It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if cancellation should happen and reset this
later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Also implement actual unlocking in s3hlssink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Also actually implement unlock() / unlock_stop() for the sink, and don't
cancel in stop() as unlock() / unlock_stop() would've been called before
that already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
It would be possible that there is no cancellable yet when unlock() is
called, then the setup task is started and it would simply run and being
waited on instead of not being run at all.
To solve this, remember if unlock() was called and reset this in
unlock_stop().
Also make sure to not keep the abort handle locked while waiting,
otherwise cancellation would never actually work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
- generate a new session id for every new client
use the session id in the resource url
- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer
- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
- Add a new structure Session
- manage each producer using a session
- avoid send EOS when a session terminates, instead keep running
waiting for any new producer to connect
- Maintain a bin element per session
- each session bin encapsulates webrtcbin and the decoder if needed
as well as the parser and filter if requested by the application
(through request-encoded-filter)
- this will be helpful to cleanup the session's respective elements
when the corresponding producer terminates the session
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
* Update "translation-languages" property to include G_PARAM_CONSTRUCT
so that it can be applied to initial state.
* Change default "translation-languages" value to be None instead of
cea608 specific one. Transcriberbin will be able to configure initia
state depending on selected mux method if "translation-languages" is
unspecified.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1589>
There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession. This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other. The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
