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>

opengl: Fix usage of eglCreate/DestroyImage

The implementation was inconsistent between create and destroy. EGLImage
creation and destruction is requires for EGL 1.5 and up, while
otherwise the KHR version is only available if EGL_KHR_image_base
feature is set. Not doing these check may lead to getting a function
pointer to a stub, which is notably the case when using apitrace.

Fixes #1389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2925>

rtsp-server: Add support for adjusting request response on pipeline errors

The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().

Fixes #1294

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>

splitmuxsrc: don't queue data on unlinked pads

Once a pad has returned NOT_LINKED, the part reader shouldn't let its
corresponding data queue run full and eventually (after 20 seconds)
stall playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3145>

rtpsource: Don't do probation for RTX sources

Disable probation for RTX sources as packets will arrive very
irregularly and waiting for a second packet usually exceeds the deadline
of the retransmission.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>

rtp: examples: client-rtpaux: Provide correct caps by payload type and RTX pt map by session

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>

tests/check/rtpsession: extend test_internal_sources_timeout

to verify that rtx SSRCs do not BYE after timeout

See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>

rtpsession: Remember the corresponding media SSRC for RTX sources

This allows timing out the RTX source and sending BYE for it when the
actual media source belonging to it is timed out.

This change only applies to sending sources from this session.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>

rtpsource: Rename rtp_source_update_caps to rtp_source_update_send_caps

To make it clear that this is only used for sending RTP sources.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>

rtpsession: Rename gst_rtp_session_sink_setcaps to gst_rtp_session_setcaps_recv_rtp

to make it clearer that this is for setting receiver caps and to make it
more consistent with gst_rtp_session_setcaps_send_rtp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>

decodebin3: allow to call "dispose" multiple times

https://docs.gtk.org/gobject/concepts.html#reference-counts-and-cycles

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3135>

decklink: reset calculation of gst_decklink_video_src_update_time_mapping on no_signal. When the HDMI cable was disconnected for a long time, the calculation took too much time. SDI cable works fine.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3132>

ci: Fix website regen on push

Don't make the integrate stage manual, we need it to regen the website

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3134>

aggregator: fix input buffering

We need to be able to buffer at least the aggregator latency +
upstream latency, which is the value used to compute the aggregator
deadline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3138>

Meson: Fix libxml2 fallback

The variable xml2lib_dep does not exist. The correct name is already in
the wrap file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3136>

rtspsrc: Retry SETUP with non-compliant URL resolution on "Bad Request" and "Not found"

Various RTSP servers/cameras assume base and control URL to be simply
appended instead of being resolved according to the relative URL
resolution algorithm as mandated by the RTSP specification.

To work around this, try using such a non-compliant control URL if the
server didn't like the URL used in the first SETUP request.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1447
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/922

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3127>

webrtcbin: Fix pointer dereference before null check

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3129>

decklink: Add new persistent-id property and sort devices by persistent ID

The order of the devices iterator from the SDK is undefined and can
randomly change.

Keep the device-number property for backwards compatibility and
simplicity but prefer the persistent-id property and also use it for the
device provider implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3078>

decodebin3: fix mutex leaks

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3126>

transcodebin: Implement support for upstream stream selection

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3128>

decodebin3: Do not try to plug a decoder on raw formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3123>

videosink: Don't return unknown end-time from get_times()

... in case of reverse playback. Otherwise basesink will not
wait for clock

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3121>