rtpssrcdemux: Remove pads and reset the element also in READY->NULL

Mostly for completeness.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>

rtpptdemux: Remove pads also in PAUSED->READY

They're based on per-stream information and that should be reset
whenever going to READY state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>

docs: update plugins cache for vp9enc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

vpx: add enum for adaptive quantization modes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

vp9enc: expose frame-parallel-decoding property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

vp9enc: expose aq-mode property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

multiudpsink: Fix broken SO_SNDBUF get/set on Windows

SO_SNDBUF has been undefined on Windows because of missing WinSock2.h
include. And don't use native socket functions (e.g., setsockopt())
if code is expected to be built on Windows. We don't link ws2_32.lib
for this plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1016>

rtpmanager: Access GstRTPHdrExt fields through accessor

This way, the implementation can be private.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1017>

qtdemux: Refuse seeks in BYTES format

If downstream tries to seek in BYTES format, don't pass that through
to upstream. The byte positions downstream requests won't make any
sense in the muxed stream. There might be other formats we want to
pass through to upstream, but BYTES is not one of them. If we get a
seeking query about BYTES format, refuse that too.

This fixes a situation where we're playing a fragmented mp4 over http
and qtdemux refuses the initial seek (in TIME format), but then
h264parse/baseparse send a seek in BYTES format and everything falls
apart.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1014>

rtph265depay: update codec_data in caps regardless of format

Updating of codec_data in the caps is important to propagate changes
in sps/pps/vps via NALs. Without this, downstream does not renegotiate
when upstream changes resolution.

The comment referring to rtph264pay is from 2015 and is out of date.
rtph264pay stopped doing that in 2017 with commit
dabeed52a995d27e16eba9e4617e61eb0bcd44c4

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1011>

doc: update gst_plugins_cache.json

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1006>

qtitem: don't potentially leak a large number of buffers

The only other place where these queued buffers are removed, is in
setCaps() but that is not called at all on shutdown so this list of
buffers could not be removed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1004>

jpegenc: Remove arbitrary encoding size limitation

The encoder is happy to encode with sizes less then 16x16, so remove this
arbitrary limitation. This also fixes the fact the sink and src template caps
disagree.

Fixes #888

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/998>

qtdemux: use g_memdup2() as g_memdup() is deprecated

- atom nodes/bytereader sizes are already checked
- palettes: are fixed/known size

g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.

Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>

matroskademux: use g_memdup2() as g_memdup() is deprecated

- ebml-read: add some sanity checks when going from 64-bit
  to 32-bit length
- matroska-ids: codec_data_size has been checked via
  gst_ebml_read_binary(), is existing allocation.
- matroska-demux: alloc size is from existing allocations

g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.

Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>

Use g_memdup2() where available and add fallback for older GLib versions

- png: alloc size variable is a png type that's always 32-bit
- vpx: alloc size based on existing allocation
- wavpack: alloc size based on existing allocation
- icles: gdkpixbufoverlay: trusted and hard-coded input data
- rtp tests: rtp-payloading, vp8, vp9, h264, h265: trusted and/or static input data

g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.

Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>

Back to development

Release 1.19.1

rtpjpegpay: fix image corruption when compiled with MSVC on Windows

On Windows with MSVC, jpeg_header_size would end up 2 bytes larger
than it should be. This then leads to the first 2 bytes of the
actual jpeg image data to be dropped, because we think those
belong to the header, which results in an undecodable image when
reconstructed in the depayloader.

What happens is that when the compiler evaluates

  jpeg_header_size = mem.offset + read_u16_and_inc_offset_by_2(&mem);

it actually uses the mem.offset value after it has been increased
by the function call on the right hand size of the equation.

From section 6.5 of the C99 spec:

  3. The grouping of operators and operands is indicated by the syntax [74].
     Except as specified later (for the function-call (), &&, ||, ?:, and
     comma operators), the order of evaluation of subexpressions and the
     order in which side effects take place are both unspecified.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/889

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/999>

v4l2videoenc: Set default latency if the frame duration is invalid

If the duration of the v4l2object is invalid, use default 25fps instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/996>

deinterlace: Drop "field-order" field while transforming caps

Like other basetransform subclasses are doing, drop field
which can be converted by deinterlace.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>

deinterlace: Drop field-order field if outputting progressive

Progressive with field-order doesn't make sense

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>

rtpssrcdemux: fix "data flow before segment event" crash

This crash could happen at any time a RTP and RTCP buffer arrived
simultaneously in ssrcdemux.

The problem was that sticky-event arriving while the rtp and rtcp pads
were being set up could arrive just too late to be included in the initial
forwarding.

The fix checks if the stickies have been sent on the srcpad about to be
pushed on, and if not sends them. It also blocks any stickes from
being forwarded *prior* to this happening, to avoid them arriving on
the srcpad multiple times.

Since the test loops 1000 times, this will make running under valgrind
take forever, so use the RUNNING_ON_VALGRIND variable to detect we
are running under valgrind, and reduce the loop-count to 2 in that case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>

rtpssrcdemux: refactor destruction of GstRtpSsrcDemuxPads

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>

rtpssrcdemux: make naming consistent

Use plural for GstRtpSsrcDemuxPads, since it contains two pads, and
use the variable-name 'dpads' everywhere.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>

wavparse: use g_strndup() for copying text data

So we don't rely on NUL terminators inside the data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>

wavparse: clean up adtl/note/labl chunk parsing

We were passing the size of the adtl chunk to the note/labl
sub-chunk parsing function, which means we may memdup lots of
data after the chunk string's NUL terminator that doesn't
really belong to it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>

wavparse: guard against overflow when comparing chunk sizes

Could be rewritten as lsize > (size - 8) a well, but the
extra check seems clearer. Doesn't look like it was problematic,
lsize wasn't actually used when parsing the sub-chunks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>

doc: update gst_plugins_cache.json

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/991>

matroskademux: fix decoder glitches with H264 content

To avoid decoder starvation causing glitches on screen,
the demuxer shall clip only when the buffer is a key frame
and the lace time is greater than the stop time.

Fixes gst-editing-services#128

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/973>

qml: don't use buffers that have invalid contents

If the GL context is not shareable, ignore it.

A future change may also not output the relevant output either.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>

qml: also use the dummy texture when no buffer has been set

Fixes corrupted texture output when changing OpenGL display/contexts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>

doc: Update cache for RGBP format addition

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

matroskademux: Advertise codec-alpha in caps

This will be used to select the appropriate decoders. We also only attach the
GstVideoCodecAlphaMeta if the AlphaMode element is set, this is to stay on the
safe side and mimic what browsers (verified in Firefox and Chromium code) do.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

matroskademux: Store alpha stream in VideoCodecAlphaMeta

This generalize the feature over using mini object quark data. If
that feature was Matroska specifc, using the new CustomMeta would have
been enough and arguably cleaner then QData, though it seems that
similar technique is use with AV1 Image Format (AVIF).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

matroska-demux: extract VP8 alpha from BlockAdditionals

And put it on buffers as qdata (which is easier in this
case than a private custom meta because it can be picked
up easily in other modules).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

qt: return a different GstGLDisplay object when the first sink requests

This allows the 'replace-gstreamer-opengl-context' context machinery to
correctly replace the OpenGL context used by the pipeline when the first
qmlglsink is added to the pipeline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/982>

udpsrc: Plug leaks of saddr in error cases

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>

udpsrc: Whitespace

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>

deinterlace: Plug a method subobject leak

Changing the method would leak the previous method.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/976>

vp9enc: Add color range support

When setting the colorspace, we now clear the range to reduced range,
the default, and then we also set the range so the VP9 encoder encodes
the right information in the bitstream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>

vp9enc: Move colorspace configuration in VP9 enc

This is not supported by VP8 and was causing a warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>

vpxdenc: Add a GstVideoCodecState to configure_encoder virtual

This will be needed to configure the VP9 specific colorimetry, which is
currently configured for VP8 casing warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>

gtk: Remove coordinates double-translation

Remove our own translation in the mouse event capture code, as that
translation will be done through the navigation interface.

Tested by resizing the window created by:
gst-launch-1.0 -v videotestsrc ! navigationtest ! glupload ! glcolorconvert ! tee name=t  ! gtkglsink
and checking that the cursor follows the mouse as expected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>

gtk: Translate navigation events coordinates

If the application passed down some pointer coordinates, translate those
from display coordinates to stream coordinates, so things work as
expected even if the video is resized.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>

gtk: Export _display_size_to_stream_size()

Export _display_size_to_stream_size() so that GstNavigation implementors
can translate from display coordinates to stream coordinates before
pushing the events upstream to the DVD source, for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>

matroska-mux: Change accepted caps width and height from [16, MAX] to [1, MAX]

There are cases where the video size might be less than 16x16.
This change allows the Matroska muxer to accept this cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/539>

Use gst_element_request_pad_simple

Instead of the deprecated gst_element_get_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/958>

qtmux: Make sure to write 64-bit STCO table when needed.

qtmux attempts to choose between writing a 32-bit stco chunk offset table
when it can, but switch to a 64-bit co64 table when file offsets go over
4GB.

This patch fixes a problem where the atom handling code was checking
mdat-relative offsets instead of the final file offset (computed by
adding the mdat position plus the mdat-relative offset) - leading to
problems where files with a size between 4GB and 4GB+offset-of-the-mdat
would write incorrect STCO tables with some samples having truncated
32-bit offsets.

Smaller files write STCO correctly, larger files would switch to
co64 and also output correctly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/970>

v4l2object: Add interlace-mode back to caps for camera

skip_try_fmt_probes is set to TRUE for v4l2src to skip interlace-mode and
colorimetry when probe caps. gst_v4l2_object_set_format_full() will add
colorimetry back to caps when iterating over the negotiated caps. There is
one case that v4l2src is first in preview state then starts recording.
v4l2src caps will change with an additional interlace-mode structure after
renegotiation, then v4l2src needs to reset. But this camera driver can't
orphan buffer pool, it causes require buffer failed as streaming is still
in active state.

To fix this, also need to add interlace-mode back to caps for camera to
avoid reset.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/961>

rtpopuspay: set MARKER flag

Set MARKER flag on first buffer after DTX.

According to RFC 3551 section 4.1 the marker bit needs to be set on
"the first packet after a silence period during which packets have
not been transmitted contiguously".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>

rtpopuspay: add DTX support

If enabled, the payloader won't transmit empty frames.

Can be tested using:
  opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>

taglib: Update createFrame() to non-deprecated version.

ID3v2::FrameFactory::createFrame() versions not taking a Header have
been deprecated since v1.5 (Jan 2008).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/965>

rtpjitterbuffer: fix divide-by-zero

The estimated packet-duration can sometimes end up as zero, and dividing
by that is never a good idea...

The test reproduces the scenario, and the fix is easy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/966>

rtpjitterbuffer: clean up and improve missing packets handling

* Try to make variable and function names more clear.
* Add plenty of comments describing the logic step-by-step.
* Improve the logging around this, making the logs easier to read and
  understand when debugging these issues.

* Revise the logic of packets that are actually beyond saving in doing
  the following:
1. Do an optimistic estimation of which packets can still arrive.
2. Based on this, find which packets (and duration) are now hopelessly
   lost.
3. Issue an immediate lost-event for the hopelessly lost and then add
   lost/rtx timers for the ones we still hope to save, meaning that if
   they are to arrive, they will not be discarded.

* Revise the use of rtx-delay:
  Earlier the rtx-delay would vary, depending on the pts of the latest
  packet and the estimated pts of the packet it being issued a RTX for,
  but now that we aim to estimate the PTS of the missing packet accurately,
  the RTX delay should remain the same for all packets.
  Meaning: If the packet have a PTS of X, the delay in asked for a RTX
  for this packet is always a constant X + delay, not a variable one.

* Finally ensure that the chaotic "check-for-stall" tests uses timestamps
  that starts from 0 to make them easier to debug.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/952>

level: make properties thread-safe

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>

level: disable passthrough when audio-level-meta is enabled

Ensure we receive a writable buffer to add the meta.

Fix #878

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>

matroskamux: Don't pass a non-GObject pointer to GST_DEBUG_OBJECT and similar

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/964>

rtpjitterbuffer: Avoid generation of invalid timestamps

When updating timestamps and timer timeouts with a new offset, make sure that
the resulting value is valid (and not a negative (signed) value which ends up in
a massive (unsigned) value).

Fixes #571

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/960>

v4l2: Fix glib warning emitted when attribute query fails

The v4l2object is not a GstObject. Logging has to go through its dbg_obj
specially meant for this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/959>

qmloverlay: Use first found GstGLVideoItem as widget property

GstGLVideoItem is required to render input video in the overlay's qml.
And currently qmlgloverlay requires to set this GstGLVideoItem to its widget property.

Instead of fetching GstGLVideoItem from the overlay's root object (root-item prop),
and setting it back as a widget (widget prop),
proposing to use found GstGLVideoItem in the current object hierarchy (passed in qml-scene) by default.

Also useful in Python, which solves the issue when casting gpointer <=> QQuickItem* is required.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/919>

v4l2: fix debug category initialisation again

Would spew warnings on the rpi4 when calling into
gst_v4l2_object_get_codec_caps() from the probe_and_register()
function since the v4l2_debug category initialisation would
only be done later as part of the element/device provider
registration.

Also log things in the probe function to the v4l2 category
instead of the default category while we're at it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/956>

rtspsrc: Fix race saving seek event seqnum.

We need to save the seek seqnum before the flush stop event
since that will start the basesrc task which may send the segment
event before we're ready.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/954>

qmlglsink: allow to set force-aspect-ratio property

Add the forceAspectRatio Q_PROPERTY to allow changing the aspect ratio
from QML code as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/937>

v4l2src: fix spurious SOURCE_CHANGED error-level log messages

They're harmless, and some drivers at least return EINVAL
instead of ENOTTY for unsupported events (here: uvcvideo).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/955>

rtpsbcpay: remove use of packed struct for payload

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/950>

dtmf: convert to bit accessors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/949>

rtspsrc: Remove some dead code

stop is not used after this point, nor do we create a new segment
here since 84725d62b57bc74ce34abde755f35bf8f948f94d

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>

rtspsrc: Do not overwrite the known duration after a seek

This breaks the duration query and also the seeking query.

Broke in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>

rtspsrc: Just assign the segment instead of memcpy

Assignments copy by value, we don't need to memcpy...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>

rtpjitterbuffer: Check srcresult before waiting on the condition variable too

It might've been set to FLUSHING between the last check and the waiting,
and in that case we'd be waiting here forever now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/944>

rtp: fix test_twcc_header_and_run to support big endian.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>

rtp: fix rtptwcc to support big endian.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>

rtp: fix rtphdrextrfc6464 to support big endian.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>

tests: Fix alpha test on big endian machines.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/943>

qtmux: Protect against writing absurd sample durations

If the input DTS goes backward or is missing, the calculated
sample duration goes negative and wraps around to a very big
number. In that case, just write a sample with a duration of
0 and hope the problem is transient.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/914>

rtspsrc: De-dup seek event seqnums to avoid multiple seeks

Seek events are sent upstream on each sink, so if we receive multiple
seeks with the same seqnum, we must only perform one seek, not N seeks
where N = the number of sinks in the pipeline connected to rtspsrc.

This is the same thing done by demuxers like qtdemux or matrsokademux.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/938>

rtspsrc: Using multicast UDP has no relation to seekability

The transport has no relation to whether a media can be seeked. The
range response having a duration is the correct thing to check for.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>

rtspsrc: Add more logging for range parsing and seekable

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>

videocrop: Update documentation cache

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

videocrop: Add support for GBR* video formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

videocrop: Added support for planar pixel formats > 8bits

- Added support for planar pixel formats with depths greater than 8bits
  to transform_planar implementation
- Added a whole lot of new pixel formats to the support-list

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

videocrop: Move supported format list into private header

- Moved declaration of supported pixel formats to private header, which
  can be shared between videocrop and aspectvideocrop

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

rtpjitterbuffer: More logging when calculating rfc7273 timestamps

This code can be fragile, since it is very exacting in the timestamps
that it will accept. Add more logging so it's easier to debug issues
and figure out whether it's a bug in the calculation or something
wrong in the incoming buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/934>

rtp: missing debug init after element splitting

- h264depay
- h265depay
- sv3vdepay

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/936>

rtp: rename gst_rtp_sbc_pay_flush_buffers()

gst_rtp_sbc_pay_flush_buffers() is a misleading name. A better name would
be gst_rtp_sbc_pay_drain_buffers(), because that's what it does, it drains
any leftover queued data and pushes it downstream. "Flushing" in GStreamer
typically means to throw away any queued data and not process/push it
downstream.

Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>

rtp: fix adapter flushing in sbc payloader

GstAdapter must be flushed in some cases (flush, new segment, state change)
Without it, it may, for example, push some leftover buffer from old
segment in new segment. This, in general, breaks timestamps.
See GstAdapter documentation for more.

Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>

vpxenc: add colorspace information into VP9 bitstream

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/871>

v4l2object: Use default colorimetry if that in caps is unknown

Some streams have unknown colorimetry in caps, but v4l2object sets
default values for each primaries. It will cause check colorimetry
fail when do gst_v4l2_video_colorimetry_matches().

To fix this, need to keep the unknown colorimetry in caps same as
the default value set by v4l2object.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/923>

matroskademux: Take segment stop into account when need_segment

Otherwise, in the case of e.g. a deferred seek event, the segment stop
would be replaced with GST_CLOCK_TIME_NONE.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/929>

gstqtoverlay: Add initialization and finalization to qml-scene prop

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/924>

qt: fix build warning with clang and c-linkage of user defined type

In file included from ../subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:17:
../subprojects/gst-plugins-good/ext/qt/gstqtglutility.h:35:16: error: 'qt_opengl_native_context_from_gst_gl_context' has C-linkage specified, but returns user-defined type 'QVariant' which is incompatible with C [-Werror,-Wreturn-type-c-linkage]
QVariant       qt_opengl_native_context_from_gst_gl_context     (GstGLContext * context);

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/928>

qt: hotfix: allow per feature registration

Fixes #869

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/925>

twolame: allow per feature registration

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

shout2: allow per feature registration

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

cairo: allow per feature registration

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

y4m: allow per feature registration

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

wavparse: allow per feature registration

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

wavenc: allow per feature registration

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

spectrum: allow per feature registration

Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>