gstreamer-rtsp-server
RTSP server based on GStreamer
- Sources inherited from project openSUSE:Backports:SLE-15-SP4
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Source Files
Filename | Size | Changed |
---|---|---|
gst-rtsp-server-1.20.1.tar.xz | 0000358032 350 KB | |
gstreamer-rtsp-server-rpmlintrc | 0000000228 228 Bytes | |
gstreamer-rtsp-server.changes | 0000028729 28.1 KB | |
gstreamer-rtsp-server.spec | 0000003552 3.47 KB |
Latest Revision
Yuchen Lin (maxlin_factory)
accepted
request 967197
from
Antonio Larrosa (alarrosa)
(revision 2)
Update gstreamer-rtsp-server to 1.20.1 to match the version submitted to SP4 - Remove BuildRequires: hotdoc and disable the doc generation. It's really not used at all. - Update to version 1.20.1: + Fix race in rtsp-client when tunneling over HTTP - Update to version 1.20.0: + GstRTSPMediaFactory gained API to disable RTCP (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property). Previously RTCP was always allowed for all RTSP medias. With this change it is possible to disable RTCP completely, irrespective of whether the client wants to do RTCP or not. + Make a mount point of / work correctly. While not allowed by the RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the wild. It is now possible to use / as a mount path in gst-rtsp-server, e.g. rtsp://example.com/ would work with this now. Note that query/fragment parts of the URI are not necessarily correctly handled, and behaviour will differ between various client/server implementations; so use it if you must but don't bug us if it doesn't work with third party clients as you'd hoped. + multithreading fixes (races, refcounting issues, deadlocks). + ONVIF audio backchannel fixes. + ONVIF trick mode optimisations. + rtspclientsink: new "update-sdp" signal that allows updating the SDP before sending it to the server via ANNOUNCE. This can be used to add additional metadata to the SDP, for example. The order and number of medias must not be changed, however.
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