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>
