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A classic case of not enough locking.

One interesting thing with this is the interaction between the
rotation value and caps negotiation.  i.e. the width/height of the caps
can be swapped depending on the video-direction property.  We can't lock
the entirety of the caps negotiation for obvious reasons so we need to
do something else.  This takes the approach of trying to use a single
rotation value throughout the entirety of the negotiation and then
subsequent output frame in a kind of latching sequence.

Fixes: #792
Part-of: <!836>

Part-of: <!836>

This is due to a bug in meson where it will not detect properly
the compiler if the symbols need an undercore.
https://github.com/mesonbuild/meson/issues/5482

Fixes #821

Part-of: <!845>

rtspsrc: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal

Part-of: <!842>

Might not drain correctly

Part-of: <!838>

For legacy drivers that don't implement ENUM_FRAMESIZE, use active
resolution to probe colorspace. This can improve the accuracy of the
result when the colorspace depends on the resolution. This fixes a
wrong colorspace issue on board with vendor bsp at resolution 2560x1440.

Part-of: <!830>

When reconstituting packets from a stream with variable packet
sizes, don't xor larger packets past the length of the protected
packet

Part-of: <!839>

When protecting packets with a variable payload length, we
reallocate the xored payload when needed. It is a good idea
to memset the extended memory to 0 so that we don't xor
data with garbage!

Part-of: <!839>

While the standard is a bit vague about whether the padding,
extension and marker bits should be protected:

> The usage, by senders and receivers, of the following bits shall
> be defined by the associated video/audio transport standards:

It is obviously necessary and useful for some formats (eg VP8)
that those indeed be protected.

Part-of: <!839>

Check the sequence of format-location/fragment-opened/fragment-closed
events is respected. There should be 1 format-location call for each
fragment-opened message, and 1 fragment-closed for each.

Part-of: <!833>

!798
introduced a check in the need-new-fragment logic to avoid starting a
new fragment unless there has been some data on the reference stream,
but the check is done against the number of bytes that have been
received on the input, not the number that were released for output
into the current fragment.

Fix the check to remember and test against bytes that have been sent
for output.

This also fixes a problem where starting a new fragment fails to
request a new filename from the format-location signal.

Part-of: <!833>

When posting fragment-opened and fragment-closed messages,
put a debug statement in the logs

Part-of: <!833>

Change some g_assert into element errors so that they can be
caught and the pipeline shut down.

Part-of: <!833>

Provide an implementation of the transport-wide-cc header extension and
use it in rtpfunnel.

Part-of: <!808>

fix: undefined reference to `vc_gencmd'
/usr/src/debug/gstreamer1.0-plugins-good/1.18.1-r0/build/../gst-plugins-good-1.18.1/sys/rpicamsrc/RaspiCamControl.c:1440: undefined reference to `vc_gencmd'

Part-of: <!818>

The function is renamed to be properly associated with AudioFormatInfo
(its instance) instead of AudioFormat (an unrelated enum), see [1] for
the rename itself.

[1]: gst-plugins-base!940

We need to remove x86inc.asm from the list of compiled assembly files
because it is not supposed to be compiled separately. It is directly
included by yadif.asm, and it exports no symbols.

The object file was getting ignored on all platforms except on msvc
where it was causing a linker hang when building with debugging
enabled because the object file had no debug symbols (or similar).
We've seen this before in FFmpeg too, which uses nasm:
gstreamer/meson-ports/ffmpeg!46

Part-of: <!825>

Especially considering some dynamic pipeline scenarios.

Part-of: <!822>

Makes the element reusable multiple times after a state change back to READY.
Fixes #105

Part-of: <!823>

gst_caps_simplify() will move interlace format before normal video
format. It will cause caps negotiate prefer interlaced caps which
isn't expected. Seperate normal caps and interlaced caps and then
merge it will keep prefer progress video format.
Add ARGB/BGRA for interlaced caps.

Fixes #802

Part-of <!813>

Part-of: <!813>

This will end up as a "received" packet, due to the code in
source_push_rtp, which will think this is a packet being received.

Instead drop the packet and hope that either:
1. Something upstream responds to the GstRTPCollision event and changes
   SSRC used for sending.
2. That the application responds to the "on-ssrc-collision" signal, and
   forces the sender (payloader) to change its SSRC.
3. That the BYE sent to the existing user of this SSRC will respond to
   the BYE, and that we timeout this source, so we can continue sending
   using the chosen SSRC.

The test reproduces a scenario where we previously would have sent
on a non-internal source.

Part-of: <!817>

If current_time is < collision_timeout, we get an uint64 underflow, and
the check will trigger prematurely.

Part-of: <!817>

In baseparse we set the fixed caps flag on all src pads, therefore the
source pad caps query in get_allowed_caps will return the current caps.
Current caps won't necessarily intersect with the new caps (e.g. sample
rate change). Replace get_allowed_caps with peer_query_caps.

Part-of: <!816>

timesacle -> timescale

Part-of: <!815>

Fixes #803

Part-of: <!815>

Part-of: <!757>

We don't know exactly when QML will stop using them but it should be
safe to unref them after at least 2 more buffers were bound.

Part-of: <!810>

Part-of: <!811>

Part-of: <!811>

This makes it easier to do development with MSVC by making it warn
on common issues that GCC/Clang error out for in our CI configuration.

Continuation from gst-build!223

Part-of: <!809>

This is useful in the Bundle case because there may be multiple local
and remote SSRCs in the same session.

Part-of: <!776>

Part-of: <!530>

Depayload for the iSAC audio codec.

Part-of: <!530>

Payload for the iSAC audio codec.

Part-of: <!530>

This can be useful to let the layouts automatically resize qtitem
and also easily query a video's width/height from QML

Part-of: <gstreamer/gst-plugins-good!802>

See !797

Part-of: <!801>

Each codec_data is put into its own SampleTableEntry inside the stsd.

Part-of: <gstreamer/gst-plugins-good!787>

This property will tell the element to hold
the EOS event and keep it until the next
keystroke.

Part-of: <!792>

Fix a comment in the splitmuxsrc robust muxing test so it
describes the test properly.

Part-of: <gstreamer/gst-plugins-good!798>

Add a new state for ending the overall stream, and use it to decide
whether to pass the final EOS message up the bus instead of dropping
it. Fixes a small race that makes the testsuite sometimes not generate
the last fragment(s) sometimes because the wrong EOS gets
allowed through too early.

Part-of: <gstreamer/gst-plugins-good!798>