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>
This commit ports functionality from the `rtpsrc` to make the `ristsrc`
work with dynamic payload types.
It adds two properties:
- `caps`
- `encoding-name`
These can be used to make the `ristsrc` receive other payload types than
the MPEG TS one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5422>
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.
With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.
Fixes#2443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
The counter was using a signed 8 bit integer, which was overflowing
after 127 entries. That was then passed as an unsigned 32 bit integer to
libflac, which caused it to be converted to a huge unsigned number.
That then caused an invalid memory access inside libflac.
As a bonus, signed integer overflow is undefined behaviour.
Instead, use an unsigned 8 bit integer. Once this overflows the existing
code already catches it and stops adding the cue. While FLAC__metadata_object_cuesheet_insert_track()
takes an unsigned 32 bit integer for the track number, FLAC__StreamMetadata_CueSheet_Track is
limiting it to an unsigned 8 bit integer.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2921
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5420>
Since DXVA does not support some profiles such as HEVC RExt,
vendor specific decoding API is still required.
When decoder is negotiated with d3d11 caps, decoder will convert
semi-planar frame to planar since semi-planar format (e.g.,
DXGI_FORMAT_NV12) is not supported by CUDA/D3D11 interop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5409>
Found that osxaudiosink could not be added standalone in gst-full build
using
-Dgst-full-elements=osxaudio:osxaudiosink because element registration
was
done at the plugin level. Now src/sink elements and deviceprovider have
their
individual registration.
Copied/adapted from the alsa plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5419>
Use gst_codec_utils_caps_get_mime_codec() in pbutils for codec
strings. That function gives more elaborate RFC 6381 compatible
strings than the helper functions in gstmdphelper.c, such as
"avc1.F4000D".
Remove the helper functions, as they were only used from dashsink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5404>
When the property "start-time-selection" is set to "first", it
calculates the start time of the output from the buffer pts
(converting it to running time of the segment), but if the
rate is negative, the real start is not the pts, but the
pts + duration, because it plays from the end of the buffer
to it's start.
As a result of this bug, in the negative rate, when the
start-time-selection=first, the first frame is dropped
by the videoaggregator (reproduced on d3d11compositor).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5276>
The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5324>
Move the GstStructure field into public struct for direct access, that's
easier than having to call a function to get it. It is not an API/ABI
breakage to extend the public structure of a GstMeta because they are
always allocated by inside GStreamer. The structure is exposed already
by gst_custom_meta_get_structure() which does not return a copy/ref, so
it is locked into holding a GstStructure forever anyway.
Also add gst_meta_register_custom_simple() because most of the time only
a name is required, tags and transform functions are more niche
use-case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5385>
If there are multiple Wayland event listeners in different threads we
get the formats and modifiers pushed concurrently which leads to
segfault from GArray methods. This patch protects the array.
The problem occurs e.g. when using vaapipostproc together with Qt
qmlglsink, QtWayland will get the events as well as VAAPI.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5280>
Moves outputting frames to a task on the source pad, bringing vtdec in line with vtenc.
This brings possible performance improvements thanks to decoupling queueing new frames from outputting processed ones.
The queue length is limited to `2*DBP` to prevent decoding too far ahead compared to what we're pushing downstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5163>
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>
scanlines->m1 = same line of the previous field
scanlines->t0 = line above of the current field
scanlines->b0 = line below of the current field
scanlines->mp = same line of the next field
Deinterlacing a field weaved frame:
When deinterlacing the top field, the next bottom field is available
(part of the same frame). but when deinterlacing the bottom field,
the next top field (part of the next frame) is not available and
scanlines->mp equals NULL.
In this case it's better to use greedy algorithm using the prevous field
(twice) rather then linear interpolation of the current field.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5331>
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet
then this can't be used for bitrate estimation, and also not for
constructing the next RTCP SR. Both would end up with completely wrong
values, and an RTCP SR with wrong values can easily break
synchronization in receivers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
