid.h | ||
| gstrtphdrext-streamid.c | ||
| gstrtphdrext-streamid.h | ||
| gstrtphdrext-twcc.c | ||
| gstrtphdrext-twcc.h | ||
| gstrtpjitterbuffer.c | ||
| gstrtpjitterbuffer.h | ||
| gstrtpmanager.c | ||
| gstrtpmux.c | ||
| gstrtpmux.h | ||
| gstrtpptdemux.c | ||
| gstrtpptdemux.h | ||
| gstrtprtxqueue.c | ||
| gstrtprtxqueue.h | ||
| gstrtprtxreceive.c | ||
| gstrtprtxreceive.h | ||
| gstrtprtxsend.c | ||
| gstrtprtxsend.h | ||
| gstrtpsession.c | ||
| gstrtpsession.h | ||
| gstrtpssrcdemux.c | ||
| gstrtpssrcdemux.h | ||
| gstrtpst2022-1-fecdec.c | ||
| gstrtpst2022-1-fecdec.h | ||
| gstrtpst2022-1-fecenc.c | ||
| gstrtpst2022-1-fecenc.h | ||
| gstrtputils.c | ||
| gstrtputils.h | ||
| meson.build | ||
| rtpjitterbuffer.c | ||
| rtpjitterbuffer.h | ||
| rtpsession.c | ||
| rtpsession.h | ||
| rtpsource.c | ||
| rtpsource.h | ||
| rtpstats.c | ||
| rtpstats.h | ||
| rtptimerqueue.c | ||
| rtptimerqueue.h | ||
| rtptwcc.c | ||
| rtptwcc.h | ||