audiobuffersplit: Add support for specifying output buffer size

Currently for buffer splitting only output duration can be specified.
Allow specifying a buffer size in bytes for splitting.

Consider a use case of the below pipeline
appsrc ! rptL16pay ! capsfilter ! rtpbin ! udpsink

Maintaining MTU for RTP transfer is desirable but in a scenario
where the buffers being pushed to appsrc do not adhere to this,
an audiobuffersplit element placed between appsrc and rtpL16pay
with output buffer size specified considering the MTU can help
mitigate this.

While rtpL16pay already has a MTU setting, in case of where an
incoming buffer has a size close to MTU, for eg. with a MTU of
1280, a buffer of size 1276 bytes would be split into two buffers,
one of 1268 and other of 8 bytes considering RTP header size of
12 bytes. Putting audiobuffersplit between appsrc and rtpL16pay
can take care of this.

While buffer duration could still be used being able to specify
the size in bytes is helpful here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1578>

wpe: Use proper callback for TLS errors signal handling

The load-failed and load-failed-with-tls-errors signals expect distinct callback
signatures.

Fixes #1388

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1566>

va: h264dec: add documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1585>

msdk: enable GPUCopy

Note it works for system memory only

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/910>

msdk: call MFXInitEx instead of MFXInit

MFXInitEx has more control than MFXInit. The current setting in this
commit is identical to MFXInit

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/910>

va: decoder: render picture only if data

Call vaRenderPicture() only if buffer or slice data is available.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1584>

va: decoder: warn if decode fails on a surface

Instead of logging error if a step fails, it logs a warning message
reducing the noise and obeying the rule for errors since the program
doesn't end

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1584>

va: h264dec: check is pad has fixed caps at caps query

Otherwise it will always reply with the possible driver caps, which
generates problems with Web MSE players.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1583>

va: h264dec: Don't need to set pts/dts/duration on outputting frame

It will be handled by videodecoder baseclass

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1582>

webrtcbin: Accept end-of-candidate pass it to libnice

libnice now supports the concept of end-of-candidate, so use the API
for it. This also means that if you don't do that, the webrtcbin will
never declared the connection as failed.

This requires bumping the dependency to libnice 0.1.16

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1139>

kmssink: Do not source using padded width/height

The width/height from the video meta can be padded width, height. But when
sourcing from padded buffer, we only want to use the valid pixels. This
rectangle is from the crop meta, orther it is deduces from the caps. The width
and height from the caps is save in the parent class, use these instead of the
GstVideoInfo when settting the src rectangle.

This fixes an issue with 1080p video displaying repeated or green at the
padded bottom 8 lines (seen with v4l2codecs).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1580>

d3d11: Add support for packed 8bits 4:2:2 YUV formats

Note that newly added formats (YUY2, UYVY, and VYUY) are not supported
render target view formats. So such formats can be only input of d3d11convert
or d3d11videosink. Another note is that YUY2 format is a very common
format for hardware en/decoders on Windows.

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

webrtcbin: Merge the RTX SSRCs from all transceivers when bundling

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1545>

avtp: avtpaafdepay: fix crash when building caps

gst_caps_new_simple gets wrong types for rate and channel which
may lead to a crash.

As 64-bit values for rate, depth, format, channels does not
make much sense and since any other functionality in gstreamer
expects G_TYPE_INT for channels and rate, we should stick to that

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1576>

h265parse: recognize more HEVC extension streams

There are streams which have the right general_profile_idc and
general_profile_compatibility_flag, but don't have the right extension
flags. We may try to use chroma_format_idc and bit_depth to
recognize these streams.

e.g.
https://www.itu.int/wftp3/av-arch/jctvc-site/bitstream_exchange/draft_conformance/SCC/IBF_Disabled_A_MediaTek_2.zip

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1328>

h265parser: select the right profile for high throughput SCC stream

Currently screen-extended-high-throughput-444 is recognized as
screen-extended-main-444, screen-extended-high-throughput-444-10 is
recognized as screen-extended-main-444-10 because they have the same
extension flags, so without this patch, it is possible that a decoder
which supports SCC but doesn't support throughput SCC will try to decode
a throughput SCC stream.

e.g.
https://www.itu.int/wftp3/av-arch/jctvc-site/bitstream_exchange/draft_conformance/SCC/HT_A_SCC_Apple_2.zip

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1328>

msdk: vpp: fixup passthrough checking for DMA

I think it is just a typo from e1a90f1ec9
msdkvpp: Disable passthrough if memory capsfeature changes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1559>

waylandsink: Use memfd_create() when available

This (so-far) Linux- and FreeBSD-only API lets users create file
descriptors purely in memory, without any backing file on the filesystem
and the race condition which could ensue when unlink()ing it.

It also allows seals to be placed on the file, ensuring to every other
process that we won’t be allowed to shrink the contents, potentially
causing a SIGBUS when they try reading it.

This patch is best viewed with the -w option of git log -p.

It is an almost exact copy of Wayland commit
6908c8c85a2e33e5654f64a55cd4f847bf385cae, see
https://gitlab.freedesktop.org/wayland/wayland/merge_requests/4

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1577>

codecs: vp9decoder: Relying on upstream vp9parse for super frame handling

By this way, we can simplify the decoding flow. Moreover, we don't
need to worry about the case where multiple visible-frames are
composed in one super-frame, since upstream vp9parse will split
them per frame unit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1567>

codecs: vp9decoder: Remove unused pts variable

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1567>

d3d11vp9dec: Don't need to consider output_picture() call without GstVideoCodecFrame

Baseclass will be updated in order to ensure GstVideoCodecFrame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1567>

d3d11vp9dec: Specify profile and alignment on sink template

Set supported profile(s) on sink template caps, so that decodebin
can filter out this element if profile of given vp9 stream is not
supported by hardware decoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1567>

d3d11decoder: Cleanup code

* Don't need to set pts/dts/duration on output buffer of frame.
  it's handled by baseclass
* Remove meaningless debug output

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1567>

codecs: h264decoder: Calculate and set latency

Add gst_h264_decoder_set_latency(), which calculates and sets
latency on base decoder class, after new_sequence is called.

This assumes that in new_sequence() vmethod, callee negotiates
downstream caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1531>

Revert "va: h264dec: set latency"

This reverts commit 3aedef4c8601dcafb065d8095a927f1cd528056f.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1531>

validate: plug leak in gssdp

These are triggered by the webrtcbin tests

https://gitlab.gnome.org/GNOME/gssdp/-/issues/10

tsdemux: Parse Audio Preselection Descriptor

For Dolby AC4 audio experience, parsing PMTs/APD from transport stream layer for all available presentations.
Refer to ETSI EN 300 468 V1.16.1 (2019-05)

1. 6.4.1 Audio preselection descriptor
2. Table M.1: Mapping of codec specific values to the audio preselection descriptor

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1555>

tsdemux: Add new API for fetching extended descriptors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1555>

webrtc: propagate more errors through the promise

Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>

meson: Do not warn when a windowing system is not found

Error out when the vulkan option is enabled, and just print
a message() otherwise. This is more correct and also allows us to pass
--fatal-meson-warnings more reliably.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1391>

opencv: set opencv_dep when option is disabled (#1406)

The examples build file checks opencv_dep, so it still needs to
be set even if the option is disabled.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1406

Signed-off-by: Adam Williamson <awilliam@redhat.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1570>

openh264dec: port to new request_sync_point() API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1571>

h264decoder: Fix various typos

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1563>

v4l2slh264dec: Minor cleanup

Move few variables in their respective scope.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1563>

v4l2slh264dec: Fix B-Frame weight table

We where not setting the luma l1 weight table.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1563>

videoparsers: Add vp9parse element

Adding vp9parse element to parse various stream information such as
resolution, profile, and so on. If upstream does not provide resolution and/or
profile, this would be useful for decodebin pipeline for autoplugging
suitable decoder element depending on template caps of each decoder element.

In addition, vp9parse element supports unpacking superframe into
single frame for decoders. The vp9 superframe is a frame which consists
of multiple frames (or superframe with one frame is allowed) followed by superframe
index block. Then unpacked each frame will be considered as normal frame
by decoder. The decision for unpacking will be done by downstream element's
"alignment" caps field, which can be "super-frame" or "frame".
If downstream specifies the "alignment" as "frame",
then vp9parse element will split an incoming superframe into single frames
and the superframe index (located at the end of the superframe) data
will be discarded by vp9parse element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1041>

mfvideosrc: Set timestamp on buffer when it's captured

Capture the timestamp immediately when new frame is arrived,
instead of doing that on ::create() method. There would be
time gap between captured time and outputting time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1351>

line21enc: add remove-caption-meta property

Similar to #GstCCExtractor:remove-caption-meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>

line21enc: heavily constrain video height

We can only determine a correct placement for the CC line
with:

* height == 525 (standard NTSC, line 21 / 22)
* height == 486 (NTSC usable lines + 6 lines for VBI, line 1 / 2)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>

line21enc: add support for CDP closed caption meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>

adaptivedemux: Add another nominal bitrate fallback calculation

Some HTTP servers don't provide fragment sizes (with the Content-Length HTTP
header). In order to still figure out a nominal bitrate (for usage by queue2),
calculate on when we're done downloading a fragment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1544>

adaptivedemux: Handle invalid HTTP duration

The default BYTE DURATION basesrc query handler will return `-1` and TRUE. In
order to properly handle cases where upstream http servers didn't return a valid
Content-Length we also need to check whether it was valid when calculating
bitrates.

Avoids returning completely bogus bitrates with gogol's video streaming services

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1544>

mfvideoenc: Add support for zero-copy encoding

Add custom IMFMediaBuffer and IMF2DBuffer implementation in order to
keep track of lifecycle of Media Foundation memory object.
By this new implementation, we can pass raw memory of upstream buffer
to Media Foundation without copy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1518>

rtmp2: Replace stats queue with stats lock

Making the thread receiving the stats wait on the loop to respond was
not a good idea, as the latter can get blocked on the streaming thread.

Have get_stats read the values directly, adding a lock to ensure we
don't read garbage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1550>

rtmp2sink: add docs section with since marker on new stop-commands property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1256>

rtmp2: fix code style, update documentation cache

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1256>

rtmp2: Clean up (improve) GstRtmpStopCommands type

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1256>

rtmp2sink: handle EOS event and close stream

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1285

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1256>

mpegtsmux: Make handling of sinkpads thread-safe

Ensure we take the object lock while accessing `GstElement.sinkpads`.
Use an iterator when the code isn't simple to avoid deadlock.

When we find the best pad, take a reference so a concurrent pad
release doesn't destroy the pad before we're done with it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1553>

hlssink2: Actually release splitmuxsink's pads

It was looking at the "outer" peer of the ghost pad, not the "inner"
peer (the target).

It provided the wrong pad to gst_element_release_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1551>

mpegtsmux: Don't create streams with reserved PID

There are quite a few reserved PID in the various MPEG-TS (and derivate)
specifications which we should definitely not use. Those PID have a certain
meaning and purpose.

Furthermore, a lot of the code in the muxer implementation also makes assumption
on the purpose of streams based on their PID.

Therefore, when requesting a pad with a specific PID, make sure it is not a
restricted PID.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1561>

ci: include template from gst-ci master branch again

Back to development

Release 1.18.0

meson: dist pot file in tarball

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1558>

gst: Update for gst_video_transfer_function_*() function renaming

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1557>

tests: mpegtsmux: Test that we can manipulate pads after stop

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1552>

mpegtsmux: Keep mux usable after stop

Otherwise you cannot request new pads until after it is started again.

gst_base_ts_mux_reset with FALSE is still called in the dispose
implementation, so the muxer still gets deallocated when we actually
clean up.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1552>

msdkvp9dec: change the include header order

This fixes the compiler error when MFX_VERSION_USE_LATEST is defined in
the SDK.

/usr/include/mfx/mfxvp9.h:48:5: error: unknown type name ‘mfxExtBuffer’
     mfxExtBuffer Header;

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1549>

rtmp2: Need to unescape the userinfo before setting

This regressed in 827afa206d8c9675f2a7af402396552c2ed1df09. The same
fix was also committed to the webrtc element, but rtmp2 was missed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1547>

webrtc: Clean up the userinfo unescaping code

Continuation from 04fd705906f4bd4cf1901001484e60c21e1b3139. This is
easier to understand and also avoids two copies.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1547>

dshowsrcwrapper: Update build instructions. Add _builddir to include search path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1537>

modplug: avoid division by zero

Under some conditions, GetMaxPosition() returns zero, which should cause
position queries to fail rather than crash.

webrtc: Unescape turnserver user and password

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1530>

meson: allow the user to disable opencv

Allow the user to really disable opencv through meson (i.e.
-Dopencv=disabled).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1533>

sctp: usrsctp: increase DIAG_MSG_LEN to accomodate longer file path

Fixes "‘%s’ directive output truncated writing XX bytes into
a region of size NN [-Wformat-truncation=]" compiler warnings.

https://github.com/sctplab/usrsctp/pull/521

Fixes #1389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1540>

dshowsrcwrapper: fix typo in variable name

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1538>

vulkan: fix the enumtypes install path

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1528>

wpe: skip glbasesrc decide_allocation when non-GL caps are negotiated

Checking for GL caps features in gl_start() was done too late in case the parent
class fails to setup a working GL context. The element now determines if GL
support should be enabled during the decide-allocation query handling.

Additionally, when no GL context was found, we need to handle the element
cleanup because in that situation glbasesrc won't call gl_stop.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1376

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1532>

codecs: h264decoder: Fix possible GstH264Picture leak and small cleanup

Don't leak pictures when dpb is full unexpectedly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1527>

d3d11videosink: Fix crash caused by missing null termination

gst_caps_features_new() must be null terminated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1524>

va: allocator: remove parameter from _create_buffer_surface()

Don't the allocator to _create_buffer_surface() since it's not used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1516>

va: utils: free allocated string

and fix a memleak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1516>

va: utils: fix log category initialization

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1516>

va: h264dec: check va allocator at decide allocation

And some code cleanups

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1516>

va: decoder: improve locks for member variable access

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1516>

webrtc/datachannel: clear the error after use

Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>

webrtc/datachannel: free previous protocol/label fields

Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>

tests/webrtc: unref GBytes after use

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>

sctpdec: unref after retrieving the static pad template

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>

Release 1.17.90

webrtc/ice: resolve .local candidates internally

Requires the system's DNS resolver to support mdns resolution.

Fixes interoperablity with recent versions of chrome/firefox that
produce .local address in for local candidates.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1139

srtobject: set error when canceled waiting for a caller

To propagate error, this commit sets a reason. Otherwise, the function
caller should check if `error` is NULL when the return value is not normal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>

srtobject: fix typo, s/errorj/error

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>

docs: fix gst-docs build if opencv is not being built

The disabler in opencv_dep (retrieved via libs_doc) will
cause a meson interpreter error if opencv is not being built:

ERROR: The += operator currently only works with arrays, dicts, strings or ints
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1519>

fdkaacenc: Implement flush function

The internal fdk encoder always produces 1024 bytes even with no input,
so special care should be taken to not drain it twice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1515>

mediafoundation: Correct wrong raw video format mapping

Was a shameful mistake

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1517>

va: h264dec: set latency

The min latency is calculated with the maximum number of frames that
precede any frame, if available, and it is lower than the maximum
number of frames in DBP.

The max latency is calculated with the maxium size of frames in DBP.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1500>

fdkaacenc: Refactor layout selection code

No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1359>

fdkaacenc: Move channel layouts to gstfdkaac.c

In preparation of sharing them with the decoder. Iteration of the
channel layouts needs to be changed to use a sentinel element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1359>

vulkan: docs annotation updates

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1506>

build/vulkan: split vulkan gir

also add to docs

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1506>

wpe: WebView and WebContext handling fixes

The WPEThreaded view is now split in 2 classes:
- WPEContextThread handles the persistent WebKit thread, where all WebKit API
calls should be handled.
- WPEView: is created from the WPEContextThread. It handles the WebView and
maintains the public interface on which wpesrc relies. This is the facade for
the WebView, basically. It takes care of dispatching API calls into the context
thread.

With these fixes it is now possible to create (and reuse) mutlple wpesrc
elements during the application lifetime.

Fixes #1372

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1484>

sctp: fix build with GST_DISABLE_GST_DEBUG

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>

sctp: hook up internal copy of libusrsctp to build

Add option 'sctp-internal-usrsctp' so people can choose
to build againts the distro version instead.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>

sctp: import internal copy of usrsctp library

There are problems with global shared state and no API stability
guarantees, and we can't rely on distros shipping the fixes we
need. Both firefox and Chrome bundle their own copies too.

Imported from https://github.com/sctplab/usrsctp,
commit 547d3b46c64876c0336b9eef297fda58dbe1adaf
Date: Thu Jul 23 21:49:32 2020 +0200

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>

proxysink: event_function needs to handle the event when it is disconnecetd from proxysrc

without this a disconneted proxysink fail when goes to play with error:

 Internal data stream error.
 streaming stopped, reason error (-5)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1508>

cccombiner: Correct sink_query chain up and fix caps leaks

Don't chain up to src_query() from sink_query() method, and
returned caps by gst_static_pad_template_get_caps() needs to be
cleared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1513>

mediafoundation: Call MFShutdown when destroying plugin

MFStartup and MFShutdown should be paired as documented in
https://docs.microsoft.com/en-us/windows/win32/api/mfapi/nf-mfapi-mfstartup#remarks
Otherwise valgrind-like tools would report false positive memory leak.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1512>

msdkvpp: do not hold too many input buffers in locked list

If the surface is locked before vpp, upstream takes the reference.
We do not need to take a reference for msdk in vpp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1490>