d3d11: Run indent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3250>

wic: Add since marker

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3250>

directshow: Add since marker

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3250>

examples: webrtc: Update to gstreamer-rs 0.19 release

Also update the macOS workaround for gstreamer-gl requiring a
`NSRunLoop` / `NSApp` on the main thread, and update from strucopt to
clap 4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3255>

qsv: Enable MinGW toolchain support

Use PCRE regex method to work around (likely) GCC std::regex bug,
and enable building for non-MSVC

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3239>

rtspsrc: Only EOS on timeout if all streams are timed out/EOS

Otherwise a stream that is just temporarily inactive might time out and
then can never become active again because the EOS event was sent
already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3238>

gl/vulkan: Fix static linking on macOS

duplicate symbol '__invoke_on_main' in:
    /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstvulkan-1.0.a(cocoa_gstvkwindow_cocoa.m.o)
    /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstgl-1.0.a(cocoa_gstglwindow_cocoa.m.o)
ld: 1 duplicate symbol for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)

Also make the same change in iOS for consistency.

Continuation of https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1132

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3242>

webrtcbin: configure rtpulpfecdec passthrough property

This allows downstream (payloaders mostly) to be able to correctly
detect actual packet loss from rtp sequence numbers.

See
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/581
for background.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1407

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3212>

rtpulpfecdec: add property for passthrough

Support for enabling and disabling decoding of FEC data decoding on
packet loss events and unconditional seqnum rewriting of packets.

See
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/581
for background.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3212>

webrtc: Fix critical in webrtc-recvonly-h264 example

This signal only takes 2 properties yet a third was passed.
This would cause a critical in GLib.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3252>

net: Add missing nullable annotation on the name parameter in the net/NTP clock constructors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3249>

padtemplate: Fix annotations

gst_caps_replace() does not take ownership of the new caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3248>

gst-libav: Fix synchronization issues and errors created by the forwarding of
segment events by demuxer.

In order to play nicely with `ffmpeg`, demuxers in `gst-libav` have to make
buffers available to `ffmpeg` while taking the blocking I/O model in `ffmpeg`
into account, which results in buffers not being sent downstream until `ffmpeg`
has processed them in its separate thread.

In constrast, many `gstreamer` events are simply forwarded downstream.
Currently `GST_EVENT_SEGMENT` events are forwarded downstream without any
processing, which can potentially result in:
 * `GST_EVENT_SEGMENT` events being out of sync with buffers
 * `GST_EVENT_SEGMENT` events going out that are incorrect because they apply
   to data seen by the demuxer, but not necessarily seen by downstream elements

I came across this bug when I was attempting to enable G723.1 demuxing/decoding
using the G723.1 demuxer and decoder provided by `ffmpeg`.  I wrote tests to
verify support for the functionality, and found that, in push mode,
`GST_EVENT_SEGMENT` events pushed to the demuxer by the upstream `filesrc`
element would be forwarded to the decoder without modification, resulting in
an internal data streaming error.  With this patch, tests work in both push and
pull mode.

This patch solves the problem by disabling the forwarding of
`GST_EVENT_SEGMENT` events downstream (an initial `GST_EVENT_SEGMENT` event is
still pushed downstream by the demuxer).  It's possible there's a better way to
do this, but, having looked at how a few different `gstreamer` demuxers deal
with `GST_EVENT_SEGMENT` events, it seems like the processing is somewhat
specific to the demuxer implementation, whereas `gst-libav` has one general way
of handling the situation for any `ffmpeg` demuxer.  Perhaps there's a better
way to solve this using the `ffmpeg` API to take advantage of specific demuxer
details.  IDK.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3218>

gst-inspect: Don't print link to doc if it's known to be unavailable

"gst_element_factory_get_skip_documentation() == true" means
documentation was intentionally skipped for the element feature

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3243>

gst: uri query: fix inconsistent `uri` nullability  assertion

Functions `gst_query_set_uri` and `gst_query_set_uri_redirection`
can both set a `NULL` `uri`, as annotated in the documentation.
However the functions bodies reject `NULL` `uri`s.

See [1] for a discussion on that matter.

[1]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1134#note_1600988

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3237>

d3d11screencapture: Add WinRT API based capture mode

Add Windows Graphics Capture (WGC) API based screen capture mode.
The conditions where this mode is used:
* Explicitly requested by user (capture-api property)
* To capture specific window
* When DXGI desktop duplication API does not work on hybrid graphics systems
  (e.g., multi-gpu laptop)

Full features of this implementation require Windows 11. And Windows 11
SDK is required to build this feature.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3144>

d3d11screencapture: Subclassing capture implementation

Preparation to use WinRT capture API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3144>

qsv: Add plugin doc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2738>

qsv: Update SDK version to v2022.2.4

See release note
https://github.com/oneapi-src/oneVPL/releases/tag/v2022.2.0

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2738>

av1parse: Correct the pts for frames and OBUs inside a TU.

When the output alignment is smaller than the input alignment, for
example, When the output alignment is "FRAME" and the parse is likely
connecting to a decoder, the current PTS setting for AV1 frames inside
a TU is not very correct.

For example, a TU may begin with non-displayed frames and end with a
displayed frame. The current way will assign the PTS to the first
non-displayed frame, which is a decode-only frame and the PTS will be
discarded in the video decoder. While the last displayed frame has
invalid PTS, and so the video decoder needs to guess its PTS based on
the frame rate and previous frame's PTS. This is not a decent and
robust way. And more important, when the previous frames provide DTS,
the video decoder will also guess the PTS based on the previous frames'
DTS and trigger the warning like:

  gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
  <vavp9dec0> decreasing timestame

It sets the reordered_output and makes the decoder in free run mode.

We should correct the PTS for a TU, let the non-displayed frames have
no PTS while set the correct PTS to the displayed one. Also, when the
AV1 stream has multi spatial layers, there are more than one displayed
frames inside one TU with the same PTS.

Note: If the input alignment is not TU aligned, we can not know the
exact PTS of this TU, and so we just clear the PTS of the decode only
frame and leave others unchanged.

We also correct all the PTS if the output is OBU aligned. All their
PTS and DTS are set to the input buffer's PTS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>

av1parse: Only check the TU bound when the alignment is TU.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>

av1parse: push all data once when input alignment not smaller than output.

When the incoming data has big alignment than the output, we do not need to
call finish_frame() and exit the current handle_frame() for each splitted
frame. We can push them all at one shot with in one handle_frame(), whcih
may improve the performance and can help us to find the edge of TU.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>

av1parse: Set the output buffer flags correctly.

The current code forgets to clear some flags and has some typo.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>

docs: plugins-scanner: Handle interface used for plugin API properties

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236>

webrtcbin: also add rtcp-fb ccm fir for video mlines by default

In addition to the 'nack pli' already added.  Both are supported by
rtpbin/rtpsession by default already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3235>

build/gl: fix automatic dispmanx detection for rpi4

rpi4 doesn't contain dispmanx but still contains bcm_host.pc which
confuses the configure detection.  Add an explicit check for
EGL_DISPMANX_WINDOW_T to ensure that we only build with dispmanx when
the types are available.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/893
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/952
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3230>

webrtcbin: Improve documentation of 'turn-server' property

Description about how to set time-limited credentials is added.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3229>

docs: plugin-scanner: Stop updating "long-name" metadata

The "long-name" value can be environment dependent, and it's not
actually used by our documentation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3208>

parse: Adjust debug log level

That's not an error case at all

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3228>

docs/glcontextconfig: rename title to not be GstGLContext

Can be confusing havint two pages that both have GstGLContext as the
title.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3227>

{element,deviceprovider}factory: g_object_new() can't ever return NULL

So treat it as the assertion it is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3226>

elementfactory: Handle element factory loading failure in gst_element_factory_create_valist() not as assertion

In gst_element_factory_create_with_properties() it is a normal error
path so let's keep this consistent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3226>

gst: Use G_TYPE_INVALID instead of 0 for GTypes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3226>

tests: parse-launch: remove assignment order tests

These tests relied on setting the name of an element twice to verify
that the last one set took precedence, however name is a CONSTRUCT property
and the parser now errors out when such properties are set twice, in
g_object_new_with_properties .

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3026>

parse: refactor to make use of gst_element_factory_make_with_properties

Instead of creating the element first, then setting properties and
presets, we gather those and construct the element with the properties.

This means users of gst_parse_launch can now set construct-only
properties.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1380

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3026>

gstminiobject: shares capacity increase

during the MSE (WebKit) tests from Apple suite:
https://hls-streaming.cdn-apple.com/hls_conformance/dist/v1.1/index.html?pretty=true&whitelist=MSE%20Suite
webkit attempts to add a single audio buffer containing ~35.5k frames.
when corresponding GstSamples are pulled buffer is being referenced
more than object capacity allows: 2^15-1. since the case could be considered
malformed a surgical patch is applied to increase the capacity.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3221>

fake{video|audio}sink: don't proxy properties at instance init.

Instead proxy properties from the GstBaseSink class at class_init time,
and duplicate the rest of the fakesink properties manually.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1442

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3073>

cdparanoia: Ignore compiler warning coming from the cdparanoia header

When trying to build the plugin, GCC starts complaining about issues
with one of the cdparanoia headers and it block us from being able
to build the plugin with Werror.

The current warning in the header look like this:

```
[1/2] Compiling C object subprojects/gst-plugins-base/ext/cdparanoia/libgstcdparanoia.so.p/gstcdparanoiasrc.c.o
In file included from ../subprojects/gst-plugins-base/ext/cdparanoia/gstcdparanoiasrc.h:37,
                 from ../subprojects/gst-plugins-base/ext/cdparanoia/gstcdparanoiasrc.c:31:
/usr/include/cdda/cdda_interface.h:164:3: warning: initialization discards ‘const’ qualifier from pointer target type [-Wdiscarded-qualifiers]
  164 |   "Success",
      |   ^~~~~~~~~
...
/usr/include/cdda/cdda_interface.h:163:14: warning: ‘strerror_tr’ defined but not used [-Wunused-variable]
  163 | static char *strerror_tr[]={
      |              ^~~~~~~~~~~
[2/2] Linking target subprojects/gst-plugins-base/ext/cdparanoia/libgstcdparanoia.so
```

Last release of cdparanoia was in 2008, so our best bet for the
time is to ignore the warnings.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2722>

webrtcbin: support adding TURN servers after pipeline start

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1680

When a TURN server is added for the first time, propagate
the change to all nice streams that already exist. This
lets us add servers after the pipeline has been started
(and streams have been added).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3217>

meson: Use run_command check: true for windows binary subprojects

Fixes a warning about using the check kwarg

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3224>

examples: webrtc: mp-sendrecv: add bus handler

Without this bus handler, messages posted to the bus will keep a ref to
their source elements, preventing them from being disposed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>

examples: webrtc: mp-sendrecv: set element states to NULL after removing them from pipeline

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>

examples: webrtc: mp-sendrecv: remove wrong gst_object_unrefs

In !2958 some gst_object_unrefs were added. However these two don't
belong there because ownership is transfered due to the gst_bin_add_many
call a bit above.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>

srtp: Fix test skipping when plugin option is disabled

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3200>

d3d11videosink: Add "emit-present" property

Controls the decision for "present" signal use, and
allows delayed "present" signal handler install via the property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3176>

play: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

sdp: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

tag: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

allocators: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

vulkan: Add/fix various annotations

And fix a memory leak in gst_vulkan_display_wayland_new() in error
cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

d3d11: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

cuda: Add/fix various annotations

And fix a memory leak when creating a CUDA context fails.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

codecs: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

badaudio: Add/fix annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

ges: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

rtsp-server: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

core: Add/fix various annotations

And fix memory leaks/null pointer dereferences in GstUri in error cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

check: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

net: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

base: Add/fix annotations in GstQueueArray

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

gl: Add/fix various annotations

And fix a memory leaks in gst_gl_display_egl_new() error cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

validate: Add/fix various annotations

And fix monitor constructors to correctly check for valid arguments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

pbutils: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

rtp: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

rtsp: Add/fix various annotations

Also initialize out parameters so that they don't have an undefined
value in case of errors.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

controller: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

app: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

audio: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

video: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

webrtc: Add/fix various annotations

And mark string parameters as const.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

urisourcebin: Fix usage of raw and non-raw source provider

The computation in analyze_source was wrong, and would state that the element
has "all raw source pads" if it had at least one.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1029

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3199>

rtpbuffer: Initialize extended timestamp to the first wraparound period

This allows correct handling of wrapping around backwards during the
first wraparound period and avoids the infamous "Cannot unwrap, any
wrapping took place yet" error message.

It allows makes sure that for actual timestamp jumps a valid value is
returned instead of 0, which then allows the caller to handle it
properly. Not having this can have the caller see the same timestamp (0)
for a very long time, which for example can cause rtpjitterbuffer to
output the same timestamp for a very long time.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1500

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3202>

examples/webrtc: don't use factory_make_full() for enums

They are not currently translated into their respective enum values and
will produce an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3210>

wasapi2: Add support for process loopback capture

Adding loopback capture mode for specified PID.

Note that this feature requires Windows 10 build 20348
(Windows 11/Windows Server 2022 or later),
and any process loopback related properties will not be exposed
if OS does not support it.

Example launch lines:
* wasapi2src loopback-mode=include-process-tree loopback-target-pid=<PID>
 Captures audio generated by an application (specified by PID)
 and its child process
* wasapi2src loopback-mode=exclude-process-tree loopback-target-pid=<PID>
 Captures desktop audio excluding PID and its child process

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3195>

ci: Run windows jobs when win-* binary subprojects are updated

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3209>

meson: Update flex, bison, and nasm

Latest flex is 2.6.4, bison is 3.8.2, nasm is 2.15.04

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3209>

examples/webrtc/signalling: Fix compatibility with Python 3.10

Fix asyncio throwing a deprecation warning when using
asyncio.get_event_loop().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3196>

rtmp2sink: Correctly return GST_FLOW_ERROR on error

If there is an error while connecting, the streaming task will be stopped, and
is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,
we perform the error check (!self->connection) first, to return an error if
that's what occured.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3189>

gst-play: Don't leak the stream collection

We are given a reference to the collection when parsing it from the
message. Just store it (instead of referencing it again).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3186>

webrtc/nice: fix small leak of split strings

g_strfreev previously stopped at our manual NULL-termination. Fix by
restoring the pointer after joining.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3188>

gst-docs: Fix typo in hotdoc kwarg

The hotdoc module passes unknown keyword arguments as arguments to
hotdoc, and the fatal warnings argument is --fatal-warnings.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972#note_1586361

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3178>

wavparse: Avoid occasional crash due to referencing freed buffer.

We've seen occasional crashes in the `wavparse` module associated with
referencing a buffer in `gst_wavparse_chain` that's already been freed.  The
reference is stolen when the buffer is transferred to the adapter with
`gst_adapter_push` and, IIUC, assuming the source doesn't hold a reference to
the buffer, the buffer could be freed during interaction with the adapter in
`gst_wavparse_stream_headers`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3179>

rtsp-client: Remove duplicate documentation

Confuses the documentation builder, since it's documented twice it complains
about a missing "Since:" marker whereas it's present in the documentation
comment further down

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>

riff: Mark jpeg as parsed

This is needed so that autoplugging works with avidemux and JPEG decoders that
need parsed sink caps (eg rockchip 'mppjpegdec' decoder). It also works fine
with 'jpegdec' decoder regardless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3175>

wavparse: Fix crash that occurs in push mode when header chunks are corrupted
in certain ways.

In the case that a test is provided for, the size of the `fmt ` chunk is
changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
```
$ hexdump -C corruptheadertestsrc.wav
00000000  52 49 46 46 e4 fd 00 00  57 41 56 45 66 6d 74 20  |RIFF....WAVEfmt |
00000010  12 00 00 00 01 00 01 00  80 3e 00 00 00 7d 00 00  |.........>...}..|
00000020  02 00 10 00 64 61 74 61                           |....data|
00000028
```

(Note that the original file is much larger.  This was the smallest sub-file
I could find that would generate the crash.)

Note that, while the same issue doesn't cause a crash in pull mode, there's a
different issue in that the file is processed successfully as if it was a .wav
file with zero samples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>

vp9parse: The show_existing_frame buffer should not be decode only.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>

vp9parse: Correct the pts for frames inside a super frame.

When the alignment is "FRAME" and the parse is likely connecting to
a decoder, the current PTS setting for VP9 frames inside a super
frame is not very correct.

For example, the super frame may begin with non-displayed frames and
end with a displayed frame. The current way will assign the PTS to
the first non-displayed frame, which is a decode-only frame and the
PTS will be discarded in the video decoder. While the last displayed
frame has invalid PTS, and so the video decoder needs to guess its
PTS based on the frame rate and previous frame's PTS. This is not a
decent and robust way. And more important, when the previous frames
provide DTS, the video decoder will also guess the PTS based on the
previous frames' DTS and trigger the warning like:

  gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
  <vavp9dec0> decreasing timestame

It sets the reordered_output and makes the decoder in free run mode.

We should correct the PTS for a super frame, let the non-displayed
frames have no PTS while set the correct PTS to the displayed one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>

avfvideosrc: Allow specifying crop coordinates during screen capture

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3153>

rtsp-server: Free client if no connection could be created

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3164>

mxfdemux: Add support for Canon XF-HEVC

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1495

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3163>

mxfdemux: Don't leak index table segments on failures

The segment was freed ... but not the contents :)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3163>

rtsp-server: Add since marker for adjust_error_code

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3157>

meson: Update libsoup.wrap

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3162>

videorate: Do not close segment when getting a same segment twice

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>

videorate: Handle closing segment on EOS right after caps event

The scenario is what we try in the tests:
- we have a segment with .stop set
- some frame(s) flow
- we get a CAPS event
- we get an EOS (before getting buffers after the CAPS event)

in that case, without that patch, the segment is not properly closed
which is not correct. In this patch we keep track of previous caps until
a new buffer arrives, this way in that situation we set previous caps
again, and close the segment with the previous buffer.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1352
in this specific case

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>

validate: Plug some leaks

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>

oss4: Fix debug category initialization

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1456

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3158>

webrtc: Move GST_WEBRTC_ERROR_TYPE_ERROR at the end of the enum to keep ABI compatibility

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3154>

tests/webrtc: Add test for 'add-turn-server' action signal

It just checks return value of the action signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3131>

sctpenc: Prohibit sending of interleaved message parts

Apparently we cannot start sending messages from another datachannel
before the previous message was completely sent. usrsctplib will
complain about being locked on another stream id and set
errno=EINVAL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2454>