When gst_element_set_state is called in _setup_locked and errors, the
callback is already processed before we reach handle_current_async, and
the timer is started even though it's finished processing, which results
in a NULL pointer crash later in async_timeout_cb.
To fix this, we check that it's still processing before calling
handle_current_async.
Fixes#1683
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4936>
Fixes test: validate.uridecodebin.expose_raw_pad_caps
testsrcbin (currently part of debugutilsbad) is an useful element for
validate tests.
validate.uridecodebin.expose_raw_pad_caps makes use of it.
Unfortunately, because validate tests with GStreamer only run with
whitelisted plugins and `debugutilsbad` wasn't in the whitelist, the
test was failing and being auto-skipped.
This patch adds debugutilsbad to the whitelists used by validate tests
in subprojects with a validate/meson.build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4931>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
Proxy the force-live and min-upstream-latency propertyies to the internal
glvideomixerelement at construction time. force-live has to be set
during construction of the glvideomixerelement, so that has to be
deferred until the _constructed() call. Make sure that all other
existing proxied properties will still get set once the element
is created.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4494>
decodebin3 will do its best to figure out whether a parsebin is required to
process the incoming stream.
The problem is that for push-based stream it could happen that the stream would
not provide any caps, resulting in nothing being linked internally.
Furthermore, there is the possibility that a stream *with* caps would not be
using a TIME segment, which is required for multiqueue to properly work.
In order to fix those two issues, we force the usage of parsebin on push-based
streams:
* When the pad is linked, if upstream can't provide any caps
* When we get a non-TIME segment
Fixes#2521
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4492>
The proxy and queue are created in the gst_gl_window_wayland_egl_open()
function and will be recreated on open. This leaks both objects, the
wayland client documentation mentions that they should be destroyed
using the appropriate destroy functions.
Found during valgrind memory leak testing, these blocks were marked as
definitely lost.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4354>
The first serialized events that can be send on a src pad are a CAPS and then a
SEGMENT event.
When handling events from user in appsrc, we used to send a segment
automatically if the SEGMENT has not been sent yet.
This breaks if the CAPS event was not send either as we were now sending
a SEGMENT before the CAPS.
Fix this by delaying such events until the CAPS has been configured.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4297>
Existing codes rely on modified argc value by g_option_context_parse()
but g_option_context_parse_strv() is used in case of Windows.
Count arguments after the option parsing manually.
Fixing command "gst-inspect-1.0.exe -b"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4313>
Moving from PLAYING to NULL will set the stop_streaming_threads to TRUE,
but when moving back upwards its not reset to FALSE (as only done in
uncalled init and resume callbacks).
Fix by reseting value in the prepare callback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4309>
Adding propose_allocation is to meet the requirement of Application to
request buffers. Application sometimes need to create buffer pool
and request buffers to maintain buffer management itself, and Gstreamer plugin
import Application's buffers to use. So, add propose_allocation in
appsink like waylandsink and kmssink etc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4185>
Fix compiler warnings about not using the return value when
freeing the GString segment with g_string_free(.., FALSE):
ignoring return value of ‘g_string_free_and_steal’ declared with attribute ‘warn_unused_result’
which we get with newer GLib versions. These were all harmless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4273>
This is a follow-up of the previous commit that enabled support for redirection.
The problem is that the urisourcebin that emitted the error redirection never
produced any pads, and therefore was never linked to decodebin3. This resulted
in the code waiting for that (output) item to finally switch over ... which will
never happen.
The fix is done by removing it early if it was never connected to decodebin3.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4252>
The goal of parsebin is to figure out which elements to link together in order
to provide elementary streams given any random input.
The problem is that deciding whether a given stream should still have more
elements plugged in or not was dependent on ... the presence of compatible
decoders (sic).
Instead of that, if we can't plug anymore elements on a given stream *and* it is
detected as being an elementary stream, expose it.
Fixes#2118
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4231>
If sticky events are present on parsebin source pads, we propagate them to the
multiqueue source pads. Those will be propagated on the new urisourcebin source
pads like in the other code paths.
This ensures that STREAM_START event are present on new source pads. If CAPS
event are also present (not guaranteed), they will also be available.
Fixes#2384
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4203>
In gst_video_info_dma_drm_to_caps() the caps are newly created, so there's no
need for make it writable. In gst_video_info_dma_drm_from_caps() a copy of the
caps is done, which implies a gst_caps_make_writable().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4195>
This allow allocating memory from any DRM driver that supports this
method. It additionally allow exporting DMABuf. This allocator depends
on libdrm and will be stubbed if the dependency is missing. This is derived
from kmssink dumb allocator.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3801>
Read and flush console buffer from the console thread immediately,
instead of main thread. Otherwise (if main thread is busy)
the console thread will keep adding idle source and then main thread
will be unresponsive.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4067>
