This should start making navigating the tree a little easier to start
with, and we can then move to allowing building specific groups of
plugins as well.
The plugins are moved into the following hierarchy:
audio
/ gst-plugin-audiofx
/ gst-plugin-claxon
/ gst-plugin-csound
/ gst-plugin-lewton
generic
/ gst-plugin-file
/ gst-plugin-sodium
/ gst-plugin-threadshare
net
/ gst-plugin-reqwest
/ gst-plugin-rusoto
utils
/ gst-plugin-fallbackswitch
/ gst-plugin-togglerecord
video
/ gst-plugin-cdg
/ gst-plugin-closedcaption
/ gst-plugin-dav1d
/ gst-plugin-flv
/ gst-plugin-gif
/ gst-plugin-rav1e
gst-plugin-tutorial
gst-plugin-version-helper
This commit fixes several issues with the `Ts*Src` elements.
The pause functions used cancel_task which breaks the Task loop at await
points. For some elements, this implies making sure no item is being lost.
Moreover, cancelling the Task also cancels downstream processing, which
makes it difficult to ensure elements can handle all cases.
This commit reimplements Task::pause which allows completing the running
loop iteration before pausing the loop.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/277#note_439529
In the Paused state, incoming items were rejected by TsAppSrc and DataQueue.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/277#note_438455
- FlushStart must engage items rejection and cancel the Task.
- FlushStop must purge the internal stream & accept items again.
If the task was cancelled, `push_prelude` could set `need_initial_events`
to `true` when the events weren't actually pushed yet.
TsAppSrc used to renew its internal channel which could cause Buffer loss
when transitionning Playing -> Paused -> Playing.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/issues/98
Returning 0 as the max latency in those sources is incorrect,
and may lead to sinks incorrectly complaining about insufficient
buffering elements.
Reproduce with:
gst-launch-1.0 ts-udpsrc port=50000 address=127.0.0.1 \
caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8" ! \
rtppcmadepay ! alawdec ! autoaudiosink
gst-launch-1.0 audiotestsrc do-timestamp=true samplesperbuffer=400 ! \
alawenc ! rtppcmapay max-ptime=50000000 min-ptime=50000000 ! \
udpsink host=127.0.0.1 port=50000
Logs:
Not enough buffering available for the processing deadline of 0:00:00.020000000,
add enough queues to buffer 0:00:00.020000000 additional data.
Shortening processing latency to 0:00:00.000000000.
This then causes glitches, there are many other ways for the problems
to manifest.
Not returning NoPreroll in that transition causes downstream sinks
to wait for preroll forever.
Reproduce with:
gst-launch-1.0 audiotestsrc ! udpsink host=127.0.0.1 port=50000
gst-launch-1.0 ts-udpsrc address=127.0.0.1 port=50000 ! fakesink
ctrl + C in the receiver pipeline -> hangs forever
The IO Context has to be alive longer than the other parts
of the state. Otherwise a deadlock can happen between shutting down
the IO context (thread join while the state lock is held) and stuff
happening on the IO context (which might take the state lock).
This reverts commit 083948e8ea1471f1d8a013a225200729cfcef2a9.
This has no effect because core is reordering events to have
stream-start and others always first.
