in the case of an upstream element proposing a buffer pool,
use it to allocate the buffer image with the given parameters
set by the upstream element.
Besides the buffer pool handling is sync'd with GstBaseTransform
base class.
See the case of vulkanupload ! vulkanh264enc
Co-authored-by: Victor Jaquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5651>
The code seems to validate that the media-level fingerprint matches
the fingerprint of the previous media or of the whole session. There
is no such requirement in any RFC I found. The session-session one
is just meant to act as a fallback when there is no media-level
fingerprint.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1118>
A payload of 0x80 0x80 means that it's padding. It's not a good idea to
throw this away though, because of the cc_valid field.
According to CEA 10-B section 25.2.1, if cc_valid is zero, the run-in
clock and start bit should not be generated. In practice, this means
that any closed captions will be erased and the end-user TV will show
that captions are not available for this stream. This might have
undesired consequences, e.g. we were just showing a long line of
captions and we disable it before the user has had time to read it, or
you can't enable closed captions during silence/music intervals.
We cannot reliably detect whether there's a currently-silent closed
caption stream or just nothing, but we have this information coming from
upstream, so we can at least not discard it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5508>
The src caps of the libde265 is now fixed to I420, and so if the
stream is other format, such as 4:4:4 or 10 bits format, the pipeline
will crash because the dowstream element accesses the video buffer as
I420 format.
We now restrain the input caps to "main" profile, which only contains
4:2:0 8 bits stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5573>
This can happen with the dummy "noopenh264" library that the freedesktop
flatpak runtime ships, and Fedora is planning on shipping as well. In
both cases the dummy implementation gets replaced with the actual
openh264 library that's downloaded directly from Cisco, but just to be
on safe side, this patch makes it careful to check the return values to
avoid crashing if the underlying library hasn't been swapped out yet.
The patch is taken from freedesktop-sdk and was originally written by
Valentin David <valentin.david@codethink.co.uk>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5581>
Previously we were checking for opencv dep in 2 different places,
and the checks would vary in terms of how complex and exhaustive
they were.
Move the check into the libs module and reuse the result later on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3016>
This element refactors functionality from gstonnxinference element,
namely separating out the ONNX inference from the subsequent analysis.
The new element runs an ONNX model on each video frame, and then
attaches a TensorMeta meta with the output tensor data. This tensor data
will then be consumed by downstream elements such as gstobjectdetector.
At the moment, a provisional TensorMeta is used just in the ONNX
plugin, but in future this will upgraded to a GStreamer API for other
plugins to consume.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4916>
There are a bunch of plugins that you need for webrtc support, and
it's not obvious at all to users which those are.
With this commit, srtp, sctp and dtls options will be auto-enabled if
the webrtc option is enabled.
Requires meson 1.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5505>
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>
There is no need to use DRM dumb pool if buffer to
render is already a DMABuf, just import it and render it.
This fixes a DMAbuf memory leakage when waylandsink downstream
element exports DMABuf while waylandsink is configured to be
DMABuf exporter (drm-device=/drv/dri/card0):
gst-launch-1.0 v4l2src io-mode=4 ! gtkwaylandsink drm-device=/dev/dri/card0
leakage identfied with command:
watch "cat /sys/kernel/debug/dma_buf/bufinfo | grep attached "
Fixes#2729
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5350>
There is no need to use DRM dumb pool if buffer to
render is already a DMABuf, just import it and render it.
This fixes a DMAbuf memory leakage when waylandsink downstream
element exports DMABuf while waylandsink is configured to be
DMABuf exporter (drm-device=/drv/dri/card0):
gst-launch-1.0 v4l2src io-mode=4 ! waylandsink drm-device=/dev/dri/card0
leakage identfied with command:
watch "cat /sys/kernel/debug/dma_buf/bufinfo | grep attached "
Fixes#2729
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5350>
* Library versioning should not be used for plugins since it will add
-{version}.dll suffix (and versioned libraries on Linux with symlink).
Then the library file name and plugin init function name mismatch
will result in blacklisted plugin.
* Don't define BUILDING_GST_CODECS, makes no sense
* Don't define G_LOG_DOMAIN, which should be used only for libraries,
not plugins
* Depends on gstcodecparsers libary, not gstcodecs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5249>
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.
By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:
http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc
This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
