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>
