If the caller passed in "audio_%u" instead of a concrete pad name into
gst_element_request_pad_simple() then the pad name will be NULL. In that case
use the pad template name for requesting the pad from splitmuxsink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8697>
`gst/validate/validate.h` includes `gst/validate/gst-validate-media-info.h`,
which in turn includes `gst/pbutils/pbutils.h` but `gstreamer-pbutils-1.0`
was only listed in `Requires.private` field of `gstreamer-validate-1.0.pc`.
This would cause projects linking against `gstreamer-validate-1.0.pc` to fail to find
the headers when using alternative interpretation of pkg-config specification
that only considers private dependencies for include path during static builds,
such as the case e.g. on Nix.
https://gitlab.freedesktop.org/pkg-config/pkg-config/-/issues/28
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8661>
`gst/analytics/analytics.h` includes `gst/analytics/gstanalyticssegmentationmtd.h`,
which in turn `gst/video/video-info.h` but `gst-video-1.0` was only listed
in `Requires.private` field of `gst-analytics-1.0.pc`.
This would cause projects linking against `gst-analytics-1.0.pc` to fail to find
the headers when using alternative interpretation of pkg-config specification
that only considers private dependencies for include path during static builds,
such as the case e.g. on Nix.
https://gitlab.freedesktop.org/pkg-config/pkg-config/-/issues/28
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8661>
The position is no longer duplicated across each pad and pad's segment. The
position is now only updated if it changes in the direction of playback so that
quickly repeated forward seeks do not cause the stream to seek from 0.
Reverse playback is expressly disallowed and an unnecessary extra flush of track
when seeking was removed.
A background task was added to periodically check on the current position and
the media source's buffering levels to keep the ready state up-to-date. The
source buffer no longer needs to trigger this update, it will happen whenever
the element state is READY or higher.
Finally, added proper error reporting when failing to push a buffer and improved
debug logging.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8512>
Since the sample map/track buffer now iterates samples in batches corresponding
to each coded frame group, the logic to feed the tracks is simpler. For media
without delta frames, it's a special case where the coded frame groups are all
size 1.
Now, all it does is skip data until the keyframe group containing the seek point
is found, then feed the track queue with the current sample and all future
samples until EOS or cancellation.
Resync of the iterator when the underlying track is modified is not necessary
because the outer loop attempts to resume feeding track data from where it was
interrupted in case of modification.
Also, the track feed task struct now holds a weak ref to its parent source
buffer to allow the task to cancel itself in any situation where the source
buffer is destroyed before the task is shut down.
Media parsing activity in the append pipeline no longer triggers ready state
recalculation on the msesrc since the msesrc now has a background task that
updates the ready state periodically when it's active which is more efficient in
cases where there is a high volume of samples being processed by the media
parser.
Finally, updated to adapt to track buffer API changes. Some functions previously
passed in a lower bound for sample timestamps. Now the source buffer is
responsible for clipping samples within a desired range of time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8512>
Now when the buffered list is requested, the tolerance for merging two ranges
when there's a small gap between them is MAX(0.1sec, max frame duration * 2).
Previously it was hardcoded to 0.01sec. The specification suggests that it
could be something like the max frame duration * 2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8512>
The source buffer currently has a thread for each track that feeds the track
with all data in the track buffer until EOS is reached.
Each pass over the track buffer currently waits for the EOS to appear when it's
done iterating the track buffer which is too restrictive.
When the source buffer reaches the end of the track buffer, it should wait for
any new data to be processed -- not just an EOS -- then check for cancellation
if the deadline expires without new data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8512>
Both operations now work on coded frame groups (GOPs). This simplifies queueing
of video data. There is rarely any point of dealing with individual video frames
when iterating in DTS order, it's most meaningful to decode or delete whole
coded frame groups at a time, so the sample map will now do that when iterating
by DTS. When iterating in PTS order, the existing behavior is preserved since
that is used for informational purposes, not media processing.
A new private boxed type for coded frame groups was added to provide each data
item to the source buffer. Another possible solution would be creation of a new
GstSample representing the whole group by merging all the samples in a group
into a single sample containing a GstBufferList.
Also, start time filtering was removed from the API since gst_iterator_filter()
can be used by callers to achieve the same result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8512>
h265parser defers linking parameter sets until slice header is parsed.
Thus valid SPS/PPS parsed by h265parser can have no linked
parent parameter set. Apply this behavior to
gst_h265_parser_update_{sps,pps} too
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8673>
When dynamic caps change in the pipeline leads to a new buffer pool
getting negotiated, the change is not propagated to Qt6GLWindow, which
keeps using the old, now defunct, pool.
Unset current pool on Qt6GLWindow in decide_allocation(). This will
trigger a switch to the new pool inside create().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8676>
gst_video_time_code_to_date_time() simply calculated the date time based on
adding the hours/minutes/seconds to the daily jam. This causes a gap every full
minute (except for every 10th minute) with drop-frame timecodes as the first 2
(29.97fps) or 4 (59.94fps) timecodes are skipped (not frames!), e.g. with
29.97fps:
timecode: 12:00:59;28 12:00:59;29 12:01:00;02 12:01:00;03
time : 12:00:59.950 12:00:59.983 12:01:00.017 12:01:00.050
and not
time : 12:00:59.934 12:00:59.968 12:01:00.067 12:01:00.100
|-- gap of 2 frames --|
The correct calculation would be to use gst_video_time_code_nsec_since_daily_jam()
and add that to the daily jam.
Also add a test for this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8649>
Fixes an issue where the webrtcbin would hangup when finalizing due
to the sctpenc hanging up when finalizing. This occurred when the
webrtcbin chose to use a sctp association ID already in use.
The sctpenc would fail to reach the paused state, but startup a task
anyways that was never stopped.
This commit modifies the behavior to not choose sctp association IDs
randomly, and instead only choose one that is free. It also prevents the
sctpenc from starting up that task if it fails to reach the paused state.
Fixes: #4188
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8607>
On Windows, if the path to gst-plugin-scanner.exe contained
whitespace, gstreamer would via CreateProcessW attempt to execute
several files "up" the path tree; e.g. if the scanner path was
"C:\Program Files\gstreamer app\gst-plugin-scanner.exe", it would try
to execute C:\Program, C:\Program.exe, C:\Program Files\gstreamer.exe"
and so on.
This is how CreateProcessW behaves with unquoted whitespace arguments
in lpCommandLine if lpApplicationName is NULL.
By passing the binary path as lpApplicationName instead, the problem
is avoided.
Also quote arguments to gst-plugin-scanner.exe as they are paths as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8614>
PyGObject 3.52.0 moved OverridesProxyModule _introspection_module to __slots__,
causing Segmentation Faults when accessing the field.
Since _introspection_module is used to get Gst.Element but is never actually
used afterward, we fix the issue by removing this part.
Fixes#4314
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8653>
There are 3 32bit integers per entry and not one more for all but the last.
Fixes a regression introduced in 5a9e80c01b4b49c6c7630a6d08b600114f38c0db
when playing back files that have one sample table entry per audio sample.
Merging the sample tables would've always failed because of the too strict size
check and one audio sample per GStreamer buffer would've been output, which
doesn't perform very well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8657>
They're not privileged so won't work anymore.
Using kvm tag for now as per discussion on IRC,
until a better solution comes along in future.
Keep placeholder-job tag on trigger job and
pre-commit checks jobs.
Add kvm tag to fluster job.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8677>
