This is based on gtksink, but similar to waylandsink uses Wayland APIs
directly instead of rendering with Gtk/Cairo primitives.
Note that the long term plan is to move this into the existing extension
in `-good`, which requires the Wayland library to move the as well.
For this reason several files like `gstgtkutils.*` and `gtkgstbasewidget.*`
are straight copies and should be kept in sync.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1515>
This can be important for instance when a container holds multiple
tracks with the same media type, with no indication (eg tags) of
which track is the default one.
In that case, players usually pick the first track by default.
This is especially useful when using smart editing with GES, as
it will result in the same ordering as the input file that was
used as a template.
For reference, this yields the same order as ffprobe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
The previous code was storing container children in reverse
addition order, this was mitigated by the fact that track elements
were also stored in reverse order, thus restoring the original
order, but it seems more consistent to preserve order throughout,
the extra cost of append operations is negligible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
when creating a profile from a discoverer info.
There is no justification for the existing code, and talking with
Thibault he cannot remember why the sort was in place.
On the other hand, this allows GES users to not have to implement
a callback for the select-tracks-for-object callback when using
it to trim a single clip, which the output profile was built from:
track elements will be placed in the appropriate track by default,
that is the one that will be connected to the matching profile.
For multi-clip timelines, the situation doesn't change, users will
still have to implement a callback and do the leg work of placing
track elements (if any) in a matching track (if any).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
chroma-format, bit-depth-chroma, bit-depth-luma are all informative
fields set by the H265 and H265 parser upon receiving an SPS.
They shouldn't be constrained downstream of the parser, instead
if a user wants those to ultimately match certain values they
should do so by constraining a profile.
In this case however, we also always remove the profile constraint
in order to let encoders pick a suitable one as a function of the
raw input video format and their own capabilities.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
With the 2.72 release, glib-networking developers have decided that
TLS certificate validation cannot be implemented correctly by them, so
they've deprecated it.
In a nutshell: a cert can have several validation errors, but there
are no guarantees that the TLS backend will return all those errors,
and things are made even more complicated by the fact that the list of
errors might refer to certs that are added for backwards-compat and
won't actually be used by the TLS library.
Our best option is to ignore the deprecation and pass the warning onto
users so they can make an appropriate security decision regarding
this.
We can't deprecate the tls-validation-flags property because it is
very useful when connecting to RTSP cameras that will never get
updates to fix certificate errors.
Relevant upstream merge requests / issues:
https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
Forgot to change the wrap type in e0014ef4fe which broke the
subproject. Wasn't noticed by CI because the subproject cache wasn't
regenerated.
The accompanied patch was included in 2.8.2, so it is not needed. It
was originally needed with 2.8.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2812>
For formats which we don't have fast-path implementation, compositor
will convert it to common unpack formats (AYUV, ARGB, AYUV64 and ARGB64)
then blending will happen using the intermediate formats.
Finally blended image will be converted back to the selected output format
if required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1486>
It is entirely possible for the cancellable to be cancelled (and freed)
in gst_rtsp_connection_flush() while there may be an ongoing read/write
operation.
Nothing prevents gst_rtsp_connection_flush() from waiting for the
outstanding read/writes.
This could lead to a crash like (where cancellable has been freed
within gst_rtsp_connection_flush()):
#0 0x00007ffff4351096 in g_output_stream_writev (stream=stream@entry=0x7fff30002950, vectors=vectors@entry=0x7ffe2c6afa80, n_vectors=n_vectors@entry=3, bytes_written=bytes_written@entry=0x7ffe2c6af950, cancellable=cancellable@entry=0x7fff300288a0, error=error@entry=0x7ffe2c6af958) at ../subprojects/glib/gio/goutputstream.c:377
#1 0x00007ffff44b2c38 in writev_bytes (stream=0x7fff30002950, vectors=vectors@entry=0x7ffe2c6afa80, n_vectors=n_vectors@entry=3, bytes_written=bytes_written@entry=0x7ffe2c6afb90, block=block@entry=1, cancellable=0x7fff300288a0) at ../subprojects/gst-plugins-base/gst-libs/gst/rtsp/gstrtspconnection.c:1320
#2 0x00007ffff44b583e in gst_rtsp_connection_send_messages_usec (conn=0x7fff30001370, messages=messages@entry=0x7ffe2c6afcc0, n_messages=n_messages@entry=1, timeout=timeout@entry=3000000) at ../subprojects/gst-plugins-base/gst-libs/gst/rtsp/gstrtspconnection.c:2056
#3 0x00007ffff44d2669 in gst_rtsp_client_sink_connection_send_messages (sink=0x7fffac0192c0, timeout=3000000, n_messages=1, messages=0x7ffe2c6afcc0, conninfo=0x7fffac019610) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:1929
#4 gst_rtsp_client_sink_try_send (sink=sink@entry=0x7fffac0192c0, conninfo=conninfo@entry=0x7fffac019610, requests=requests@entry=0x7ffe2c6afcc0, n_requests=n_requests@entry=1, response=response@entry=0x0, code=code@entry=0x0) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:2845
#5 0x00007ffff44d3077 in do_send_data (buffer=0x7fff38075c60, channel=<optimized out>, context=0x7fffac042640) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:3896
#6 0x00007ffff4281cc6 in gst_rtsp_stream_transport_send_rtp (trans=trans@entry=0x7fff20061f80, buffer=<optimized out>) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream-transport.c:632
#7 0x00007ffff4278e9b in push_data (stream=0x7fff40019bf0, is_rtp=<optimized out>, buffer_list=0x0, buffer=<optimized out>, trans=0x7fff20061f80) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2586
#8 check_transport_backlog (stream=0x7fff40019bf0, trans=0x7fff20061f80) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2645
#9 0x00007ffff42793b3 in send_tcp_message (idx=<optimized out>, stream=0x7fff40019bf0) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2741
#10 send_func (stream=0x7fff40019bf0) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2776
#11 0x00007ffff7d59fad in g_thread_proxy (data=0x7fffbc062920) at ../subprojects/glib/glib/gthread.c:827
#12 0x00007ffff7a8ce2d in start_thread () from /lib64/libc.so.6
#13 0x00007ffff7b12620 in clone3 () from /lib64/libc.so.6
Fix by adding a cancellable lock and returning an extra reference used
across all read/write operations. gst_rtsp_connection_flush() can free
the in-use cancellable and it will no longer affect any in progress
read/write.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2799>
1.21.0.1 should not satisfy a check for 1.22.0.
If someone needs more control they should do a feature check for
the symbol in the headers or lib.
Based on a similar patch by Tim-Philipp Müller for libnice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2501>
The caps negotiation should respect the selected method to the test pipeline below works properly.
gst-launch-1.0 videotestsrc ! video/x-raw,width=320,height=600 ! videoflip method=clockwise ! video/x-raw,width=600,height=320 ! fakesink
Signed-off-by: Adrian Fiergolski <adrian.fiergolski@fastree3d.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2803>
In the trick mode, driver may queue a valid buffer follow by an
empty buffer which has no valid data to indicate EOS.For the empty
buffer whose memory is multi-plane, need to resize it before
unreference it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2731>
Depending on device feature level, d3d11 runtime can support
ID3D11Fence which is equivalent to ID3D12Fence.
Waiting using fence has performance-wise benefit over pulling
ID3D11Query status. If ID3D11Fence is not supported by device,
then ID3D11Query will be used instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2790>
4x downscaling of chroma with co-sited chroma has never worked
it seems.
Fixes incorrect videotestsrc output and videoconvert conversions
to Y41B, YUV9, YVU9 and IYU9 with co-sited chroma.
e.g.
gst-launch-1.0 videotestsrc ! video/x-raw,format=Y41B,width=1280,height=720 ! \
videoconvert ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2789>
It may happens that bitstream doesn't provided SPS in decoding order
(like in VPSSPSPPS_A_MainConcept_1 conformance test file).
To be sure that the decoder got the correct SPS parameters process
SPS just before start decoding the frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2575>
