If the first buffers have no timestamp then the sink position would be
initialized to 0. The source pad might output this buffer, which would then
initialize the source position to 0 too.
Afterwards two buffers with a valid but huge timestamp might arrive before any
of them are output on the source pad. The first one would set the sink position
to a huge value, the second one would notice that the difference between the
huge value and 0 is certainly larger than max-size-time and consider the queue
as full.
Instead, simply don't update the times from buffers without timestamps and
assume whatever was set before is still valid, i.e. the buffer has the same
timestamp as the previous one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7075>
We were storing the probe id in a different structure (DecodebinOutputStream)
than the pad it is targetting (which is in MultiQueueSlot).
The problem is that when re-targetting outputs (to a different slot)... we would
end up having an invalid probe id, or not have a reference to an existing one.
Instead, store the probe id in the same structure as the pad it's targetting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7074>
According to the SPEC:
The frame id numbers (represented in display_frame_id, current_frame_id,
and RefFrameId[ i ]) are not needed by the decoding process, but allow
decoders to spot when frames have been missed and take an appropriate action.
So we should just print out warning and should not return error in parser when
mismatching. The decoder itself is already robust to handle the reference missing.
Fixes#3622
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7052>
Specify "layout" field in src template to make sure it's
set and gets fixated properly if the downstream element
supports both interleaved and non-interleaved caps.
Fixes
gst_pad_set_caps: assertion 'caps != NULL && gst_caps_is_fixed (caps)' failed
critical with e.g.
gst-launch-1.0 rtpdtmfsrc ! rtpdtmfdepay ! audioconvert ! fakesink
Not that the layout really matters in our case since we always
output mono anyway, but non-interleaved requires adding AudioMeta,
so this is the easiest fix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7048>
This was already being used in handle_frame() for errors that happen when queueing a frame for decoding,
let's do the same when a frame is flagged with an error in the output callback.
From quick testing, this makes seeking more reliable (previously, it would sometimes cause a decoding error
and shut the whole decoder down due to GST_FLOW_ERROR).
Also manually sets the max error count to actually stop processing if too many errors occur.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7044>
ReferenceMissingErr is not critical and the simplest solution is to just ignore it. The frame has
the FrameDropped flag set when it occurs, so we can just drop it as usual.
BadDataErr is also not immediately critical, but in its case let's set the ERROR flag,
so the output loop can use GST_VIDEO_DECODER_ERROR to count and error out if it happens too many times.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7044>
Upon fatal errors the loop function will first post an error message
then push out an EOS event.
An application may react immediately to the error message by setting the
state of the pipeline to NULL, meaning by the time we push out the EOS
event PAUSED_TO_READY may have reset the seek seqnum to -1.
While this is harmless, the assertion when setting an invalid seqnum
isn't tidy, fix this by simply not resetting to INVALID as it serves no
practical purpose and the next READY_TO_PAUSED will select a new seqnum
anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7034>
Certain V4L2 drivers can report that a video receiver is seeing
some signal, but that it is unable to synchronize to it. IOW: the driver
can sometimes report V4L2_IN_ST_NO_SYNC and not report V4L2_IN_ST_NO_SIGNAL.
In particular, I've seen the tc358743 (HDMI-to-CSI2 converter) driver
sometimes report this when deployed to a fleet of embedded Raspberry Pis.
The relevant kernel code is in [1]. The video output is not practically
usable when V4L2_IN_ST_NO_SYNC is reported (only visually corrupted frames,
sometimes with random "snow", are received). I assume that this happens when
either the HDMI cable is poorly plugged in or damaged or when a CSI2 FFC
cable is used and is damaged.
The change in this commit is useful for detecting this working-but-not-really
condition in application code. Applications already listening for the "Signal lost"
message will gain the ability to handle this condition.
There seem to be more V4L2 error flags like this, see [2]. However, I do not
have practical experience with them and adding only V4L2_IN_ST_NO_SYNC seems
like a safer option.
[1]: https://github.com/raspberrypi/linux/blob/be8498ee21aa/drivers/media/i2c/tc358743.c#L1534
[2]: https://www.kernel.org/doc/html/v6.6/userspace-api/media/v4l/vidioc-enuminput.html
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7027>
Musl does not implement GNU basename and have fixed a bug where the
prototype was leaked into string.h [1], which resullts in compile errors
with GCC-14 and Clang-17+
| sys/uvcgadget/configfs.c:262:21: error: call to undeclared function 'basename'
ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
| 262 | const char *v = basename (globbuf.gl_pathv[i]);
| | ^
Use glib function instead makes it portable across musl and glibc on
linux
[1] https://git.musl-libc.org/cgit/musl/commit/?id=725e17ed6dff4d0cd22487bb64470881e86a92e7a
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7028>
This fixes a regression introduced by 6c4f52ea20
There are cases where the input stream will be push-based, time-segment and not
have a collection nor caps. This means the event-based checks are not sufficient
to decide when/where to plug in a identity or parsebin to process the input.
For those corner cases we setup a buffer probe to ensure we always end up with
at least a parsebin
Fixes#3609
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7018>
When dealing with push-based inputs, we are now delaying the creation of
parsebin/identity until we get all pre-buffer events.
We therefore can simplify the handling of new pads being linked and only have to
check if upstream can handle pull-based or not.
Avoids creating parsebin for parsed upstream data altogether
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6995>
When we are dealing with parsed inputs (i.e. using identity), we need to ensure
that we have a valid stream collection (and therefore DBCollection) before
anything flows dowsntream.
In those cases, we hold onto those events until we get such a collection.
Fixes#3356
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7002>
This commit separates collection and selections into a new separate structure:
DecodebinCollection.
This provides a much cleaner/saner way of dealing with collections being
updated, gapless playback, etc...
There is now a list of DecodebinCollection in flight, of which two are special:
* input_collection, the currently inputted/merged collection
* output_collection, the currently active collection on the output of multiqueue
Handling GST_EVENT_SELECT_STREAMS is split, by looking for the collection to
which it applies. And the requested streams are stored in it. IIF that
collection is output_collection we can do the switch, else it will be updated
when it becomes active.
Detecting which collection/selection is active is done by looking at the
GST_EVENT_STREAM_START on the output of the multiqueue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7002>
* Move the handling of GST_EVENT_STREAM_START on a slot to a separate function
* There was a lot of usage of `gst_stream_get_stream_id()` for the slot
active_stream. Cache that instead of constantly querying it.
* Rename the variables in `handle_stream_switch()` to be clearer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7002>
* Centralize associating an output to a slot in one function, including properly
resetting those fields
* Rename functions to be more explicit
* Move code to "reset" an output stream into a dedicated function (will be used
later)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7002>
