The internal elements are only created when caps on both video and subtitle pads
are known.
Prior to that, a GST_QUERY_CAPS on a video sink pad would just return ANY
instead of giving a hint of what downstream can actually handle and
prefers. This could result in upstream elements (such as decoders) deciding on
chosing (in the best cases) a non-optimal caps or (in the worst case) caps that
couldn't be handled by the elements downstream of subtitleoverlay.
In order to fix that, we assume that all subtitle "elements" handle the subtitle
overlay composition feature/meta and handle `GST_QUERY_CAPS` ourselves if the
internal elements aren't present yet.
Fixes#3176
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5834>
We access fields that are protected by the lock and this was already
held in other places where we call the method. I have got cases where
we get the following stack/assertion:
```
#0 g_logv (log_domain=0x7fb9d84e6cd5 "GStreamer", log_level=G_LOG_LEVEL_CRITICAL, format=<optimized out>, args=args@entry=0x7fb9d4de54e0) at ../glib/gmessages.c:1433
#1 0x00007fb9d802d0f3 in g_log (log_domain=<optimized out>, log_level=<optimized out>, format=<optimized out>) at ../glib/gmessages.c:1471
#2 0x00007fb9d845bc2c in gst_pad_send_event (pad=0x7fb98c01e050, event=0x7fb9c4105b90) at ../subprojects/gstreamer/gst/gstpad.c:6096
#3 0x00007fb9d6541c35 in gst_uri_decode_bin3_set_uri (dec=0x7fb9bc450960 [GstURIDecodeBin3], uri=0x7fb9c40f5410 "file:///var/home/thiblahute/devel/gstreamer/gstreamer/subprojects/gst-integration-testsuites/medias/defaults/mp4/mp3_h264.0.mp4") at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:1918
#4 0x00007fb9d6540c40 in gst_uri_decode_bin3_set_property (object=0x7fb9bc450960 [GstURIDecodeBin3], prop_id=1, value=0x7fb9d4de57b0, pspec=0x7fb9bcee5280 [GParamString]) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:1569
#5 0x00007fb9d7f8f73d in object_set_property (object=0x7fb9bc450960 [GstURIDecodeBin3], pspec=0x7fb9bcee5280 [GParamString], value=0x7fb9d4de57b0, nqueue=0x7fb9c40d0c40, user_specified=<optimized out>) at ../gobject/gobject.c:1794
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5968>
In `parse_chain_output_probe()` the corresponding input stream might receive EOS
and thus be removed before the actual pad is removed. So we cannot assert about
this in `parsebin_pad_removed_cb()`.
Also, driving-by, protect `find_input_stream_for_pad()` with the selection lock
similarly to other functions accessing the input streams list.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5887>
This causes a lot of nasty side effects (like decoders assuming they are
actually linked downstream).
The reason why this was done was to check whether a decoder could handle the
actual caps, but this is the wrong way to do it.
The proper way to query whether a decoder can handle certain caps is via
`GST_QUERY_ACCEPT_CAPS` which is already done just before.
Partially reverts !4677 and partially fixes#3160
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5821>
Whenever that caps changes does not imply that a new segment will start.
Don't reset the last_ts if only the caps have changed. This fixes issues
if you have a stream without only first frame with TS=0, and have resolution
change happening. This was a regression introduced by !3059, which issue was
described in #1352. The reported issue is still fix after this change.
Fixes#1034
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5687>
The fake video decoder ignores input bitstream except
to enforce caps restrictions. It reads video width,
height and framerate from caps. Then it just pushes
video frames without doing any decoding.
The fake video decoder just draws a snake moving from
left to right in the middle of the frame. This is a
light weight drawing while it still provides an idea
about how smooth is the rendering.
The fake video decoder inherits from GstVideoDecoder.
It is useful to measure how smooth will be the whole
rendering pipeline if you had the most efficient video
decoder. Also useful to bisect issues for example when
suspecting issues in a specific video decoder.
Handles mpeg2, mpeg4, h263, h264, theora, vp8, wmv3, msmpeg,
flash-video, vp6, vp9, wmv1, wmv2, divx but more can be
added if needed.
For now it can only output RGBA, RGBx, BGRA, BGRx.
Its rank is 0 (none) but I added a property to change it so
that it can be selected by decodebin.
gst-launch-1.0 fakevideodec rank=512 \
playbin uri=http://clips.vorwaerts-gmbh.de/big_buck_bunny.mp4http://bugzilla.gnome.org/show_bug.cgi?id=723778Closes#679
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5636>
Update connection-speed at runtime in playbin, uridecodebin and decodebin
also do the same thing in urisourcebin.
With contributions from Philippe Normand <philn@igalia.com> (build fixes and
rebase on mono-repo).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4713>
Back in the mists of time[1], we switched `giostream*` elements to not close the
stream on stop() so that applications that needed a handle to the stream after
the element stopped had it.
Unfortunately, we also have cases[2] where waiting for the element to be
finalized is too late for the stream to be closed.
In order to not change the behaviour of the element, we add a property to allow
users to select the desired behaviour.
[1]: https://bugzilla.gnome.org/show_bug.cgi?id=587896
[2]: gst-plugins-rs#423
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5372>
The current limit is `x10`, which allows just `+20 dB` of gain.
While it may seem sufficient, this came up as a problem
in a real-world, non-specially-engineered situation,
in strawberry's EBU R 128 loudness normalization.
(https://github.com/strawberrymusicplayer/strawberry/pull/1216)
There is an audio track (that was not intentionally engineered that way),
that has integrated loudness of `-38 LUFS`,
and if we want to normalize it's loudness to e.g. `-16 LUFS`,
which is a very reasonable thing to do,
we need to apply gain of `+22 dB`,
which is larger than `+20 dB`, and we fail...
I think it should allow at least `+96 dB` of gain,
and therefore should be at `10^(96/20) ~= 63096`.
But, i don't see why we need to put any specific restriction
on that parameter in the first place, other than the fact
that the fixed-point multiplication scheme does not support volume
larger than 15x-ish.
So let's just implement a floating-point fall-back path
that does not involve fixed-point multiplication
and lift the restriction altogether?
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5063>
When switching from a raw stream to an encoded stream we need to make sure the
slot is unlinked, there is code in place for this but it wasn't triggered
because the slot being reconfigured wasn't advertised as linked beforehand.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5126>
Do not attempt to send a streams-selected message when reassigning
an output slot in case upstream signalled that it is handling stream selection.
In this case decodebin3 doesn't keep track of stream
collections (`dbin->collection` is NULL).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5059>
Setting the input field on the empty slot prevents future linking of it and will
result in flow errors later on.
This was observed in WebKit's MediaStream source element, when it changes the
caps on one of its associated streams, from an encoded format to a raw video
format. The associated stream-id on the sticky stream-start event doesn´t
change, but the element creates a new GstStream with a different ID and sets it
on the stream-start event. Stream parsing is disabled in urisourcebin, so
decodebin3 handles the parsing. Without this patch we would end-up with unlinked
pads in decodebin3 after switching to the raw video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5048>
This is a small optimization and avoids restarting the next parsing
iteration on already accepted data.
On its own it would also fix ZDI-CAN-20968 (see previous commit) but the
previous commit independently is also a valid fix for it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4895>
Add support for generation of 10/12/14/16 bit bayer test pattern.
The implementation is rather simplistic, just take the ARGB
input, generate 16-bit data out of it instead of 8-bit, shift
them as required by the output bitness, and apply endian swap.
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
filter the error message and don't forward it as there might be a
following candidate decoder that can be used.
If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
store the latency message and handle it after decoder is accepted.
This is to avoid the selection lock failure if decodebin3 needs to
handle latency message for candidate decoders when sending sticky event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
Send sticky events to the new created decoder after it switches
to PAUSED state. It it fails, just skip this decoder and try the
next one until finding one that works. Otherwise remove this
failing stream after trying all decoders and no one can work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
