Update openjp2 and libxml2 from wrapdb

This fix their static link for Windows.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2084>

libs: va: add VA allocator parameter for derived images usage.

Added GstVaFeature enum type, and new parameter for VA allocator's
set_format() and get_format(). Also added a new parameter in VA pool
gst_va_pool_new_with_config() and
gst_buffer_pool_config_set_va_allocation_params().

This new parameter will define if derived images will by used for
buffer mapping.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2057>

webrtcstats: Unify 'packets-lost' data type to int64

Previously, 'packets-lost' member of RTCReceivedRtpStreamStats had
a value of G_TYPE_INT from rtpsource or a value of G_TYPE_UINT64
from rtpjitterbuffer.
Because of the negative value of estimated amount of packets lost
in rtpsource as well as the description in
https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats
it is fixed to set this value with G_TYPE_INT64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2049>

overlay: Fix qt support detection

On Ubuntu moc-qt5 command is called moc. This requires Meson 0.54.0 for
the new has_tools() method.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2075>

va: encoder: Remove unused allocation config.

Since it's also done inside of gst_va_pool_new_with_config().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2072>

vadisplay: Add description readonly property

Expose description of vendor for user information, similar to
the description property of d3d11device.
Also, set description and DRM device path on GstContext structure
so that user can read them and it will be printed on terminal
when gst-launch-1.0 is used

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2064>

gst-play: Improve Win32 keyboard input handling

The console HANDLE will be keep signalled state unless application
reads console input buffer immediately. So we should read and flush
console input buffer from the thread where the event is signalled,
instead of GMain context thread.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2058>

tools: Add support for building gstreamer tools against gst-full

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1581>

playsink: improve GL context sharing

Configure playsink tried element with the bus of the main pipeline.
That tried element can be a gl video sink, which would benefit from being
able to propagate context messages to the main pipeline and have other
internal pipeline elements configured with it. Having different elements
configured with the same GL context allows them to share buffers with
video/x-raw(memory:GLMemory) caps and achieving zero-copy.

Thanks to Alicia Boya García <aboya@igalia.com> for her work co-debugging
the issue and contributing to find a solution.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2056>

docs/design: Updates for upstream selection

Update existing documentation for various behaviours.

Add new information on how selection "upstream" of decodebin3 happens.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>

urisourcebin: When streams-aware, remove pads immediately

For the same reason we add them immediately

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>

urisourcebin: Don't wait for pads content when streams-aware

If the adaptive demux is streams-aware it can add/remove pads at any point in
time without the need for no-more-pads or data blocking

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>

urisourcebin: Don't do buffering if source already does

Sources that can internally handle buffering shouldn't have yet-another
buffering element after it. This can be simply detected by checking if it can
answer a TIME BUFFERING query just after creation.

If that is the case, we can expose the element source pads directly

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>

decodebin3: Handle upstream selection

Detect if upstream handles stream-selection, and if so bypass all stream
selection handling (streams are forwarded as-is).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>

query: Add a new stream selection query

This new API allows querying whether elements can handle stream selection
themselves or not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>

ci: Don't tie up a full job slot for cerbero trigger

We don't have a huge amount of capacity on shared runners, so better if
we don't use that limited capacity on a job which sits there doing
~nothing.

There is a new runner class called 'placeholder-job' which accepts a
huge number of parallel jobs, separately to normal jobs, on the
assumption that they will consume ~no resources; start using that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2071>

va: Don't expose internal classes.

VA allocators and pools classes don't need to be exposed as external
symbols.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2070>

gstplay: don't print error log in warning_cb

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2069>

vapostproc: Build classification dynamically.

By default, the classification is
"Converter/Filter/Colorspace/Scaler/Video/Hardware", but if VA
post-processor driver supports either color balance, skin tone
enhancement, sharpening or noise reduction, "Effect" is added.

Thus, if vapostproc ranking is raised, it can be chosen by
autovideosink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2066>

webrtc: produce stats for all relevant streams

Instead of only using the last ssrc that was pushed into a sink pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtc: avoid different versions of gnu-indent always wanting to change !!

Add some sneaky parenthesis to avoid always having to use git commit -n
or revert out hunk of the change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtc: implement initial simulcast fec/rtx usage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtc/datachannel: fix use-after-free in sctp state notification

g_signal_disconnect*() doesn't stop any existing callbacks from running
which means that if the notify::state callback is in progress in one
thread and the data channel object is finalize()ed in another thread,
then there could be a use-after-free trying lock the data channel
object.

We can't reasonably use a GWeakRef as we don't have a 'parent' object to
free the GWeakRef after the data channel is finalized.  This is also
complicated by the fact that the application can hold a reference to the
data channel object that would live beyond the lifetime of webrtcbin
itself.

We solve this by implementing a ghetto weak-ref solution internally with
a list of outstanding data channels.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

tests/webrtc: fix a use-after-free in test_data_channel_close

g_object_weak_ref() is not thread-safe and the data channel object's
refs/unrefs can happen on multiple threads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

tests/webrtc: fix a race in the tests related to state tracking

If things progress fast enough, some state changes may not be seen be
the waiting code.

Fix by:
1. keeping a list of all the state changes
2. waiting checks each entry and if the relevant state is found, all
   states up to and including then are removed.

This ensures that any waits will see all the state sets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

tests/webrtc: factor out src pad property checking to a separate function

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: initial support for sending and receiving simulcast streams

Input (sink pads) is the already-ssrc-muxed stream with the relevant rtp
sdes header extensions already applied:
  - mid
  - stream-id
  - repaired-stream-id

Output (src pads) have the pads separated into individual ssrc's as
that's what rtpbin gives us.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

tests/webrtc: test for enabled bundled fec/rtx

Doesn't actually check that any fec/rtx happens, just that the pipeline
is vaguely sane and doesn't error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: support multiple received streams for a single mline

Each rtpbin exposed recv_src pad is now exposed as webrtcbin src_%u pad
now with no meaining applied to the value of %u.  Previously this used
to mean the mline in the SDP.  If this is is still required, then the
transceiver can be retrieved from the pad and the "mlineindex" property
from the transciever.  The "mid" is also retrievable from the
transceiver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtc: explicitly error out in a couple of renegotiation cases

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtc/transportstream: add debug category

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

tests/webrtc: also check valid mline for srcpad codec-preferences negotiation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: don't name src pads based on the mline specifically anymore

Naming based on the mline doesn't really work with e.g. simulcast
scenarios.

It is entirely possible to retrieve the transceiver and then the mline
from that if that is so required.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtc/tests: print the correct media idx on error

Instead of the attribute index

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtc/tests: give slightly better names to the dot file dumps

Don't use printf-specifiers with g_strconcat().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: improve some debugging output

- Put human readable names into debug strings.
- Demote some frequent rtpbin signal logging
- Don't use GST_PTR_FORMAT in g_set_error()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: silence spurious warning when creating answer transceiver

When creating a transceiver when creating an answer, the media kind of the
transceiver was never set correctly initially.  This would lead to a
GST_WARNING being produced about changin a transceiver's media kind.

Fix by retrieving the GstSDPMedia kind from the offer instead as the answer
GstSDPMedia has not been set as the answer caps have not been chosen yet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

tests/webrtc: always use a unique SSRC for each stream

Will become more relevant with mid/rid->ssrc mappings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: add a specific find_transceiver_by_mid function

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: explicitly use a variable for the rtp session idx

Slightly clearer in meaning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

webrtcbin: support using an a=mid value from the sink/transceiver caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

va: Set <gst/va/gstva.h> as library single point entry header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2048>

va: Move allocators and pool objects into gstva library.

In order to other plugins use gstva objects, such as allocators and buffer
pools, this merge request move them from the va plugin to the gstva library.

This objects are not exposed in <gst/va/gstva.h> since they are not expected
to be used by users, only by plugin implementators.

Because of the surface copy design, which is used to implement allocator's
mem_copy() virtual function, depends on the vafilter, which is kept inside
the plugin, memory copy through VAPosproc is disabled and removed temporarly.

Also added some missing parameter validation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2048>

va: Move the video format functions into gstlibva.

Untabifying header file.

The logging category was moved from the plugin generic category to
the display category. It can argue that video formats hacks are
display dependant.

Added validations for input parameters.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2048>

meson: Add some messages when selecting libsoup

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2032>

soup: Fix usage of symbols / defines that are gone in libsoup3

I am not sure about the SOUP_MESSAGE_OVERWRITE_CHUNKS change, but it
was definitely already broken when using libsoup-3.0 in a shared
build. souphttpsrc probably needs to be ported from SoupMessage to
SoupServerMessage when using libsoup-3.0.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1111

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2032>

soup: Fix pre-processor macros in souploader for libsoup-3.0

Some of the preprocessor conditionals in the loader were very broken
with libsoup-3.0 + --default-library=static

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1111

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2032>

pbutils: Fix wmv screen detection

strncmp vs !strncmp :)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2050>

ges: remove memory leak with description

free the capsdesc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2047>

nle: clear seek event properly

Use gst_clear_event instead of g_clear_object
avoiding a failing gobject unref

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2047>

qsvencoder: Add support for dynamic bitrate update

... and add more encoding options.

QSV API supports dynamic bitrate change without IDR insertion.
That's more efficient way of runtime encoding option update
than starting from new sequence with IDR per bitrate option change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2039>

qsvh265enc: Add missing gop-size property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2039>

qsv: Fix mfxFrameAllocator::Lock for encoder

Only read map is possible for encoder input system memory

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2039>

python: Add a suppression file for a leak in PyGObject

And ensure that the CI runs GES valgrind test when we change the
overrides as many GES tests are implemented in python

Proper fix is at: https://gitlab.gnome.org/GNOME/pygobject/-/merge_requests/204

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2060>

avviddec: Remove vc1/wmv3 override

FFMPEG 5+ doesn't allow overriding the codec anymore (causes a segfault if you
attempt to do that). But the best part is ... that with the current caps
implementation in pad template and gst_ffmpeg_caps_to_codecid() we would never
replace it by anything different than the existing codec id.

Fixes #1054

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2052>

docs: Fix doc generation example

They need to be generated from the devenv else it just hangs for no reason.

See #1108

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2051>

python: Add support for the GstURIHandlerInterface

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1423>

validate/media-info: silence unsed variable warning

Fixes:

../validate/gst/validate/gst-validate-media-info.c:714:28: error: variable 'total_sink_count' set but not used [-Werror,-Wunused-but-set-variable]
  guint id, ncounters = 0, total_sink_count = 0;
                           ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

rtsp-stream: remove unused variable:

Fixes:

../gst/rtsp-server/rtsp-stream.c:2670:9: error: variable 'n_messages' set but not used [-Werror,-Wunused-but-set-variable]
  guint n_messages = 0;
        ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

applemeida/texturecache: remove unused variable

Fixes:

../sys/applemedia/videotexturecache.m:71:20: error: variable 'features' set but not used [-Werror,-Wunused-but-set-variable]
  GstCapsFeatures *features;
                   ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

applemedia/corevideobuffer: remove unused variable

Fixes:

../sys/applemedia/corevideobuffer.c:209:19: error: variable 'video_meta' set but not used [-Werror,-Wunused-but-set-variable]
    GstVideoMeta *video_meta;
                  ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

applemedia/iosgl: remove unused variable

Fixes:

../sys/applemedia/iosurfaceglmemory.c:219:41: error: variable 'texfmt' set but not used [-Werror,-Wunused-but-set-variable]
    GLuint tex_id, tex_target, texifmt, texfmt;
                                        ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

vtdec: remove unused variable

Fixes:

../sys/applemedia/vtdec.c:611:35: error: variable 'output_flags' set but not used [-Werror,-Wunused-but-set-variable]
  VTDecodeFrameFlags input_flags, output_flags;
                                  ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

avsamplevideosink: remove unused variable

Fixes

../sys/applemedia/avsamplevideosink.m:80:20: error: variable 'gstelement_class' set but not used [-Werror,-Wunused-but-set-variable]
  GstElementClass *gstelement_class;
                   ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

avfassetsrc: fix unused-but-set warning

../sys/applemedia/avfassetsrc.m:1014:12: error: variable 'caps' set but not used [-Werror,-Wunused-but-set-variable]
  GstCaps *caps;
           ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

speed: fix unused-but-set warning

../gst/speed/gstspeed.c:523:39: error: variable 'base' set but not used [-Werror,-Wunused-but-set-variable]
      gint64 start_value, stop_value, base;
                                      ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

resindvd: silence unused-but-set warning

../ext/resindvd/gstpesfilter.c:117:11: error: variable 'STD_buffer_size_bound' set but not used [-Werror,-Wunused-but-set-variable]
  guint16 STD_buffer_size_bound;
          ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

mpegts: don't shadow res variable

Fixes unused-but-set warning:

../gst/mpegtsmux/gstbasetsmux.c:2115:43: error: variable 'res' set but not used [-Werror,-Wunused-but-set-variable]
      gboolean all_headers, done = FALSE, res = FALSE;
                                          ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

mpeg: fix unused-but-set warning

../gst-libs/gst/mpegts/gst-dvb-section.c:206:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
  guint i = 0, allocated_events = 12;
        ^
../gst-libs/gst/mpegts/gst-dvb-section.c:365:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
  guint i = 0, allocated_streams = 12;
        ^
../gst-libs/gst/mpegts/gst-dvb-section.c:543:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
  guint i = 0, allocated_streams = 12;
        ^
../gst-libs/gst/mpegts/gst-dvb-section.c:885:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
  guint i = 0, allocated_services = 8;
        ^
../gst-libs/gst/mpegts/gst-dvb-section.c:1316:9: error: variable 'i' set but not used [-Werror,-Wunused-but-set-variable]
  guint i = 0, allocated_services = 8;
        ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

osxcoreaudio: fix unused-but-set warning

../sys/osxaudio/gstosxcoreaudio.c:480:18: error: variable 'interleaved' set but not used [-Werror,-Wunused-but-set-variable]
  gboolean sign, interleaved;
                 ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>

video-format: Move NV12_8L128 into the correct position in GST_VIDEO_FORMATS_ALL

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2045>

mpegdemux: silence unused-but-set werror

../gst/mpegdemux/gstpesfilter.c:117:11: error: variable 'STD_buffer_size_bound' set but not used [-Werror,-Wunused-but-set-variable]
  guint16 STD_buffer_size_bound;
          ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2044>

Fix GstAmcSurfaceTexture segfault
Check that `self` and `self->callback` are defined. `self` can be set to
`NULL` in `remove_listener`, and `self->callback` can be set to `NULL`
inside `gst_amc_surface_texture_jni_set_on_frame_available_callback`.
This can cause a segfault since the Java object can outlive the C
object, and call the callback after `remove_listener` is called.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2024>

examples/player/android: add missing dummy.cpp

allows libc++_shared.so to be placed in the application

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2043>

deinterlace: silence unused-but-set werror from imported code

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2042>

examples/player/android: update for android changes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2041>

osxvideosink: fix unused-but-set-variable warning

../sys/osxvideo/osxvideosink.m:859:11: error: variable 'data' set but not used [-Werror,-Wunused-but-set-variable]
  guint8 *data, *readp, *writep;
          ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2040>

rawvideoparse: set format from caps in gst_raw_video_parse_set_config_from_caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1970>

glmixerbin: slightly better pad/element creation

Use the return value from gst_element_link_pads() and gst_bin_add()

Fixes:

../ext/gl/gstglmixerbin.c:305:12: error: variable 'res' set but not used [-Werror,-Wunused-but-set-variable]
  gboolean res = TRUE;
           ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2038>

ccconverter: ensure correct ordering of cea608 across output buffers

e.g. if a 60fps output is configured, we can only produce a single field
of cea608 that must alternate between field 1 and field 2.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2019>

ccconverter: ignore padding cea608 data even if marked as 'valid'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2019>

v4l2codecs: Detect missing M2M_HOLD_CAPTURE_BUF

Produce an error if we try to use the feature of holding capture buffer
but it is not supported by the driver. Ingoring this can lead to stalls
as the driver will run-out of capture buffer to decode into. This
affects slice decoders but also split-field interlaced decoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2009>

v4l2codecs: h264: Set other field poc if available

Whenever the other field POC is available, pass it back to the
driver. This should not be strictly required, but it makes things
easier to debug and more forgiven.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2009>

v4l2codecs: Trace request by it FD

This helps when crossing the debug logs with the kernel logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2009>

v4l2codecs: h264: Fix dpb entry flag FLAG_FIELD

The logic to guess back this value was broken. Use the value
now saved into the picture instead. This was tested using
LibreELEC patched 5.15 kernel, and fixed the interlaced decode
issues.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2009>

codecs: h264: Save the field_pic_flag

This flag is set when the stream is interlaced and the specific
slice is made of single parity fields rather the paired at the
macroblock layer. This is rarely needed in late decoding process
but the Rockchip RKVDEC HW interface requires it, hence needs to
be passed through V4L2 Stateless interface.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2009>

d3d11: Fix example build error

```
../subprojects/gst-plugins-bad/tests/examples/d3d11/d3d11device.cpp:129:50: error: invalid conversion from ‘int’ to ‘DXGI_FORMAT’ [-fpermissive]
  129 |     D3D11_SHADER_RESOURCE_VIEW_DESC srv_desc = { 0, };
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2029>

gl: cocoa: fix warnings of unused variables

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2026>

Fetch libxml2 subproject using http instead of ftp

The official releases of libxml2 have been migrated to gitlab where
they are published for download via HTTP instead of FTP. Besides
adapting to the new location we now also get the benefit that the
tarball can be downloaded in restricted networks where FTP might be
blocked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2020>

navigationtest: Add some support for modifiers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2010>

navigation: Add support for key Modifiers in all relevant events

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2010>

va: h265dec: Fix a crash because of missing reference frame.

Some problematic H265 stream may miss the reference frame in the DPB,
and get some message like: "No short term reference picture for xxx".
So there may be empty entries in ref_pic_list0/1 when passing to
decode_slice() function of sub class. We need to check the NULL pointer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2018>

openexr: Specify modules when finding OpenEXR.

Specify modules to look for OpenEXR when CMake is used, as we may have
CMake config files instead of pkg-config files that result from building
OpenEXR, which may be built with CMake which is typically the case on Visual
Studio builds.

In this case, Meson does seem to find the 'OpenEXR' package with CMake
after trying pkg-config, but does not consider it enough without the
'modules:' argument.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2014>

webrtcbin: Update documentation of 'get-stats' action signal

Some stats fields are updated according to the current implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2005>

rtpptdemux: fix leak of caps when ignoring a pt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2025>

Revert "webrtcbin: fix msid line and allow customization"

This reverts commit 3cad3455377d5a22faa138d9df840257059776c8.

That commit was breaking the association between an audio and
a video track in the standard case.

In practice, to support carrying separate MediaStream, we are
going a way to map what MediaStreamTrack belong to what MediaStream,
but that will require some thinking about the API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2023>

tsdemux: Don't check SCTE events for being too far from the PCR

Otherwise it happens that SCTE events can only be scheduled up to 15
seconds since the last seen PCR, which is a useless restriction.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2021>

navigation: Add missing annotation to send_event_simple

Adds the missing "transfer full" annotation for the event argument.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2017>

webrtcbin: fix msid line and allow customization

From https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-msid-16:

> Multiple media descriptions with the same value for msid-id and
> msid-appdata are not permitted.

Our previous implementation of simply using the CNAME as the msid
identifier and the name of the transceiver as the msid appdata was
misguided and incorrect, and created issues when bundling multiple
video streams together: the ontrack event was emitted with the same
streams for the two bundled medias, at least in Firefox.

Instead, use the transceiver name as the identifier, and expose
a msid-appdata property on transceivers to allow for further
customization by the application. When the property is not set,
msid-appdata can be left empty as it is specified as optional.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2003>

webrtc: Update dependencies of the Rust examples

And also clean up code a bit while updating to new APIs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2016>