If collecting the backchannel stream after the stream has begun
preparing, make sure to join it to the RTSP media bin, as is
done for dynamic payloaders in the pad-added handler. Add a
new internal stream creation method for it, that could be
publically exposed later if it proves useful.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8378>
When prerolling / blocking an RTSP media, don't consider receiver
streams like onvif backchannel in the 'is everything blocked' check,
as those streams won't receive data until after the RTSP stream
is running.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8378>
When decoupling send/receive loops in rtsp-media using appsink/appsrc,
put the appsink in the bin before linking it, as this code path
had that wrong.
For both sender and receiver paths, if the pipeline already started
also add the other appsrc/appsink of the pair to the pipeline
immediately instead of putting it in the pending elements list,
as they won't get added later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8378>
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.
With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.
Fixes#2443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
Previously it was possible that a shared media was just in the process
of being unprepared because the last client disappeared, while another
client retrieved it from the cache and then tried to use it. Unless the
media was reusable this would've then failed unnecessarily.
To avoid this it is necessary to lock the media directly in
gst_rtsp_media_factory_construct() and return a locked media. After
locking the cached media it is necessary to check if the media was ever
unprepared or is actually reusable and based on that either reuse it or
create a new media.
This minimally changes the gst_rtsp_media_factory_construct() API to
always return a locked media, and adds a new
gst_rtsp_media_can_be_shared() function to check if a media can actually
be shared in practice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().
Fixes#1294
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
Previously suspended medias immediately reached the UNPREPARED state
without going through the media's unprepare() vfunc. This didn't allow
the media subclass to do any additional cleanup, and for example the
shutdown-eos property of GstRTSPMedia was ignored.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
