Back in the mists of time[1], we switched `giostream*` elements to not close the
stream on stop() so that applications that needed a handle to the stream after
the element stopped had it.
Unfortunately, we also have cases[2] where waiting for the element to be
finalized is too late for the stream to be closed.
In order to not change the behaviour of the element, we add a property to allow
users to select the desired behaviour.
[1]: https://bugzilla.gnome.org/show_bug.cgi?id=587896
[2]: gst-plugins-rs#423
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5372>
By using the gst_caps_set_simple() to set the format on all structures, the
compositor may create invalid combinations as the caps may contain passthrough
caps. Avoid this issue by intersecting the resul with its original.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
Adds list of formats that should be used by element in needs to passthrough
video. It contains the full list of video format plus DMA_DRM format
and will be extended in the future as needed. This patches includes 3 new
symbols:
- GST_VIDEO_FORMATS_ANY_STR
- GST_VIDEO_FORMATS_ANY
- gst_video_formats_any()
The last one can be used by bindings or for code that prefers having
GstVideoFormat values instead of strings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
Right now we split the RTP header from the current buffer into a new
buffer and aggregate those buffers for later processing if the
depayloader creates an output buffer.
This is cumbersome as it happens even if none of the incoming RTP
buffers carries RTP header extensions at all just because header
aggregation has been enabled in the depayloader class.
This commit will start aggregation only in case that there really are
RTP header extensions available on an incoming RTP buffer. The check
is trivial and cheap. Once activated we keep aggregation active for
all buffers. The active state is reset on state change READY_TO_PAUSE.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5278>
The GST_VIDEO_FORMAT_Y410, GST_VIDEO_FORMAT_Y412_LE and GST_VIDEO_FORMAT_Y412_BE
formats in fact are packed formats, which have just 1 plane. But we have special
setting for them rather than using get_single_planar_format_gl_swizzle_order().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5314>
As we don't have any mapping from YUV formats + modifiers to an equivalent
emulated format (e.g. NV12 + modifier -> R8+modifier/RG88+modifier), do no
allow these formats to be used with the indirect DMABuf uploader.
Fixes#2942
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5270>
The same is done in the set_property function. This was noticed when attempting
to dump a pipeline containing glsinkbin sink=gtk4paintablesink to dot format.
Critical warnings were raised due to the missing force-aspect-ratio property on
that sink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5311>
Fixes a potential GPU stall if an immediately freed texture/buffer is
attempted to be reused immediately by the CPU, e.g. when uploading.
Problematic scenario is this:
1. element does GPU processing reading from texture
2. frees the buffer back to the pool
3. pool acquire returns the just released buffer
4. GPU processing then has to wait for the previous GPU operation to
complete causing a stall
If there was a reliable way to know whether a buffer had been finished
with across all GPU drivers, we would use it. However as that does not
exist, this workaround is to keep the released buffer unusable until the
next released buffer.
This is the same approach as is used in the qml (Qt5) elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5144>
When using `gst_sdp_media_set_media_from_caps` on `application/x-rtp` caps
without `clock-rate` it wrongly reports missing payload type even if `payload`
is present in the caps.
This seems to be a copy&paste error from the error message for missing payload
type.
When using payload=10, both `clock-rate` and some other media properties are
defined by the RTP standard so I was wondering whether I could omit `clock-rate`
and was confused about the error message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5250>
Add gst_audio_ring_buffer_set_errored() that will mark the
ringbuffer as errored only if it is currently started or paused,
so gst_audio_ringbuffer_stop() can be sure that the error
state means that the ringbuffer was started and needs stop called.
Fixes a crash with osxaudiosrc if the source element posts
an error, because the ringbuffer would not get stopped and CoreAudio
would continue trying to do callbacks.
Also, anywhere that modifies the ringbuffer state, make sure to
use atomic operations, to guarantee their visibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5205>
If a depayloader aggregates multiple RTP buffers into one buffer only
the last RTP buffer was checked for header extensions. Now the
depayloader remembers all RTP packets pushed before a output buffer is
pushed and checks all RTP buffers for header extensions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
Don't call wait_event() at all for gap events, as basesink will
end up waiting for the time that the gap event would be rendered
out at the audio device. There's no need to render it at all,
just treat it as a handy point to resync the audio if needed,
let the ringbuffer render silence, and place the next buffer
into the ringbuffer where it belongs.
The only thing we really need to do is make sure the ringbuffer
and clock are running, and wait for preroll.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2749
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5178>
These 10bit formats are identical to NV12_16L32S, but 64bytes of data is being
prefixed with 16bytes data with four pixels of lower 2bits per byte. For
MT2110T, the lower two bits set so each bytes contains a column of 4 pixels,
also describe as tiled lower 2 bits. MT2110T has been chosen as a name to match
the vendor chosen name. This format is unlikely to exist for other vendors.
For MT2110R, the 2 low bits are in raster order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3444>
The current limit is `x10`, which allows just `+20 dB` of gain.
While it may seem sufficient, this came up as a problem
in a real-world, non-specially-engineered situation,
in strawberry's EBU R 128 loudness normalization.
(https://github.com/strawberrymusicplayer/strawberry/pull/1216)
There is an audio track (that was not intentionally engineered that way),
that has integrated loudness of `-38 LUFS`,
and if we want to normalize it's loudness to e.g. `-16 LUFS`,
which is a very reasonable thing to do,
we need to apply gain of `+22 dB`,
which is larger than `+20 dB`, and we fail...
I think it should allow at least `+96 dB` of gain,
and therefore should be at `10^(96/20) ~= 63096`.
But, i don't see why we need to put any specific restriction
on that parameter in the first place, other than the fact
that the fixed-point multiplication scheme does not support volume
larger than 15x-ish.
So let's just implement a floating-point fall-back path
that does not involve fixed-point multiplication
and lift the restriction altogether?
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5063>
When this flag is enabled, the transform_caps() simply set passthrough
to generate the raw caps. This is not correct, because the sink and
src have different format/drm-format fields.
We already add system memory conversion for DMABuf manner, so no more
need for this flag.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _nvmm_upload_transform_caps() only simply apply
"memory:NVMM" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
And in _nvmm_upload_accept(), we should only accept the "memory:NVMM"
feature in input caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _directviv_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
And in _directviv_upload_accept(), we should only accept the system
memory as input caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _gl_memory_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
And in _gl_memory_upload_propose_allocation(), we should only allocate
the allocator and buffer pool for the caps with "memory:GLMemory"
feature.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _upload_meta_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _raw_data_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
We also should recognize the system memory caps in _accept() early, if
the input is not system memory, we just return early.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
Most of the time, the RGB kind formats are OpenGL native supported
format which has only one plane. They can be imported at one shot
using no matter DIRECT or INDIRECT mode.
While YUV kind formats which have multi planes have two ways to import.
They can be DIRECT imported, which requires GL_OES_EGL_image_external
extension. The output format should be RGBA and TARGET should be set
as OES after imported. The other way, they can be INDIRECT imported,
which makes each plane as a texture. In this mode, the imported textures
have different fourcc from the original format. For example, the NV12
format can be imported as a R8 texture for the first plane and RG88
texture for the second plane. The output TARGET should be sets as 2D
in this mode.
When converting sink caps to src caps, we first filter the feature of
"video/x-raw(memory:DMABuf)" and system memory. Then Based on the
external_only flag (INDIRECT mode does not care while DIRECT mode cares),
we transform the drm-format into the gst video format.
When converting src caps into sink caps, we first filter the correct
TARGET(INDIRECT mode contains 2D only while DIRECT mode contains 2D,
OES or both of them) gstructure. Then Based on the include_external flag
(INDIRECT mode always true while DIRECT mode depends on TARGET), we
transform the gst video format into drm-format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
When switching from a raw stream to an encoded stream we need to make sure the
slot is unlinked, there is code in place for this but it wasn't triggered
because the slot being reconfigured wasn't advertised as linked beforehand.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5126>
Shader compilation was failing on macOS:
gstglslstage.c:519:_compile_shader:<glslstage1> fragment shader compilation failed:
ERROR: 0:10: 'input_swizzle' : syntax error: Array size must appear after variable name
Co-authored-by: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5123>
Don't assume that compositor will output only single buffer
for single input buffer. If buffer's running time is not completly
aligned to output buffer running time or duration, compositor
can generate multiple buffers. If that happens, two threads,
one is aggregator output thread and main thread were trying
to modify buffer in this test. Clear the buffer after
shutting down pipeline to avoid the race.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2836
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5081>
Do not attempt to send a streams-selected message when reassigning
an output slot in case upstream signalled that it is handling stream selection.
In this case decodebin3 doesn't keep track of stream
collections (`dbin->collection` is NULL).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5059>
The current way of dma caps uses the drm-format to replace the orginal
format field. The absence of format field means it can accept all formats.
It causes problems when clipping with other old DMA or video/x-raw(ANY)
caps, the result will contain both format field and drm-format field,
which is not valid DMA caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
This GST_VIDEO_FORMAT_DMA_DRM is introduced for DMABuf kind feature
usage. It represent the DMA DRM kind memory. And like the ENCODED
format, it should not be interpreted and mapped as normal video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
Setting the input field on the empty slot prevents future linking of it and will
result in flow errors later on.
This was observed in WebKit's MediaStream source element, when it changes the
caps on one of its associated streams, from an encoded format to a raw video
format. The associated stream-id on the sticky stream-start event doesn´t
change, but the element creates a new GstStream with a different ID and sets it
on the stream-start event. Stream parsing is disabled in urisourcebin, so
decodebin3 handles the parsing. Without this patch we would end-up with unlinked
pads in decodebin3 after switching to the raw video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5048>
The drop-frame rules are specified in “SMPTE ST 12-3:2016” and are
consistent with the traditional ones:
“
To minimize fractional time deviation from real time, the first two
super-frame numbers (00 and 01) shall be omitted from the count at the
start of each minute except minutes 00, 10, 20, 30, 40, and 50. Thus the
first eight frame numbers (0 through 7) are omitted from the count at
the start of each minute except minutes 00, 10, 20, 30, 40, and 50.
”
Where “super-frame” is a group of 4 frames for 120 FPS.
Fixes#2797
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5028>
The current implementation copies metas without checking if the buffer
is writable.
The operation that needs to be done, replacing the input buffer and
copying the metas, is only part of that process. We create a new function
that does both.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4912>
This reverts commit 893e4ed0dd.
This caused regressions in existing elements which override/set things
like QoS and such in their own init functions. If the base class does
this in ::constructed() now it will override the subclass settings
again with its own, which can have unintended side-effects.
Case in point is gdkpixbufsink which disabled QoS there, and this
patch would reliably make the unit test fail in valgrind because
now frames are dropped because of QoS (when QoS should really be
disabled).
Fixes#2794
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5006>
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>
