Some NLE tests were calling ges_init() from the test suite
initialization function. This was causing them to deadlock when running
with glib 2.76.3.
ges_init() causes a GThreadPool to be initialized. Even if GES at this
point doesn't request any thread to be created, since glib 2.63.4+
(see 8aeca4fa64)
the first time a GThreadPool is initialized a "pool-spawner" thread is
created, which is later used by g_thread_pool_push().
The default behavior of the GStreamer check tests is to fork for every
test case. This is not safe if any thread has been created at this
point. In this particular case, GThreadPool preserves the state that
says a "pool-spawner" thread has been created, and will have access to
its mutex and condition variable, but their queues will have different
contents as the memory has been forked. In consequence, calls to
g_thread_pool_push() will deadlock.
The deadlock will not occur if running the tests with CK_FORK=no.
This patch modifies the affected tests to only call ges_init() from
inside the test cases, fixing the deadlock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4915>
A significant portion of the NLE test suite was often timing out due to
the tests taking way longer than necessary because the sinks were
synchronizing to the clock, which is the default behavior for
fakevideosink and fakeaudiosink.
Notable was the case of nleoperation.c:test_pyramid_operations, that ran
through a 10 second stream twice. As the default timeout is 20 seconds,
this made the test flaky.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4914>
Refusing an incoming segment in < GST_MATROSKA_READ_STATE_DATA should only be
done if the incoming segment is not in GST_FORMAT_TIME.
In GST_FORMAT_TIME, we are just storing the values and returning, so we can
invert the order of the checks.
Fixes proper segment propagation in matroska/webm DASH use-cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
Is a seek is done on stream-collection post, there are no selected streams
yet. Therefore none would be chosen to adjust the key-unit seek.
If no streams are selected, fallback to a default stream (i.e. one which has
track(s) with GST_STREAM_FLAG_SELECT).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
When seeking is handled by the collection posting thread, there is a possibility
that some leftover data will be pushed by the stream thread.
Properly detect and reject those early segments (and buffers) by comparing it to
the main segment seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
As VK_FORMAT_FEATURE_2_xxx are defined as static const variable, the
vscoce C compiler prevents the initialization of the vk_usage_map
structure with error "C2099: initializer is not a constant".
Init the structure separately.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4904>
The current way of using parent's copy_metadata() virtual function will
selectively filter out some meta such as crop meta. That virtual function
should be used when copying input buffer's meta data into output buffer,
not suitable when importing the input buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
When the input buffer has crop meta, and we need to do copy, we
should consider the uncropped video size and copy the full size
of video memory.
The video meta in this case should contain the full uncropped
resolution info. We can use it to create full size va buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
Adding Direct3D11 backend Qt6 QML videosink element, qml6d3d11sink.
Implementation details are similar to the qt6 plugin in -good
but there are a few notable differences.
* qml6d3d11sink accepts all GstD3D11 supported video formats (e.g., NV12).
* Scene graph (owned by qml6d3d11sink) will hold dedicated and sharable
RGBA texture which belongs to Qt6's Direct3D11 device, instead of sharing
GStreamer's own texture with Qt6.
* All rendering operations will be done by using GStreamer's Direct3D11 device.
Specifically, upstream texture will be copied (in case of RGBA)
or converted to the above mentioned Qt6's sharable texture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3707>
... otherwise streams with constant size samples defined with a single
`sample_size` for all samples in the `stsz` box fall in the category
`chunks_are_samples` in `qtdemux_stbl_init`, overriding the actual
sample count.
`FOURCC_soun` would set this automatically for `compression_id == 0xfffe`,
however `compression_id` is read from the Audio Sample Entry box at an offset
marked as "pre-defined" in some version of the spec and set to 0 both by
GStreamer and FFmpeg for opus streams.
Considering the stream `sampled` flag is set explicitely by other fourcc
variants, doing so for opus seems consistent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4903>
This will cause an integer overflow a little bit further down because we
allocate a bit more memory to allow for a NUL-terminator.
The caller should've avoided passing that much data in already as it's
not going to be a valid image and there's likely not even that much data
available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4894>
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>
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.
In `build_opus_extension`, `gst_byte_writer_put*_le ()` variants were used,
causing audio streams conversion to Opus in mp4 to offset samples due to the
PreSkip field incorrect value (29ms early in our test cases).
[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875>
Now all codec baseclasses can inform subclasses of correct max DPB size,
and exception handling (e.g., emergency bumping in h.264) has been
improved as well. Smaller number of additional DPB frame allocation
seems to be safe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4878>
Adding GST_CUDA_CRITICAL_ERRORS env variable so that program can be
terminated on unrecoverable error.
Example)
GST_CUDA_CRITICAL_ERRORS=2,700 gst-launch-1.0 ...
In this example, CUDA_ERROR_OUT_OF_MEMORY(2) and
CUDA_ERROR_ILLEGAL_ADDRESS(700) are registered as critical error
and program will be aborted on those errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4729>
If two senders use the same multicast IP and port then new_session_pad()
may try to add a srcpad to the same stream twice.
stream->srcpad is updated but gst_element_add_pad() fails the second
time. As a result stream->srcpad points to a deleted object and
access in gst_sdp_demux_stream_free() fails with a segfault.
Just ignore the second pad. Nothing useful can be done with it anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4603>
- Adding bayer 10,12,14,16 bits components with 16 bits storage. These
changes only adds capabilities. Capability format string is a complete
description of the frame and pixels layout. Only mapping LE bayer
formats as v4l2 only define LE bayer formats.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4852>
The `gst_video_decoder_negotiate_pool` function expects the
`decide_allocation` function to always provide a pool and will fail to
negotiate if the pool is missing. If we return immediately (even if we
don't need to do anything special) negotiation will fail if the
downstream element does not propose a pool.
Fix by chaining up to the default `decide_allocation` function which
adds a fallback pool if one was not already proposed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4630>
Adding DirectWrite text rendering elements
* dwriteclockoverlay: Equivalent to clockoverlay
* dwritetimeoverlay: Equivalent to timeoverlay
* dwritetextoverlay: Similar to textoverlay but subtitle is not
supported
Newly added elements support system memory and d3d11 memory
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4826>
This new property allows setting of PES stream number for AAC audio
and AVC video streams.
The stream number is subject to the following constraints:
1. it must be between 0 and 15 for video
2. it must be between 0 and 31 for audio
Currently the PES stream number is hard-coded to zero for these
stream types.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4822>
Add support for 10/12/14/16 bit depths . This consists of multiple parts.
First is the parsing of caps, which pulls out the bitness and endianness
from the video/x-bayer format.
Second, gst_bayer2rgb_split_and_upsample_horiz() is split into two similar
functions, one for 8bit bayer handling and another for 16bit bayer handling.
The content is basically identical, except one uses 8bpp and the other 16bpp
inputs and outputs, and they each use different ORC code to match. The 16bpp
variant also handles endian swapping. There is now a wrapper called
gst_bayer2rgb_split_and_upsample_horiz() which selects the correct function
based on bpp from the parser.
Third, gst_bayer2rgb_process() is extended to handle both 8bit and 16bit
bayer data. Yet again there are matching ORC functions to handle the 16bit
data. This time however the 16bit handling of data is slightly special. The
ORC is not able to emit opcodes for 'x2 mergelq', so the trick here is to
store the BG and GR longs into separate 'dtmp' temporary buffer, and then
do one more ORC post-processing step, compensate for the less-than-16bpp
bitness using left shift, and reorder them into the destination frame
using 'mergelq' .
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr16le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add comments regarding which LINE()s point to which data in the
temporary buffer and a large comment explaining how the buffer
is processed. This will hopefully be useful to someone, as the
code is not obvious. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Instead of passing a single element of GstBayer2RGB structure into the
gst_bayer2rgb_split_and_upsample_horiz(), pass the entire pointer and
let the funciton pick out whatever it needs out of the structure. This
is a preparatory patch. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Pass all three parameters used by the LINE() macro to the LINE() macro
and unroll the code for readability. Add more comments regarding which
of these LINE()s point to which data in the temporary buffer to make
the code less confusing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The j variable is used as an iterator further down in this code, but
here it can be just inlined in the macro parameters to make the code
easier to read. This is done in preparation for further changes. No
functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The bayer2rgb process implemented doesn't support in-place tranform.
This element doesn't implement a "transform_ip" vmethod of
GstBaseTransform it will revert to using the "tranform" vmethod.
It's misleading to set it to TRUE, here. Change this to FALSE.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add support for conversion to 10/12/14/16 bit bayer 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 num-buffers=1 ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
filesink location=/tmp/bayer12.raw
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
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>
Even if we don't yet know what the echo probe format is, we want to be able to
provide silence for the reverse path, so that when the probe becomes available,
there is no ambiguity around what time period the new set of samples are for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
The probe's info may not precisely match the dsp's info. For instance,
the number of channels or their layout might be different.
```
GStreamer-Audio-CRITICAL **: 16:21:32.899: the GstAudioInfo argument is not equal to the GstAudioMeta's attached info
```
This broke in d5755744c3.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
Race condition without this patch:
- srcpad task is being stopped in gst_aggregator_stop_srcpad_task()
- at that moment, in pre-queue event handler, gst_pad_get_task_state()
returned GST_TASK_PAUSED
- then in srcpad task got stopped in gst_aggregator_stop_srcpad_task()
- finally srcpad task got resumed in pre-queue event handler
To address it, checks "running" flag in pre-queue event handler.
Both pre-queue stream-start event handler and "running" flag
are protected by SRC_LOCK already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4847>
A race condition can occur in `srtpdec` during the READY -> NULL transition:
an RTCP buffer can make its way to `gst_srtp_dec_chain` while the element is
partially stopped, resulting in the following critical warning:
> Got data flow before segment event
The problematic sequence is the following:
1. An RTCP buffer is being handled by the chain function for the
`rtcp_sinkpad`. Since, this is the first buffer, we try pushing the sticky
events to `rtcp_srcpad`.
2. At the same moment, the element is being transitioned from PAUSED to READY.
3. While checking and pushing the sticky events for `rtcp_srcpad`, we reach the
Segment event. For this, we try to get it from the "otherpad", in this case
`rtp_srcpad`. In the problematic case, `rtp_srcpad` has already been
deactivated so its sticky events have been cleared. We won't be pushing any
Segment event to `rtcp_srcpad`.
4. We return to the chain function for `rtcp_sinkpad` and try pushing the
buffer to `rtcp_srcpad` for which deactivation hasn't started yet, hence the
"Got data flow before segment event".
This commit:
- Adds a boolean return value to `gst_srtp_dec_push_early_events`: in case the
Segment event can't be retrieved, `gst_srtp_dec_chain` can return an error
instead of calling `gst_pad_push`.
- Replaces the obsolete `gst_pad_set_caps` with `gst_pad_push_event`. The
additional preconditions checked by previous function are guaranteed here
since we push a fixed Caps which was built in the same function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4844>
A race condition can occur in `srtpdec` during the READY -> NULL transition:
an RTCP buffer can make its way to `gst_srtp_dec_chain` while the element is
partially stopped, resulting in the following critical warning:
> assertion 'parent->numsinkpads <= 1' failed
This can occur when the first RTCP buffer is received during the READY -> NULL
transition. If deactivation of the `rtp_srcpad` has already reached
`post_activate`, the sticky events are removed from this Pad. In this case,
`gst_srtp_dec_push_early_events` branches to the generation of a stream id
using `gst_pad_create_stream_id`. This function ensures that the element
doesn't own more than 1 sink pad. Since `srtpdec` owns two of them, the
assertion fails.
This commit uses `gst_element_decorate_stream_id` which doesn't perform this
check. The preconditions is not necessary in this particular context since the
stream id for the RTP / RTCP pads are derived from the same id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4844>
Change the internal GstVideoInfo structure in the GstVaDmabufAllocator to
GstVideoInfoDmaDrm in order to keep track of the exported DRM format by the
driver, and thus removing the DRMModifier quark attached as qdata in the
GstMemory. Though, the exposed API isn't updated yet; that has to go in a
second iteration.
Also this patch clean up some code (remove an unused buffer size assignation)
and fix some typos in documentation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4821>
The VA has its internal video format mapping(because different drivers may
have different interpretation for the same format), so we should convert the
info in GstVideoInfoDmaDrm into the according video info based on that mapping.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4821>
Some surface formats such as GST_VIDEO_FORMAT_Y42B and GST_VIDEO_FORMAT_RGB
can be created but can not be exported as DMA buffer. You can not say that
this is a driver bug because the driver may never want to share this kind of
surface out of libva.
And this function will be used to detect modifiers later, so the error message
will be annoying.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4821>
The proxy callback for the notify::last-message was emiting the signal
again on the child, which caused an infinit loop. We could swap the child
and the user data to signal to the bin instead, but it was found that proxying
this signal was not very useful. Typical use case it to set silent=0 and use
deep-notify feature. Proxying that signal just duplicate that output which
isn't very useful.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4766>
If the time server is restarted with a time in the past the net client
clock will not report the new time anymore as this would mean that the
clock moves back in time which it does not do.
Now the clock will be kept alive but marked as corrupted and will not
be re-used from the cache.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4802>
When doing a segment seek, the base offset in the new segment
would be increased by segment.position which is basically the
timestamp of the last packet. This does not include the duration
of the last packet though, so might be slightly shorter than the
actual duration of the clip or the requested segment.
Increase the base offset by the segment duration instead when
accumulating segments, which is more correct as it doesn't cut
off the last frame and makes the effective loop segment duration
consistent with the actual duration returned from a duration
query.
In case a segment stop was specified it's also possible that
some data was sent beyond the stop that's necessary for decoding
so the base offset increment should be based on that then and
not on the timestamp of the last buffer pushed out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4604>
glfilter will unref input buffer after _transform() call immidiately,
but gpu may still reading input buffer for rendering because gl
api is executed async. Need hold reference for input buffer by
adding parent meta to output buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4801>
This adds code to detect when the hex form of the string we are to
parse exceeds the number of bytes that would form a 32bit flag. This will
avoid treating as flagset anything above then the expected 32 bits and also
stop treading DRM format with modifiers as flagset (like
drm-format=AB24:0x0100000000000002).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4775>
Appsink will unref prev sample in dispose function. Which is later
when V4L2 video decoder link with appsink as V4L2 video decoder
will close V4L2 device fd during GST_STATE_CHANGE_READY_TO_NULL.
If the video buffer return to V4L2 video decoder after the decoder
closed V4L2 device fd, V4L2 can't release the video frame buffer
which allocated with MMAP mode as application can't call
VIDIOC_REQBUFS 0 to release the video frame buffer by V4L2 driver.
The memory of the video frame will leak.
Unref the gstbuffer in stop() function, so V4L2 video decoder
can received all video frame buffers and release it before close
V4L2 device fd.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4818>
Decoder bounded CUDA memory is allocated by driver and the pool size
is fixed. Since we don't know how many buffers would be held by
downstream non-CUDA element, we should download such CUDA memory
and release it back to decoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4810>
Otherwise it only works if GStreamer is binding the first socket on this
port.
Unfortunately this requires duplicating a bit more of Rust std because
`UdpSocket` can only be created already bound without allowing to set
any options between socket creation and binding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4807>
The muxer used a fixed value of 2 channels because the TR 102 366 spec
says they're to be ignored. However, the demuxer still trusted them,
resulting in bad caps.
Make the muxer fill in the correct channel count anyway (FFmpeg already
does) and make the demuxer ignore the value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4739>
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>
Configuration of our debugging system is possible before init, and in
fact is necessary too, otherwise the settings won't apply to logging
that happens during init.
For instance, since you cannot register a log function before you call
init in python, there is no way for you to log errors during init to
whatever logging service your app uses.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4753>
Generating the source element is done when urisourcebin is doing the READY to
PAUSED state change, so it is reasonable to set the new source element to that
state.
This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).
Finally it makes more sense to have an element in READY when attempting to query
information from it (such as SCHEDULING queries or probing live-ness).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3856>
Due to the alpha value being inserted with _BEFORE, we were ending up
with ARGB instead of RGBA, thus displaying completely wrong colours.
According to libpng's manual, "to add an opaque alpha channel, use filler=0xff
or 0xffff and PNG_FILLER_AFTER which will generate RGBA pixels".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4756>
Currently the uvcsink is only capable to run in an application
that is handling the state transitions of the pipeline properly
by checking on streaming event from the uvcsink.
This code is improving the element by adding an fakesink to
consume possible videostream flow in case the pipeline state
is not changing on hosts streamoff.
This is helpfull when using local gst-launch pipelines where
the streaming event is not monitored to change the pipelines
state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1304>
This patch adds an element to stream video data to an uvc video gadget.
The element handles the uvc events STREAMON, STREAMOFF, SETUP and DATA.
to start, stop and configure the video buffer flow by the use of pad
probes. It works with linux kernels of versions higher than v6.1.
The element makes use of the v4l2sink proxy property v4l2sink::device
to locate the corresponding device to parse the configfs for additional
data.
The code in uvc.c is basically derived from /lib/uvc.c in
https://git.ideasonboard.org/uvc-gadget.git.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1304>
Updated API usage appropriately, and now we have a versioned package to
track breaking vs. non-breaking updates.
Deprecates a number of properties (and we have to plug in our own values
for related enums which are now gone):
* echo-suprression-level
* experimental-agc
* extended-filter
* delay-agnostic
* voice-detection-frame-size-ms
* voice-detection-likelihood
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
self->eos was never reset after streamsynchronizer has sent EOS
(except on explicit flush or switching back to PAUSED).
As a result, synchronization was broken if new streams were pushed later
as gst_stream_synchronizer_wait() does not wait if self->eos is set.
Fix this by reseting self->eos on STREAM_START as that means a new
stream is being sent upstream and so a new EOS will follow later on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4749>
In the case of a gstreamer-full target type to static,
the GST_STATIC_COMPILATION is necessary on Windows to avoid
a different mangling from the external project using the
gstreamer-full libraries (ie dllimport).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
Allow a project to use gstreamer-full as a static library
and link to create a binary without dependencies.
Introduce the option 'gst-full-target-type' to
select the build type, dynamic(default) or static.
In gstreamer-full/static build configuration gstreamer (gst.c)
needs the symbol gst_init_static_plugins which is defined
in gstreamer-full.
All the tests and examples are linking with gstreamer but the
symbol gst_init_static_plugins is only defined in the gstreamer-full
library. gstreamer-full can not be built first as it needs to know what plugins
will be built.
One option would be to build all the examples and tests after
gstreamer-full as the tools.
Disable tools build in subprojects too as it will be built at the end of
build process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
According to the documentation this should never happen but apparently
does under certain circumstances. As the sockets are set non-blocking,
trying to read from them regardless should not cause any problems.
In all cases that were observed so far, the socket in question actually
has a packet queued up for reading.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4748>
This patch adds gst_egl_image_from_dmabuf_direct_target_with_dma_drm() and
add gst_egl_image_from_dmabuf_with_dma_drm() functions
New function gst_egl_image_from_dmabuf_direct_target_with_dma_drm(), where
gst_egl_image_from_dmabuf_direct_target() is a specialization of the first.
And gst_egl_image_from_dmabuf() is a specialization of new function
gst_egl_image_from_dmabuf_with_dma_drm()
Co-authored-by: Victor Jaquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4680>
It internally uses gst_gl_context_egl_get_dma_formats() instead of fetching
modifiers by itself.
Thus gst_egl_image_check_dmabuf_direct() is a decorator of this new function.
Co-authored-by: He Junyan <junyan.he@intel.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4680>
By calling the internal function gst_gl_context_egl_fetch_dma_formats() the an
array of structures holding a DMA fourcc format and its modifiers (another array of
structure holing modifier and if it's external only) will be stored.
Users would call gst_gl_context_egl_get_format_modifiers() to get the array of
modifiers of a specific DMA fourcc format.
Co-authored-by: He Junyan <junyan.he@intel.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4680>
The `switch (n_rear)` supports up to 5 rear channels, but our channel
set only had space for 3. Size the set properly to fix this.
This didn't actually cause any memory unsafety as `PUSH_CHAN` would stop
incrementing `n_rear` if the channel set is already full.
Thanks to @alatiera for noticing this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4712>
Since c0bf793c05 ("flvmux: Set PTS based on
running time") the timestamp of the output buffer is already in running
time. So using that for 'srcpad->segment.position' does not work correctly
because gst_aggregator_simple_get_next_time() will convert it again with
gst_segment_to_running_time().
This means that the timestamp returned by
gst_aggregator_simple_get_next_time() may be incorrect. For example, if
flvmux is added to a already runinng pipeline then the timestamp is too
small and gst_aggregator_wait_and_check() returns immediately. As a result,
buffers may be muxed in the wrong order.
To fix this, use the PTS of the incoming buffer instead of the outgoing
buffer. Also add the duration as get_next_time() is supposed to return the
timestamp of the next buffer, not the current one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4701>
While decodebin3 could handle changes in inputs (ex: changing codecs), there was
still one limitation which was when changing between sources which had
non-intersecting stream types (ex: switching from a video-only source to a
audio-only source). While the decoder *could* change to the proper codec ... it
would carry on using a `DecodebinOutputStream` associated to that stream
type (and therefore with pads with the wrong name).
In order to handle this:
* We notify the `MultiQueueSlot` of the change in `GstStreamType` if it already
had an associated inputstream (ex: the one associated with the static sink
pad)
* We detect such changes on the output of multiqueue as soon as
possible (i.e. when we get the GST_EVENT_STREAM_START for the new stream type)
by discarding the associated output.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1669
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4703>
There are broken(?) mjpeg videos that are incorrectly detected as
interlaced. This happens because 'info.height > height' (e.g. 1088 > 1080).
In the interlaced case info.height is approximately 'height * 2' but not
exactly because height is a multiple of DCTSIZE. Make the check more
restrictive but take the rounding effect into account.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4696>
For interlaced jpeg, gst_jpeg_dec_decode_direct() is called twice, once for each
field. In this case, stride[n] is plane_stride[n] * 2 to ensure that only every
other line is written. So the loop must stop at height / num_fields.
If the frame is really interlaced then continuing beyound this, is not harmful,
because jpeg_read_raw_data() will do nothing and return 0, so am info message is
printed.
However, if the frame is not actually interlaced, just misdetected as interlaced
then there is still data available from the second half of the frame. Now
line[0][j] is set to the scratch buffer. If the scratch buffer is not allocated
(because the height is a multiple of v_samp[0] * DCTSIZE) then the result is a
segfault due to a null-pointer dereference.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4696>
When the alignment contains nothing, all its fields are 0 and always
can be satisfied. So there is no need to validate it in this case.
And there are a lot of places just setting this alignment to default
all zero value, this validation generates lots of warnings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4674>
While this doesn't yet use any OS provided times from the actual network
stack, this still gets rid of any IPC jitter between the helper process
and the main process as part of the PTP time calculations and should
improve accuracy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4665>
On Windows and macOS always use the proper monotonic clock, including
for gst_util_get_timestamp(), and initialize its state only once.
Also on macOS use clock_gettime() for the realtime clock, if available
instead of always falling back to GLib.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4658>
Add d3d11 conversion path to make gst_video_convert_sample() work
for GstD3D11Memory.
Note that just adding "d3d11download" to the exisitng code is
suboptimal from GstD3D11 point of view because:
* d3d11convert element can support crop/colorspace-conversion/scale
all at once while existing software pipeline needs intermediate steps
for the conversion
* "Process everything on GPU then download it to CPU memory" would be likely
faster than "download GPU memory to CPU then processing it on CPU"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2715>
adjust log level from GST_ERROR to GST_WARNING when h264 caps have
codec_data but no avc format or have no codec data or stream-format.
Because theses are not real errors, it is easy to mislead if print error
logs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4675>
Drivers may signal end of sequence using an empty buffer and LAST buffer
set, or just an empty buffer on certain legacy implementation. When this
occured, we'd send GST_V4L2_FLOW_LAST_BUFFER were the code expected
GST_FLOW_EOS. Stop abusing GST_FLOW_EOS and port all the code to the new
GST_V4L2_FLOW_LAST_BUFFER.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4669>
ptpd is defaulting to the hybrid mode, and was sending invalid multicast
PTP messages in that configuration until ce96c742a88792a8d92deebaf03927e1b367f4a9.
While this commit was made in 2015 there was no release in the meantime.
Work around this by detecting this case and defaulting to the default
values for the given intervals as given by the PTP standard.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4654>
Previously it was possible that a shared media was just in the process
of being unprepared because the last client disappeared, while another
client retrieved it from the cache and then tried to use it. Unless the
media was reusable this would've then failed unnecessarily.
To avoid this it is necessary to lock the media directly in
gst_rtsp_media_factory_construct() and return a locked media. After
locking the cached media it is necessary to check if the media was ever
unprepared or is actually reusable and based on that either reuse it or
create a new media.
This minimally changes the gst_rtsp_media_factory_construct() API to
always return a locked media, and adds a new
gst_rtsp_media_can_be_shared() function to check if a media can actually
be shared in practice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
The resolution of VP9 video can be changed without keyframe.
The change detected by MSDK/VPL should be negotiated with downstream.
Only the situation can be fixed here if the changed resolution is less than or equal to the initial surface resolution.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4450>
New vulkan formats don't match the number of planes with the number of memories
attached to the buffer. This patch changes the pattern of using planes for
traverse the memories with the number of attached memories.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
It's a generalization of the original gst_vulkan_get_or_create_image_view().
The reason for passing the whole VkImageViewCreateInfo structure rather than
just the missing fields, is because VkImageSubresourceRange and
VkComponentMapping can be different and those are most of VkImageViewCreateInfo.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
This is going to be used when the pool is used by a video decoder for
VK_IMAGE_USAGE_VIDEO_DECODE_DST_BIT_KHR, since the frame allocation needs the
VkVideoProfileInfoKHR, and for that here GstCaps is used to wire it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
The specification says:
VUID-vkAllocateMemory-pAllocateInfo-01713
must pAllocateInfo->allocationSize be less than or equal to
VkPhysicalDeviceMemoryProperties::memoryHeaps[memindex].size where memindex =
VkPhysicalDeviceMemoryProperties::memoryTypes[pAllocateInfo->memoryTypeIndex].heapIndex
as returned by vkGetPhysicalDeviceMemoryProperties for the VkPhysicalDevice that
device was created from.
Though this can be catch by the validation layer, the requested frame size
depends on the use case so it's better to check this restriction by our code.
This patch also makes use of this new function to find memory type index,
and removes the unused function to find memory type index, which, as GstVulkan is
considered unstable, we can do it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
The purpose of this function is to get more info about the mapped Vulkan format
from the GStreamer format, since they can be multiple Vulkan formats for one
GStreamer format.
Also a Vulkan format may have certain usage and aspects that must be verified.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
Originally the opened device only created one queue of one family queue, to say
graphics one. This approach felt short when other queue family is required not
shared with the graphics queue family, for example video decoding.
This new approach proposes to create those queues with supported families. For
now, only video decoding and encoder are created, if they are available.
In order to hold multiple queues opened, an array of VkDeviceQueueCreateInfo is
held along the live the device object, because it's used to traverse or get the
opened queues.
The algorithm to choose which queues create (or open) is to look for the queue
with more family bits, which also supports the one we are requesting, thus
minimizing the number of global queues of a certain family to create.
Nonetheless, the number of queues to open per family is set to be all of them,
widening the possibility of parallelism.
Also, this commit do a cosmetic refactor the assigning the physical device
nearer where it's used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
Also adds a meson option to enable them.
The symbol GST_VULKAN_HAVE_VIDEO_EXTENSIONS is an alias of
defined(VK_VERSION_1_4) || (defined(VK_VERSION_1_3) && VK_HEADER_VERSION >= 238
if the option is allowed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
Subclasses may want to override the pad template with different formats
or with a different pad subclass.
The original beahviour is still available by calling
gst_gl_mixer_class_add_rgba_pad_templates() in _class_init() of the
subclass.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4608>
max-qp and min-qp will set the same quantizer scale for I/P/B frames,
while max-qp-i/p/b and min-qp-i/p/b enable the max/min quantizer for I,P,B
frame separately. When max/min-qp and max/min-qp-i/p/b are given
simultaneously, the later set one will overide the previous one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4589>
when play rtsp stream with playbin3 enabled, there are some critical logs:
g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-video'
g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-audio'
g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-text'
self->collection could be NULL when READY->PAUSED if the pipeline
is live, then it will fallback to query playbin2's property,
we can call gst_play_streams_info_create_from_collection
directly, it will check self->collection internal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4460>
In cases that encoder needs to reset format, there is race while draining.
v4l2videoenc finish() sends CMD_STOP command to driver, and desire to return
GST_FLOW_OK. But at this time, encoder CAPTURE may have dequeued the last
buffer and got eos. finish() return value changes to be GST_FLOW_EOS which
causes set format fail. So there is no need to check return value for finish()
when set format.
Also need to flush encoder after draining to make sure flush is finished.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4495>
When reconfigure_output_stream entry missing decoder path,
requested_selection should been update with what is really
active/selected immdiately with SELECTION_LOCK hold. So
use an optional message return from reconfigure_output_stream
and post it after release SELECTION_LOCK. This can make sure
other thread call to check_slot_reconfiguration will got
a correct requested_selection.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4599>
The index is already incremented by 3 every iteration so multiplying it
by 3 additionally on each array access is doing it twice and does not
work.
This caused invalid files to be created if there's more than one CEA608
triplet in a buffer, and out of bounds memory reads.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4634>
The devices list returned by g_udev_client_query_by_subsystem() may
contain udev devices in disorder path name. For example, on some
platform it may contain renderD129 before renderD128 device. This
will cause we register wrong va plugin name. In this case, the
renderD129 will be registered as default plugins such as vah265dec,
while the renderD128 will be registered as varenderD128h265dec.
This conflicts with the non-udev version of gst_va_device_find_devices().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4643>
Make splitmuxsrc deal better with stream reordering by
making the largest observed PTS contiguous in the
next fragment. Previously, it selected DTS, but then
aligned that with the segment start of the next fragment,
which holds PTS values - leading to glitches in
streams that don't have PTS = DTS at the start.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4637>
When transitioning from state PAUSED to READY, the sctpenc element
could previously be stuck in an endless loop trying to resend data
in case the underlying sctp stream was in the process of
resetting. usrsctp_sendv() would repeatedly return EAGAIN with the
result that 0 bytes were sent and then sctpenc would retry forever.
To bring sctpenc out of the resend loop we just need to inform the
sink pad that it is flushing, which is already done for the associated
data queue, but we also need to set the bools associated with the
sinkpads that are used as the loop criterion.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4601>
Assigning TRUE (1) to a signed 1 bit integer will cause truncation
from 1 to -1 because the only non-zero value that can be stored is -1
due to how two's-complement works.
As this is a proper GObject let's not bother with all this and simply
use a normal gboolean instead.
../subprojects/gst-plugins-good/ext/pulse/pulsesink.c:1490:19: warning: implicit truncation from 'int' to a one-bit
wide bit-field changes value from 1 to -1 [-Wsingle-bit-bitfield-constant-conversion]
pbuf->in_commit = TRUE;
^ ~~~~
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4617>
It could indeed be used uninitialized, but only if one of the
g_return_val_if_fail() caused an early return.
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c: In function ‘rtp_jitter_buffer_append_query’:
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c🔢10: warning: ‘head’ may be used uninitialized
[-Wmaybe-uninitialized]
1234 | return head;
| ^~~~
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c:1232:12: note: ‘head’ was declared here
1232 | gboolean head;
| ^~~~
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4616>
If gst_buffer_pool_set_config() fails then the pool will use its old
config. This may include different width or height when
pic_width/pic_height != frame_width/frame_height.
As a result, the assertions in theora_handle_image() will fail.
So check the result of gst_buffer_pool_set_config() and only use the pool
if it succeeds. Otherwise let the parrent decide_allocation() create a new
pool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4600>
If the buffer has no video meta then the meta is created from the local
data. In this case, the other asserts don't actually check anything. So add
another one to ensure that the buffer is actually large enough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4600>
Upon creating a window, glimagesink and osxvideosink now set the policy to
NSApplicationActivationPolicyRegular, which lets us show an icon in the Dock
for convenience and appear in the top menu bar like other apps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4573>
Setting the policy to NSApplicationActivationPolicyAccessory by default makes
sure that we can activate windows programmatically or by clicking on them.
Without that, windows would disappear if you clicked outside them and there
would be no way to bring them to front again. This change also allows osxvideosink
to receive navigation events correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4573>
This is no longer needed since the introduction of `gst_macos_main()` in 1.22.
Before that existed, we had a patch for GLib in Cerbero, which did work but made it
impossible to update GLib at all. The code being removed was a fail-safe in case of
running without said patch being applied. It's no longer needed, since for macOS
we just wrap our GStreamer with an NSApplication using `gst_macos_main()`.
Warnings will be displayed if no NSApp/NSRunLoop is found wherever needed,
pointing the user towards using the new API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4366>
Invoking gst_osx_video_sink_osxwindow_destroy() can currently cause a deadlock
because showFrame() keeps trying to get the same lock as well. Moving the lock
closer to where it's actually needed seems to be enough to fix the issue for now.
Reported-by: Alexande B <abobrikovich@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4559>
Depending on the exact output format, 0x00 may be a better default for
padding than 0x80. 0x00 is the recommended padding value when used in
CDP (and cc_data) but is not when used in s334-1a. See CTA-708-E 4.3.5
amd SMPTE 334-1-2007 5.3.2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4578>
This is a fix for a data race leading to:
> GLib-CRITICAL: g_hash_table_foreach:
> assertion 'version == hash_table->version' failed
Identified sequence:
* `rtp_session_on_timeout` acquires the lock on `session` and proceeds with its
processing.
* `rtp_session_process_rtcp` is called (debug log : received RTCP packet) and
attempts to acquire the lock on `session`, which is still held by
`rtp_session_on_timeout`.
* as part of an hash table iterator, `rtp_session_on_timeout` transitively
invokes `source_caps` which releases the lock on `session` so as to call
`session->callbacks.caps`.
* Since `rtp_session_process_rtcp` was waiting for the lock to be released, it
succeeds in acquiring it and proceeds with `rtp_session_process_rr` which
transitively calls `g_hash_table_insert` via `add_source`.
* After `source_caps` re-acquires the lock and gives the control flow back to
`rtp_session_on_timeout`, the hash table iterator is changed, resulting in the
assertion failure.
This commits copies `sess->ssrcs[sess->mask_idx]` and iterates on the copy so
the iterator is not affected by a concurrent change due to the lock being
released in the `source_caps` callback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4555>
The qt5 and qt6 plugins will now correctly error out if you enable the
option, and you can also now explicitly ensure that wayland, x11,
eglfs support is actually functional by enabling the options. It was
too easy to build non-functional support for these.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4537>
jackaudiosink and jackaudiosrc have a rank and might be plugged
as part of auto-plugging inside playbin and playsink or the
autoaudiosink/autoaudiosrc elements, so we don't really want to
spew ERROR log messages in that case, which is consistent with
what alsasink and pulseaudiosink do.
This is less noticable on Linux because pulseaudiosink has a
higher and alsasink which has the same rank comes before jack
in the alphabet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4545>
