videoscale does not have convert function, so remove the convert
description in it's classification. Otherwise, if we want use
autovideoconvert to convert colorsapce, autovideoconvert will select
videoscale to do convert and this will cause to fail.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7215>
${libdir}/gstreamer-1.0/include is only valid after installation, but
extra_cflags are added unconditionally, so we can't use that for
include flags.
Instead, let's add the include flag via variables, which are different
for installed and uninstalled pc files.
This is particularly bad for consuming GStreamer via CMake which barfs
on non-existent include paths.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7130>
The video meta API now is a mandatory request for DMA kind negotiation. In
current code, the raw method always adds that video meta API in the query
of propose_allocation(), so we do not need to do the duplicated task here.
Just adding a comment to declare that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6698>
- Align `glib_debug`, `glib_assert` and `glib_checks` options with GLib,
otherwise glib subproject won't inherit their value. Previous names
and values are preserved using Meson's deprecation mechanism.
- Add `extra-checks` and `benchmarks` options in the main project so it
can be inherited in GStreamer subprojects.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1165>
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/7069>
ensure_input_parsebin() has a top comment saying it must be called with
INPUT_LOCK taken, but 2 out of 3 usages of the function call it without
taking that mutex.
This patch adds locking in these two remaining usages.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5279>
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/7010>
The typefind code was rejecting content smaller than 128 bytes making it
impossible to play files with very small srt files.
But those can actually be properly detected so fix typefind to allow
smaller content and try its best with it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6937>
When measuring video latency, one mechanism involves taking a photo
with a camera of two screens showing the test video overlayed with
timeoverlay or clockoverlay. In these cases, if the display's pixel
response time is crappy, you will see ghosting due to which it can be
quite difficult to discern what the current timestamp being shown is.
This commit adds a property that *also* shows the timestamp in
a different (sequentially predictable) location every frame, which
makes it easy to tell what the latest rendered timestamp is.
For bonus points, you can also use the fade-time of the previous frame
to measure with sub-framerate accuracy when the photo was taken, not
just clamped to the framerate, giving you a higher precision latency
value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6935>
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/6953>
Simple gst_gl_sync_meta_wait() is not sufficient to ensure GL commands
are executed before dma-buf devices get to see the buffer.
This is the first step that should make the code behave correctly for
everybody, although there may be performance penalty. In the future we
should introduce a more general sync meta that would allow to move the
waiting from gldownload (the producer) to the sink elements (the
consumers).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6968>
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/6774>
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/6774>
* 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/6774>
* 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/6774>
* Rename the function names to be clearer, with prefixes
* Pass the input (or stream) directly where appropriate
* Document usage, inputs, ownership
* Rename variables for clarity where applicable
* Avoid double lock/unlock if callee can handle it directly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
Simplify its usage by having it directly create the message if the collection
changed. This is what caller were always doing and avoids releasing selection
locks yet-another-time
Also use it in more places to avoid code repetition
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
To simplify the description, I'm assuming we only have two streams: video and audio.
For the video stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(1) => blocked waiting in gst_stream_synchronizer_wait
- FLUSH_START => unblocked
- FLUSH_STOP => stream->wait reset to false
- NEW_SEGMENT(2) => not waiting, since stream->wait is false
Then for the audio stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(2) => blocked waiting in gst_stream_synchronizer_wait for ever.
Note: The first NEW_SEGMENT event and the FLUSH_START, FLUSH_STOP events of the audio stream
are dropped before being received by the streamsynchronizer element, because the decodebin audio pad src
is not yet linked to the playsink audio pad sink.
To fix this deadlock, we don't reset stream->wait to false in the FLUSH_STOP event when it is not
waiting for the EOS of the other streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6763>
In some cases you want to ensure that a specific element factory is used
while requiring some specific caps but this was not possible. You can
now do `qtmux:video/x-prores,variant=standard|factory-name=avenc_prores_ks`
to ensure that the `avenc_prores_ks` factory is used to produce the
'standard' variant of prores video stream.
This also enhances a bit the documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6875>
This counter is incremented once for every segment, meaning it would
e.g. overflow after 24 days when using 1ms segments. Once that happens,
completely wrong positions are reported and invalid memory is handed out
for writing/reading the next segments.
As the affected variables are unfortunately part of the public API of
the struct, a second set of variables is added together with accessor
functions and both variables are kept in sync for backwards
compatibility.
All existing users of the two variables are moved to the new ones but
external code might still run into the overflow.
This also slightly breaks API as external code updating the variables
will have no effect anymore but the only known user of this is
pulsesink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6740>
There's nothing requiring <= 64 channels except for getting the reorder
map and creating a channel mixing matrix, but those won't be possible to
call anyway as channel positions can only express up to 64 channels.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6819>
when pipline is
glvideomixerelement->glcolorconvertelement->gldownloadelement and
glcolorconvertelement is not passthrough, the gl bufferpool between
glvideomixerelement and glcolorconvertelement will not add gl sync meta
during allocating buffer. This will cause that glcolorconvert's inbuf
has no sync meta to wait for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6756>
The assertion that was present before is a bit too harsh, since there is now
a (understandable) use-case where this could happen.
In gapless use-case, with two files containing the same type (ex:audio). The
first one *does* expose a collection with an audio stream, but decoding
fails (for whatever reason).
That would cause us to have configured a audio combiner, which was never
used (i.e. not active).
Then the second file plays and we (wrongly) assume it should be activated
... whereas the combiner was indeed present.
Demote the assertion to a warning and properly handle it
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3389
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6737>
Clarify the fact that `encodebasebin->src_pad` is set when using a static source
pad (`encodebin`) and when not set it's dynamically added source
pads (`encodebin2`).
Fixes usage of encodebin2 when profiles are updated
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6667>
Since https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153 ,
subtitle "decoders" (i.e. which decode to raw text) are no longer auto-plugged
by parsebin.
But if a given format does not have a parser at all, we would end up outputting
non-time/non-parsed outputs.
In order to mitigate the issue, until such parsers are available, we check if
the subtitle stream is in TIME format or not (i.e. whether it comes from a
parser or demuxer). If not, we attempt to plug in a subtitle "decoder".
Fixes#3463
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6592>
Reset the waiting thread counter in all places to be consistent
when sending signal for the audio ring buffer. This fix applies it to
pause, stop and release, which are states that will go into a callback
of the subclass. Having the waiting counter reset will avoid having
executing thread of the same subclass trying to take the mutex when
callong gst_audio_ring_buffer_advance.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6195>
On practice a failure happened due to a race condition, the instance
already have been freed, but it could also happen if the instance
would be null.
Instead of crashing this sanity check is a more suitable option,
since with G_DEBUG=fatal-warnings it will crash too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6475>
This reverts commit 8e923a8e2d.
This caused regressions, see #3303.
Without this commit, osxaudiosrc ! osxaudiosink won't work
right, but since that hasn't really been a huge problem
for years it's probably best to revert this until a proper
solution can be figured out.
(cherry picked from commit f04f86f3ee)
(cherry picked from commit 93255efece)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6405>
None of the GL allocators actually offer a generic alloc() implementation. As a
side effect, they cannot be offered as they don't work with generic video
buffer pool.
Our specialized buffer pool can be dropped by tee or alphacombine as sharing the
same buffer pool over two branch is not supported by the pool API.
Fixes#3372
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6327>
Memory from gst_adapter_map() could live shorter than the GstMemory that the GstBuffer wraps around it, which in lucky
cases 'just' caused a re-use of the same memory for multiple (potentially still in use!) input buffers, but could easily
end up pointing to an already-freed memory.
Manifested when an AudioToolbox encoder kept getting silence inserted in seemingly random circumstances, turned out
to be the memory being re-used by GStreamer at the same time that the AT API was processing it...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
This will mimic the playbin2 behaviour, which sets the "next" entry to be
NULL.
The biggest impact this has is that when going back to READY the current play
entry will be discarded (instead of being kept around for when you go back to
PAUSED/PLAYING).
Fixes#3371
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6324>
find_slot_for_stream_id() will return a slot which has the request stream-id as
active_stream *or* pending_stream (i.e. the slot on which that stream is
currently being outputted or will be outputted).
When figuring out which slot to use (if any) we want to consider stream-id
which *will* appear on a given slot which isn't outputting anything yet the same
way as if we didn't find a slot yet.
Fixes races when doing intensive state changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>
Cocoa version of glwindow only checks the preferred size upon window creation. glimagesink sets the size right before
calling gst_gl_window_show(), which might be way after the window is created in some cases. If the size was set too
late, glimagesink on macOS would remain 320x240 unless manually resized.
This change makes sure to resize the existing window when _show() is called.
Curiously, this has always been an issue, but went from manifesting every once in a while to being almost completely
broken once old event loop workarounds were removed and gst_macos_main() was introduced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6185>
The `G_DECLARE_FINAL_TYPE` macro does not need to be terminated with a
semicolon and the extra semicolon breaks building e.g. libcamera with
clang because `-Wextra-semi` is used which produces the following
error in conjunction with `-Werror`:
```
gstreamer-1.0/gst/allocators/gstdrmdumb.h:61:43: error: extra ';' outside
of a function is incompatible with C++98 [-Werror,-Wc++98-compat-extra-semi]
61 | GST, DRM_DUMB_ALLOCATOR, GstAllocator);
| ^
1 error generated.
```
Fix this by removing the extra semicolon
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6239>
When dealing with demuxers which aren't streams-aware, we need to handle the
old-school "stream replacement" dance from `parsebin` and hide that in such a
way that output pads are re-used (if compatible).
By analyzing the collection posted by parsebin, we can:
* Identify whether some output slots are no longer used (because the stream they
currently handle is not present in the collection)
* Decide if some upcoming streams could re-use the existing slot
This supports both buffering and non-buffering modes.
Fixes#1651
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6201>
When switching urisourcebin, ensure that we first unlink *all* pads from
decodebin3 before linking them again.
This is to ensure that decodebin3 completely knows that all previous pads are no
longer needed and can prepare itself to being re-used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6179>
The value is stored as an 8 bit integer, with 0 meaning that there is
not data for this extension. That means that the maximum length is 255
bytes and not 256 bytes.
On the other hand, the one-byte RTP header extensions are storing the
length as a 4 bit integer with an offset of 1 (i.e. 0 means 1 byte
extension length), so here 16 is the correct maximum length.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6180>
Some subtitle "decoders" had a wrong category of "Parser", which `parsebin`
relies on to identify elements which do not *decode* streams but *parse* them.
This would cause such subtitle decoders to be plugged in within parsebin,
preventing the original stream to be properly used by (more efficient)
downstream decoders or subtitle renderers.
Fixes#1757
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153>
When allocating buffers with alignment parameters specified, it
may be necessary to overallocate memory to adjust to the requested
alignment. Previously the padding length was not included in the mmaped
buffer size, leaving unmapped bytes at the end of the buffer.
This caused intermittent SEGV faults and valgrind failures when running
the wayland_threads example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6104>
* Bump the rank of the musepack v7/v8 FFmpeg demuxers to SECONDARY
* Bump the rank of the musepack v7/v8 FFmpeg audio decoders to SECONDARY
* Demote the rank of the musepackdec element to MARGINAL
This is for two reasons:
* The musepack library is no longer maintained, whereas the FFmpeg
implementation can/will receive fixes
* The `musepackdec` implementation was a all-in-one "parsing and decoding" blob
which doesn't play nicely with decodebin3 and others
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3033
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6074>
It was adding and subtracting the segment base here and there, but it
was also doing so incorrectly, leading to various calculation errors.
Fixed a few bugs uncovered, related to getting a new segment:
* If we reset base_ts/next_ts/out_frame_count, also reset prevbuf
* Only do so if the new segment is different than the previous one
Also replaced a few occurrences of GST_BUFFER_TIMESTAMP with
GST_BUFFER_PTS for consistency.
Integrated the tests of
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2186
, now passing. The test_segment_update_same test had to be fixed,
because it was wrongly assuming that we would not fill the gap inside
the new-but-same segment.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6031>
If the allocation function get called from multiple threads at the same time,
multiple allocators may get created but only one get saved. Leading to other
allocators to be leaked. Simply create it once in the instance initialization.
Fixes: #2456
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6052>
- Support HTTP redirect codes (301,302,307,308) on response to GET.
"Location" field is extracted and used for following GET and POST.
- Notify caller a redirect took place using return value
- log source and destination url on redirect
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5222>
Converting from RGB to YUV: When comparing the info.colorimetry to
GST_VIDEO_COLORIMETRY_BT709 it does not make sense to look at the input
signal because that is of type of RGB. The plugin needs to look at the
output YUV-type and compare GST_VIDEO_COLORIMETRY_BT709 to that, because
that is the YUV-type the plugin needs to convert input-RGB into.
Converting from YUV to RGB: Comparing to the input is correct, but because
here the color encoding info BT601/BT709 is on input side of the plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5998>
In the following backtrace for the deadlock, we can see that:
- In T8 `uridecodebin3` is exposing a new pad, in `pad_added_cb`,
`playbin3` is trying to get `GST_PLAY_BIN3_LOCK` in the callback. This
threads holds its `SELECTION_LOCK` in F17 `reconfigure_output_stream`,
which is looks right `decodebin3` is handling its selection state
in that code path
- In T7 `playbin3` holds the `GST_PLAY_BIN3_LOCK` when calling
`gst_element_post_message` in `gst_play_bin3_send_event` which is
not necessary in that section of the code.
``` bt
Thread 8 (Thread 0x7f0b78ee36c0 (LWP 2952467) "multiqueue0:src"):
#0 futex_wait (private=0, expected=2, futex_word=0x1fa6d60) at ../sysdeps/nptl/futex-internal.h:146
#1 __GI___lll_lock_wait (futex=futex@entry=0x1fa6d60, private=0) at lowlevellock.c:49
#2 0x00007f0b858cd46a in lll_mutex_lock_optimized (mutex=0x1fa6d60) at pthread_mutex_lock.c:48
#3 ___pthread_mutex_lock (mutex=0x1fa6d60) at pthread_mutex_lock.c:128
#4 0x00007f0b7e665720 in pad_added_cb (uridecodebin=0x1fb4050, pad=0x7f0b54022060, playbin=0x1fb00e0) at ../subprojects/gst-plugins-base/gst/playback/gstplaybin3.c:2463
#5 0x00007f0b85c00060 in g_closure_invoke (closure=0x1fa9eb0, return_value=0x0, n_param_values=2, param_values=0x7f0b78ee1dd0, invocation_hint=0x7f0b78ee1d50) at ../gobject/gclosure.c:832
#6 0x00007f0b85c2cf66 in signal_emit_unlocked_R.isra.0 (node=node@entry=0x1c5bf30, detail=detail@entry=0, instance=instance@entry=0x1fb4050, emission_return=emission_return@entry=0x0, instance_and_params=instance_and_params@entry=0x7f0b78ee1dd0) at ../gobject/gsignal.c:3796
#7 0x00007f0b85c1d4da in g_signal_emit_valist (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>, var_args=var_args@entry=0x7f0b78ee1f90) at ../gobject/gsignal.c:3549
#8 0x00007f0b85c1d6f3 in g_signal_emit (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>) at ../gobject/gsignal.c:3606
#9 0x00007f0b85e20c3e in gst_element_add_pad (element=0x1fb4050, pad=0x7f0b54022060) at ../subprojects/gstreamer/gst/gstelement.c:802
#10 0x00007f0b7e632620 in add_output_pad (dec=0x1fb4050, target_pad=0x7f0b6400fda0) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:717
#11 0x00007f0b7e632788 in db_pad_added_cb (element=0x1fb8020, pad=0x7f0b6400fda0, dec=0x1fb4050) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:736
#12 0x00007f0b85c00060 in g_closure_invoke (closure=0x1fb7fc0, return_value=0x0, n_param_values=2, param_values=0x7f0b78ee2300, invocation_hint=0x7f0b78ee2280) at ../gobject/gclosure.c:832
#13 0x00007f0b85c2cf66 in signal_emit_unlocked_R.isra.0 (node=node@entry=0x1c5bf30, detail=detail@entry=0, instance=instance@entry=0x1fb8020, emission_return=emission_return@entry=0x0, instance_and_params=instance_and_params@entry=0x7f0b78ee2300) at ../gobject/gsignal.c:3796
#14 0x00007f0b85c1d4da in g_signal_emit_valist (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>, var_args=var_args@entry=0x7f0b78ee24c0) at ../gobject/gsignal.c:3549
#15 0x00007f0b85c1d6f3 in g_signal_emit (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>) at ../gobject/gsignal.c:3606
#16 0x00007f0b85e20c3e in gst_element_add_pad (element=0x1fb8020, pad=0x7f0b6400fda0) at ../subprojects/gstreamer/gst/gstelement.c:802
#17 0x00007f0b7e6260b4 in reconfigure_output_stream (output=0x7f0b5400def0, slot=0x7f0b64013dd0, msg=0x7f0b78ee26b8) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:3086
#18 0x00007f0b7e623700 in check_slot_reconfiguration (dbin=0x1fb8020, slot=0x7f0b64013dd0) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:2455
#19 0x00007f0b7e623e62 in multiqueue_src_probe (pad=0x7f0b6001e600, info=0x7f0b78ee2930, slot=0x7f0b64013dd0) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:2544
#20 0x00007f0b85e53aaa in probe_hook_marshal (hook=0x7f0b74040500, data=0x7f0b78ee28c0) at ../subprojects/gstreamer/gst/gstpad.c:3669
#21 0x00007f0b85c88a3e in g_hook_list_marshal (hook_list=0x7f0b6001e698, may_recurse=1, marshaller=0x7f0b85e53786 <probe_hook_marshal>, data=0x7f0b78ee28c0) at ../glib/ghook.c:674
#22 0x00007f0b85e541be in do_probe_callbacks (pad=0x7f0b6001e600, info=0x7f0b78ee2930, defaultval=GST_FLOW_OK) at ../subprojects/gstreamer/gst/gstpad.c:3853
#23 0x00007f0b85e5ac9c in gst_pad_push_event_unchecked (pad=0x7f0b6001e600, event=0x7f0b64002120, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5538
#24 0x00007f0b85e54bc2 in push_sticky (pad=0x7f0b6001e600, ev=0x7f0b78ee2a60, user_data=0x7f0b78ee2ac0) at ../subprojects/gstreamer/gst/gstpad.c:4057
#25 0x00007f0b85e4a13c in events_foreach (pad=0x7f0b6001e600, func=0x7f0b85e54a8e <push_sticky>, user_data=0x7f0b78ee2ac0) at ../subprojects/gstreamer/gst/gstpad.c:613
#26 0x00007f0b85e54f91 in check_sticky (pad=0x7f0b6001e600, event=0x7f0b64002120) at ../subprojects/gstreamer/gst/gstpad.c:4116
#27 0x00007f0b85e5b65e in gst_pad_push_event (pad=0x7f0b6001e600, event=0x7f0b64002120) at ../subprojects/gstreamer/gst/gstpad.c:5706
#28 0x00007f0b7e5888e7 in gst_single_queue_push_one (mq=0x1fbb000, sq=0x7f0b64013b70, object=0x7f0b64002120, allow_drop=0x7f0b78ee2c3c) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2068
#29 0x00007f0b7e58a1bc in gst_multi_queue_loop (pad=0x7f0b6001e600) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2347
#30 0x00007f0b85e96a36 in gst_task_func (task=0x7f0b64016050) at ../subprojects/gstreamer/gst/gsttask.c:399
#31 0x00007f0b85e97e41 in default_func (tdata=0x7f0b640138d0, pool=0x1fbe9c0) at ../subprojects/gstreamer/gst/gsttaskpool.c:70
#32 0x00007f0b85cd1ab2 in g_thread_pool_thread_proxy (data=<optimized out>) at ../glib/gthreadpool.c:352
#33 0x00007f0b85ccc982 in g_thread_proxy (data=0x7f0b6006f640) at ../glib/gthread.c:831
#34 0x00007f0b858ca12d in start_thread (arg=<optimized out>) at pthread_create.c:442
#35 0x00007f0b8594bbc0 in clone3 () at ../sysdeps/unix/sysv/linux/x86_64/clone3.S:81
Thread 7 (Thread 0x7f0b7a7646c0 (LWP 2952434) "multiqueue3:src"):
#0 syscall () at ../sysdeps/unix/sysv/linux/x86_64/syscall.S:38
#1 0x00007f0b85cf470c in g_mutex_lock_slowpath (mutex=0x1fb81d0) at ../glib/gthread-posix.c:1494
#2 0x00007f0b7e6281a2 in gst_decodebin3_send_event (element=0x1fb8020, event=0x7f0b6800a650) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:3561
#3 0x00007f0b85e23ca3 in gst_element_send_event (element=0x1fb8020, event=0x7f0b6800a650) at ../subprojects/gstreamer/gst/gstelement.c:1994
#4 0x00007f0b7e63806b in gst_uri_decodebin3_send_event (element=0x1fb4050, event=0x7f0b6800a650) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:2227
#5 0x00007f0b85e23ca3 in gst_element_send_event (element=0x1fb4050, event=0x7f0b6800a650) at ../subprojects/gstreamer/gst/gstelement.c:1994
#6 0x00007f0b7e66375a in gst_play_bin3_send_event (element=0x1fb00e0, event=0x7f0b6800a650) at ../subprojects/gst-plugins-base/gst/playback/gstplaybin3.c:1863
#7 0x00007f0b85e23ca3 in gst_element_send_event (element=0x1fb00e0, event=0x7f0b6800a650) at ../subprojects/gstreamer/gst/gstelement.c:1994
#8 0x00007f0b85f61b5b in stream_selection_cb (bus=0x1dc2d80, message=0x7f0b68008b00, d=0x1d7de30) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2235
#9 0x00007f0b85c00060 in g_closure_invoke (closure=0x1d2a5b0, return_value=0x0, n_param_values=2, param_values=0x7f0b7a7627b0, invocation_hint=0x7f0b7a762730) at ../gobject/gclosure.c:832
#10 0x00007f0b85c2cf66 in signal_emit_unlocked_R.isra.0 (node=node@entry=0x1c5e5f0, detail=detail@entry=235, instance=instance@entry=0x1dc2d80, emission_return=emission_return@entry=0x0, instance_and_params=instance_and_params@entry=0x7f0b7a7627b0) at ../gobject/gsignal.c:3796
#11 0x00007f0b85c1d4da in g_signal_emit_valist (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>, var_args=var_args@entry=0x7f0b7a762970) at ../gobject/gsignal.c:3549
#12 0x00007f0b85c1d6f3 in g_signal_emit (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>) at ../gobject/gsignal.c:3606
#13 0x00007f0b85e05be9 in gst_bus_sync_signal_handler (bus=0x1dc2d80, message=0x7f0b68008b00, data=0x0) at ../subprojects/gstreamer/gst/gstbus.c:1307
#14 0x00007f0b85e03834 in gst_bus_post (bus=0x1dc2d80, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbus.c:364
#15 0x00007f0b85e2436c in gst_element_post_message_default (element=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2127
#16 0x00007f0b85df42b1 in gst_bin_post_message (element=0x1fb00e0, msg=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:2789
#17 0x00007f0b85e24627 in gst_element_post_message (element=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2170
#18 0x00007f0b85df7c12 in gst_bin_handle_message_func (bin=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:4041
#19 0x00007f0b85e61bd3 in gst_pipeline_handle_message (bin=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstpipeline.c:669
#20 0x00007f0b7e663fa4 in gst_play_bin3_handle_message (bin=0x1fb00e0, msg=0x7f0b68008b00) at ../subprojects/gst-plugins-base/gst/playback/gstplaybin3.c:2030
#21 0x00007f0b85df5a98 in bin_bus_handler (bus=0x1dc2cc0, message=0x7f0b68008b00, bin=0x1fb00e0) at ../subprojects/gstreamer/gst/gstbin.c:3263
#22 0x00007f0b85e037f1 in gst_bus_post (bus=0x1dc2cc0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbus.c:357
#23 0x00007f0b85e2436c in gst_element_post_message_default (element=0x1fb4050, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2127
#24 0x00007f0b85df42b1 in gst_bin_post_message (element=0x1fb4050, msg=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:2789
#25 0x00007f0b85e24627 in gst_element_post_message (element=0x1fb4050, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2170
#26 0x00007f0b85df7c12 in gst_bin_handle_message_func (bin=0x1fb4050, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:4041
#27 0x00007f0b7e638005 in gst_uri_decode_bin3_handle_message (bin=0x1fb4050, msg=0x7f0b68008b00) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:2218
#28 0x00007f0b85df5a98 in bin_bus_handler (bus=0x1dc2e40, message=0x7f0b68008b00, bin=0x1fb4050) at ../subprojects/gstreamer/gst/gstbin.c:3263
#29 0x00007f0b85e037f1 in gst_bus_post (bus=0x1dc2e40, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbus.c:357
#30 0x00007f0b85e2436c in gst_element_post_message_default (element=0x1fb8020, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2127
#31 0x00007f0b85df42b1 in gst_bin_post_message (element=0x1fb8020, msg=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:2789
#32 0x00007f0b85e24627 in gst_element_post_message (element=0x1fb8020, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2170
#33 0x00007f0b7e61ee43 in sink_event_function (sinkpad=0x7f0b6400f8c0, dbin=0x1fb8020, event=0x7f0b6c097870) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:1450
#34 0x00007f0b85f51122 in gst_validate_pad_monitor_downstream_event_check (pad_monitor=0x7f0b6c094a80, parent=0x1fb8020, event=0x7f0b6c097870, handler=0x7f0b7e61e797 <sink_event_function>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-pad-monitor.c:2101
#35 0x00007f0b85f535bf in gst_validate_pad_monitor_sink_event_full_func (pad=0x7f0b6400f8c0, parent=0x1fb8020, event=0x7f0b6c097870) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-pad-monitor.c:2406
#36 0x00007f0b85f537fa in gst_validate_pad_monitor_sink_event_func (pad=0x7f0b6400f8c0, parent=0x1fb8020, event=0x7f0b6c097870) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-pad-monitor.c:2418
#37 0x00007f0b85e5c523 in gst_pad_send_event_unchecked (pad=0x7f0b6400f8c0, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5940
#38 0x00007f0b85e5ae65 in gst_pad_push_event_unchecked (pad=0x7f0b6400f650, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5573
#39 0x00007f0b85e54bc2 in push_sticky (pad=0x7f0b6400f650, ev=0x7f0b7a763620, user_data=0x7f0b7a763680) at ../subprojects/gstreamer/gst/gstpad.c:4057
#40 0x00007f0b85e4a13c in events_foreach (pad=0x7f0b6400f650, func=0x7f0b85e54a8e <push_sticky>, user_data=0x7f0b7a763680) at ../subprojects/gstreamer/gst/gstpad.c:613
#41 0x00007f0b85e54f91 in check_sticky (pad=0x7f0b6400f650, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:4116
#42 0x00007f0b85e5b65e in gst_pad_push_event (pad=0x7f0b6400f650, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:5706
#43 0x00007f0b85e523e7 in event_forward_func (pad=0x7f0b6400f650, data=0x7f0b7a763820) at ../subprojects/gstreamer/gst/gstpad.c:3130
#44 0x00007f0b85e521e3 in gst_pad_forward (pad=0x7f0b6004f180, forward=0x7f0b85e522bd <event_forward_func>, user_data=0x7f0b7a763820) at ../subprojects/gstreamer/gst/gstpad.c:3084
#45 0x00007f0b85e525ab in gst_pad_event_default (pad=0x7f0b6004f180, parent=0x7f0b6400f650, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:3181
#46 0x00007f0b85e5c523 in gst_pad_send_event_unchecked (pad=0x7f0b6004f180, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5940
#47 0x00007f0b85e5ae65 in gst_pad_push_event_unchecked (pad=0x7f0b64008360, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5573
#48 0x00007f0b85e54bc2 in push_sticky (pad=0x7f0b64008360, ev=0x7f0b7a763a60, user_data=0x7f0b7a763ac0) at ../subprojects/gstreamer/gst/gstpad.c:4057
#49 0x00007f0b85e4a13c in events_foreach (pad=0x7f0b64008360, func=0x7f0b85e54a8e <push_sticky>, user_data=0x7f0b7a763ac0) at ../subprojects/gstreamer/gst/gstpad.c:613
#50 0x00007f0b85e54f91 in check_sticky (pad=0x7f0b64008360, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:4116
#51 0x00007f0b85e5b65e in gst_pad_push_event (pad=0x7f0b64008360, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:5706
#52 0x00007f0b7e5888e7 in gst_single_queue_push_one (mq=0x7f0b60076540, sq=0x7f0b6c093300, object=0x7f0b6c097870, allow_drop=0x7f0b7a763c3c) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2068
#53 0x00007f0b7e58a1bc in gst_multi_queue_loop (pad=0x7f0b64008360) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2347
#54 0x00007f0b85e96a36 in gst_task_func (task=0x7f0b6c072050) at ../subprojects/gstreamer/gst/gsttask.c:399
#55 0x00007f0b85e97e41 in default_func (tdata=0x7f0b6c093ef0, pool=0x1fbe9c0) at ../subprojects/gstreamer/gst/gsttaskpool.c:70
#56 0x00007f0b85cd1ab2 in g_thread_pool_thread_proxy (data=<optimized out>) at ../glib/gthreadpool.c:352
#57 0x00007f0b85ccc982 in g_thread_proxy (data=0x7f0b70033800) at ../glib/gthread.c:831
#58 0x00007f0b858ca12d in start_thread (arg=<optimized out>) at pthread_create.c:442
#59 0x00007f0b8594bbc0 in clone3 () at ../sysdeps/unix/sysv/linux/x86_64/clone3.S:81
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5982>
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>
If GST_PAD_SINK is passed in this means that we're supposed to convert
from sink caps to src caps, not the other way around. In other words, if
GST_PAD_SINK is passed we're supposed to produce the possible output
caps.
Previously this was inverted. This had the effect that glcolorconvert
pretended to be able to convert *to* I420 without glDrawBuffers, which is
not possible, and pretended not to be able to convert *from* I420
without glDrawBuffers, which it always supports.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5941>
