Before trying to retrieve a GMainContext from a provided
GstPlayerSignalDispatcher, check that it is actually
GstPlayerGMainContextSignalDispatcher. If not, use the
default GMainContext for dispatching signals via the adapter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7398>
Since bins can set the context of their children elements, the set_context()
vmethod shouldn't call bus messages post methods, since it locks the parent
object, the bin, which might be already locked, leading to a deadlock.
Fixes: #3706
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7379>
This patch addresses the issue where GStreamer would throw an error when
attempting to use bt2100-hlg colorimetry with V4L2, which is not
supported by the current V4L2 kernel. When bt2100-hlg colorimetry is set
from caps, the check for transfer (GST_VIDEO_TRANSFER_ARIB_STD_B67) is
bypassed.
The main improvement is to avoid checking the transfer value in
gst_v4l2_video_colorimetry_matches when it is
GST_VIDEO_TRANSFER_ARIB_STD_B67. This is because the transfer value in
the cinfo parameter comes from gst_v4l2_object_get_colorspace, which
converts the transfer to another value, causing a mismatch.
Since the kernel does not support GST_VIDEO_TRANSFER_ARIB_STD_B67,
gst_v4l2_object_get_colorspace cannot map it correctly from V4L2 to
GStreamer. Therefore, we ignore this check to prevent errors.
changes:
- Added a condition in gst_v4l2_video_colorimetry_matches to bypass the
transfer check when the transfer is GST_VIDEO_TRANSFER_ARIB_STD_B67.
- Ensured that the pipeline does not throw errors due to unsupported
bt2100-hlg colorimetry in V4L2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7369>
fix playback fail, when some file with length_size_minus_one == 2
According to the spec 2 cannot be a valid value, so that stream has a
bad config record. but breaking the decoding because of that, perhaps is too much.
and ffmpeg seem not check this
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7358>
With GLES 2.0 we are forced to use CopyTextImage2D which requires
passing an internal format. With QT6 eglfs, we need to pass GL_RGB
instead, probably because of how the texture has been created. As its
hard to guess, simply fallback to GL_RGB on failure. This fixes usage
or qml6glsrc with eglfs backend, without loosing support for
semi-transparent window on other platforms.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7345>
In case when conn->input_stream is NULL and glib was built with
"glib_checks" enabled, g_pollable_input_stream_read_nonblocking()
returns -1, but does not set the "err".
The call stack:
read_bytes() ->
fill_bytes() ->
fill_raw_bytes()
The return value -1 passed up to read_bytes() and incorrectly
processed there after "error:" label.
This changes the return value to EINVAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7309>
According to the ffmpeg documentation[1] the read_packet function should never
return 0. ffmpegdata_peek returns 0 when the stream is EOF causing us to fail
detecting EOF and never close the pipeline, continually spinning on more data.
ffmpeg instead wants an AVERROR_EOF code for to signal EOF.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7299>
A previous fix, a275e1e029, is correct but was too
permissive since it treats all un-matched NAL units the same as AU delimiters
even though some other NAL unit types can be encountered in the processing loop.
The problem this can cause is that some hardware decoders experience bad
performance when handling FD units that precede the SPS.
This change restores the original behavior for FDs so that they're ignored until
the SPS is received and it preserves the codec conformance test gains that the
fix has achieved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7296>
glCheckFrameStatus() can fail by returning 0, and otherwise return a
status. Fix the trace to make it clear when we get an unkown status
compare to having an error, in which case we also trace the error code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7295>
video-info supports encoded format to have RGB color-matrix, while
v4l2object just leave the v4l2 matrix to default when mapping
GST_VIDEO_COLOR_MATRIX_RGB. It causes gst matrix changed to be
GST_VIDEO_COLOR_MATRIX_BT601 when mapping v4l2 colorimetry.
So add support for encoded format with RGB color-matrix in v4l2object.
Note that for M2M encoders, we should in theory assume that that we can
transfer this value from OUTPUT to CAPTURE queues, though its only true
if the drivers does not do CSC. For now, we don't support any RGB
codecs, but leaving a note for the future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7284>
The V4L2_MAP_QUANTIZATION macro has been fixed to something a lot saner,
fix our replica accordingly. The new macro now simply set the quantization
to full range is the pixel formats is RGB based, or if the JPEG
colorspace is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7284>
Fixing warnings
GStreamer-CRITICAL **: 01:21:25.862: gst_value_set_int_range_step:
assertion 'start < end' failed
Although when QSV runtime reports a codec is supported, resolution query
fails sometimes, espeically VP9 encoder case on Windows.
Don't try to register an element if resolution query returned an error
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7252>
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/7235>
VLC counts METADATA as 1 even if the specification states you must not.
This leads to asfdemux failing since there are no bytes left when asfdemux
tries to extract the "last" header.
Do not fail hard in this case and try to proceed when everything else went
fine.
So at least gst-discoverer will see what's in the file.
Closes#3684
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7216>
None of the symbols in webrtc-audio-coding-1 are marked with
`__declspec(dllexport)`, rendering the library usable only if
it was built with GCC/Clang.
The only fix available (as the pulseaudio copy has not been updated
with Google's upstream) is to ensure the fallback builds statically.
Although this change will also affect webrtcdsp's dependency on
webrtc-audio-processing-1, it does not break its compilation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7206>
Doing so resets the stride from the VideoMeta and it wasn't done before
the commit below. While on it, drop the plane size check as we can't
reliably predict the correct size when using DRM modifiers.
Fixes: 89b0a6fa23 ("va: refactor buffer import")
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7196>
The driver - AKA intel-vaapi-driver - has been unmaintained for four years
now and encoding appears to be broken in various cases. As it's unlikely
that the situation will improve, blocklist the driver for encoding.
Decoding appears to be stable enough to keep it enabled.
The driver can still be used by setting the `GST_VA_ALL_DRIVERS` env
variable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7186>
GstFFMpegAudDec.context can be nullptr if decoder got closed
without opening new context. Note that we don't need to clear
AVCodecContext.extradata there since avcodec_free_context()
will do clear the data if needed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7184>
avcodec_close() is deprecated and it's not supported anymore to re-open
a codec, so we only ever allocate the codec in set_format() now and
always free it after usage.
As part of this, also fix various memory leaks in related code paths.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7148>
* gst_ffmpegviddec_frame() is the only caller of gst_ffmpegviddec_video_frame()
and has the same signature. Just move the checks into a single function and
use that.
* Make it clear which frames are the input and output ones in
gst_ffmpegviddec_video_frame() to make issues like the one fixed in the previous
commit more obvious.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7148>
Direct access to AVInputFormat::read_probe() is not possible anymore
with ffmpeg 7.0, and the usefulness of this typefinder seems limited
anyway. An alternative implementation around av_probe_input_format3() or
similar would be possible but it would be going over all possible ffmpeg
probes at once.
Having a typefinder here means that basically every application will
load the gst-libav plugin when typefinding is necessary, which has
unnecessary performance impacts. If a typefinder from here was indeed
missing from typefindfunctions in gst-plugins-base then it would be
better to add it there directly.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3378
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7148>
${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/7143>
since the encoded output is changing based on version
it does not make sense to check the output bitstream with a fixed
bytearray since the version in the target might vary. So sticking
to checking the number of output buffers and encoded frame size
similar to the other tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7142>
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>
