tests: vkimage: add VK_IMAGE_USAGE_SAMPLED_BIT usage bit

Running tests with Vulkan Validation enabled show an error on vkimage tests:

Code 0 : Validation Error: [ VUID-VkImageViewCreateInfo-image-04441 ]
Object 0: VK_NULL_HANDLE, type = VK_OBJECT_TYPE_COMMAND_BUFFER; Object 1: handle
= 0x50000000005, type = VK_OBJECT_TYPE_IMAGE;
| MessageID = 0xb75da543
| Invalid usage flag for VkImage 0x50000000005[] used by vkCreateImageView(). In
this case, VkImage should have VK_IMAGE_USAGE_SAMPLED_BIT |
VK_IMAGE_USAGE_STORAGE_BIT | VK_IMAGE_USAGE_COLOR_ATTACHMENT_BIT |
VK_IMAGE_USAGE_DEPTH_STENCIL_ATTACHMENT_BIT |
VK_IMAGE_USAGE_TRANSIENT_ATTACHMENT_BIT | VK_IMAGE_USAGE_INPUT_ATTACHMENT_BIT |
VK_IMAGE_USAGE_FRAGMENT_SHADING_RATE_ATTACHMENT_BIT_KHR |
VK_IMAGE_USAGE_FRAGMENT_DENSITY_MAP_BIT_EXT |
VK_IMAGE_USAGE_VIDEO_DECODE_DST_BIT_KHR |
VK_IMAGE_USAGE_VIDEO_DECODE_DPB_BIT_KHR |
VK_IMAGE_USAGE_VIDEO_ENCODE_SRC_BIT_KHR |
VK_IMAGE_USAGE_VIDEO_ENCODE_DPB_BIT_KHR | VK_IMAGE_USAGE_SAMPLE_WEIGHT_BIT_QCOM
| VK_IMAGE_USAGE_SAMPLE_BLOCK_MATCH_BIT_QCOM set during creation.
The Vulkan spec states: image must have been created with a usage value
containing at least one of the usages defined in the valid image usage list for
image views

This patch adds VK_IMAGE_USAGE_SAMPLED_BIT to the usage bits in test.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4311>

vkfullscreenquad: set enable_clear as TRUE by default

While using the validation layer with this pipeline:

gst-launch-1.0 videotestsrc num-buffers=10 ! vulkanupload ! vulkancolorconvert ! vulkansink

The validation layer throws this message:

Code 0 : Validation Error: [ VUID-VkAttachmentDescription-format-06699 ]
Object 0: handle = 0x5555562e9610, type = VK_OBJECT_TYPE_DEVICE; | MessageID = 0x52b3229e |
vkCreateRenderPass: pCreateInfo->pAttachments[0] format is
VK_FORMAT_B8G8R8A8_UNORM and loadOp is VK_ATTACHMENT_LOAD_OP_LOAD, but
initialLayout is VK_IMAGE_LAYOUT_UNDEFINED.
The Vulkan spec states: If format includes a color or depth aspect and loadOp is
VK_ATTACHMENT_LOAD_OP_LOAD, then initialLayout must not be VK_IMAGE_LAYOUT_UNDEFINED

When creating the render pass the loadOp can be either
`VK_ATTACHMENT_LOAD_OP_CLEAR` or `VK_ATTACHMENT_LOAD_OP_LOAD` depending on
`enable_clear`. While `enable_clear` is FALSE by default (which means
`VK_ATTACHMENT_LOAD_OP_LOAD`). Nonetheless, its value is explicitly changed by
`vkoverlaycompositor` to FALSE too!

This behavior was introduced in merge request #2470 where
`VK_ATTACHMENT_LOAD_OP_CLEAR` was a fixed value for loadOp. Thus, the bug
consists in a missing initialization of `enable_clear` to TRUE from that merge
request.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4311>

vulkanswapper: correctly handle force-aspect-ratio=false

It was simply ignored so actually handle it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4290>

openjpegdec: allow multithread decoding only in subframe mode

To avoid mis-ordered frames, allow multithread decoding only in
subframe mode.

Fixes #1786

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4275>

datetime: Return G_MAXFLOAT instead of G_MAXDOUBLE for no timezone offset

Returning G_MAXDOUBLE from a function returning a float is not going to
work well and MSVC also correctly warns about this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4285>

decodebin3: fix hang issue when remove failing stream

Need mark selection_update to true when update selection,
otherwise, pipeline will not handle this selection update
sometimes when this flag has been reset

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4284>

rtpjpegdepay: fix logic error when checking if an EOI is present

We wouldn't add the missing EOI marker if the frame ended with
either 0xFF NN or 0xNN D9.

Fixes #2407

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4269>

d3d11bufferpool: Fix invalid access in debug print loop

Add missing condition check in for loop

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4258>

nvh264encoder: Fix template caps

It should include progressive as well

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4253>

qtdemux: Fix seek adjustment with SNAP_AFTER flag

With GST_SEEK_FLAG_SNAP_AFTER present, the previous version would
adjust seek time based on the keyframe farthest away from desired_time.
This was incorrect, because we always want the *earliest* suitable keyframe
to seek to, not the last one.
With this fix, in case of the SNAP_AFTER, we now look for the closest keyframe
that can be found after desired_time. Behaviour for SNAP_BEFORE should remain
unchanged.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4251>

v4l2object: mark jpeg as parsed

Assuming that V4L2 CAPTURE devices always use one buffer per JPEG image, we can
always mark JPEGs provided by a V4L2 element as parsed.

The V4L2 elements require that JPEG images sent to V4L2 OUTPUT devices must
always be parsed.

This is necessary to link a V4L2 CAPTURE device with a V4L2 OUTPUT device
without explicitly marking the stream as parsed or adding a jpegparse into the
pipeline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4247>

gtkwaylandsink: Fix crash when rendering after the window is closed

Continuation of https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4197

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4238>

parsebin: Improve elementary stream handling

The goal of parsebin is to figure out which elements to link together in order
to provide elementary streams given any random input.

The problem is that deciding whether a given stream should still have more
elements plugged in or not was dependent on ... the presence of compatible
decoders (sic).

Instead of that, if we can't plug anymore elements on a given stream *and* it is
detected as being an elementary stream, expose it.

Fixes #2118

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4237>

urisourcebin: Activate pad before transferring sticky events

Otherwise they get refused since the pad is flushing

Fixes #2384 for good

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4233>

ksdeviceprovider: Fix leak in gst_dshow_device_provider_start

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4227>

dshowdeviceprovider: Fix leak in gst_dshow_device_provider_start

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4227>

matroskademux: Make gst_byte_reader_get_data() usage less confusing

This is effectively the same behaviour but retrieving 0 bytes of data is
confusing to read.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4221>

flacenc: Fix mapping of GStreamer image tag type to FLAC image tag type

These enums are not compatible so just casting them does not work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4221>

plugins: Fix various trivial clang compiler warnings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4221>

tsdemux: Fix reading of extended Opus channel configuration

Argument evaluation order is implementation defined in C, and gcc is
evaluating right-to-left (works) while clang is evaluating left-to-right
(does not work).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4221>

h264decoder: Fix DPB bumping process

As per spec C.4.5.3 "Bumping", if bumping is needed but DPB holds
no "output needed" picture, then a picture that has the smallest
POC should be considered first for output

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4219>

urisourcebin: Propagate sticky events from parsebin

If sticky events are present on parsebin source pads, we propagate them to the
multiqueue source pads. Those will be propagated on the new urisourcebin source
pads like in the other code paths.

This ensures that STREAM_START event are present on new source pads. If CAPS
event are also present (not guaranteed), they will also be available.

Fixes #2384

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4218>

nvencoder: Fix CQP option setting

... and zero initialize LUID and CUDA device list to address
coverity issue

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4217>

webrtc: Fix segfault traversing ice transports

Previously, reassigning loop index l in nicestream.c
could cause a segfault if l->data was null, as it could
reassign l to a null variable, triggering the loop
postassignment l->next, which then segfaults due to
l now being null. It is instead moved into the loop.
_delete_transport already performs the reassignment
inline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4213>

wasapi2: Fix potential crash on device activation failure

The activation object is live in COM thread already and therefore
self refcount hack is pointless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4207>

webrtc: patch leak caused by early return

In webrtc_data_channel_send functions, both data and string,
an early return on a non-open datachannel caused it to leak
the buffer used for pushing to appsrc, meaning any buffer
sent after leaving the open state was leaked in full.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4206>

inputselector: Wake up streaming thread before PLAYING_TO_PAUSED transition

Also take object lock before iterating the pads.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1772

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4196>

d3d11videosink: Fix tearing in case of fullscreen mode

DXGI_SWAP_CHAIN_FLAG_ALLOW_TEARING and DXGI_PRESENT_ALLOW_TEARING should
be used only if required, when user wants Vsync off for example.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2376
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4188>

d3d11converter: Fix conversion backend selection

Intended behavior was selecting video processor only if pixel shader
is disabled, since we prefer shader over video processor

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4189>

avviddec: Drop decoder stream lock when calling send_packet

This is already done for every other calls to send_packet. The deadlock occures
since FFMPeg 6.0. The decoder tries to get a buffer from a thread during
the draining process, and blocks trying to get the video decoder stream lock
already heald by the drain function.

Fixes #2383

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4187>

d3d11compositor: Fix composition error on release_pad()

Composition might be still in progress while it's being released

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4186>

matroskamux: Set rate/channels in Opus template caps

For some reason these were missed, and if caps didn't have them, we would emit
an invalid Matroska file with a 0 value for Sampling Frequency or channels.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4172>

rtpopusdepay: Assume 48 kHz if sprop-maxcapturerate is missing

This matches 7587, section 6.1:

>   sprop-maxcapturerate:  a hint about the maximum input sampling rate
>      [...]
>      bandwidths (Table 1).  By default, the sender is assumed to have
>      no limitations, i.e., 48000.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4172>

zxing: add support for zxing-c++ 2.0

format is a C++ string in 2.0

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4148>

vulkanoverlaycompositor: don't do a potential use after free

Removing a meta from a buffer means one doesn't have access to it
anymore.  Instead use the already reffed composition directly.

Fixes a use-after-free in the following pipeline:

... ! vulkanupload ! timeoverlay ! vulkanoverlaycompositor ! ...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4147>

cea708overlay: fix HCR interpretation

An unsigned counter with a for (i = end; i >= 0; i--), can be optimized to
infinite loop as an unsigned value will always be >= 0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4141>

h264parse: Validate VUI framerate

A few streams contain invalid/wrong framerate in VUI. Do validate
it based on the spec and ignore invalid VUI framerate values.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1753
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4131>

jpegparse: demote and promote log messages level

Before those levels where when implementing the plugin. Now these
levels are for reporting issues from users.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4130>

jpegparse: reset parse state when the SOI is not the first marker

There may be garbage or some bits before a SOI comes in some problematic
mjpeg streams. For example, some network error may cause the EOI marker
of the previous frame lost, and when the new frame's SOI comes, we still
use the state of the last frame, which will generate errors.

For this kind of frames without EOI, if that frame already has some data
(the SOS segment is detected), we still push it as a frame with CORRUPTED
flag set. But if not, we just discard all the data before the new SOI.

Co-Authored-By: Víctor Jáquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4130>

rtspsrc: Consider "451: Parameter Not Understood" when handling broken control urls

similar to https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3854

it seems that some implementations return this when
the server does not implement URL handling correctly

this fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2334

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4129>

Back to development

Release 1.22.1

vtdec: Correctly retrieve reorder queue length from SPS

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4110>

alsasink: Fix stall when going from PLAYING to NULL (stucked at PAUSED) with uac1 gadget

This happened with a uac1 gadget which for some reason does not behave nicely.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4104>

adaptivedemux2: Fix buffering treshold initialization

Properly initialize the stream default recommended buffering threshold so that
a default (10s) value is used until the subclass can provide a proper value

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2064

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4105>

inputselector: Avoid deadlock when shutting down

Transition from PLAYING to PAUSED unschedules any pending clock wait,
but there was no guard that prevented another input buffer then waiting
again while in PAUSED before dataflow stops. Use a new `playing` flag to
avoid this, and exit the chain function if instead of playing we're now
flushing. Basically the same as the clocksync element.

Closes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1772

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4099>

validate: Protect init function with a recursive mutex

In tests in the rust bindings we end up with 2 thread initializing
concurrently, and it should not be a problem, -validate should be MT
safe.

Using a recursive mutex as we might recursively init for some reason
and we are not on the hot path here in any case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4097>

validate:scenario: sink refs when building

Scenarios are no owned by anyone, even if they are GstObjects

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4097>

avfvideosrc: Fix description and trailing whitespace

Minor fix to mention that the element is also useful on MacOS,
and remove some trailing whitespace

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4096>

avfvideosrc: Don't wait on main thread for permissions request

Recursively invoking the NSMainLoop can cause crashes in
applications that don't expect it. Instead of waiting for
permission to be granted, move the wait later - until we
actually need device permissions when starting the capture
session. That moves the wait into the streaming thread
instead of the application thread that's setting the pipeline
state to READY.

Instead of a manual state change implementation to open
and close the device, use the basesrc start/stop methods that
are intended for the purpose.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4096>

python: Use arch-specific install dir

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4093>

gldisplay: Remove unused code

The code related with environment variable `GST_GL_PLATFORM` is unused
since commit 33c60bdbf9ae2fd2e88099ad89ad836b8b78c2f5.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4091>

vtenc: Fix checking for certain CPU variants when running in VMs

These checks were introduced to prevent exposing ARGB64/RGBA64 in the caps
when running on M1 Pro/Max with macOS <13 because of a bug in VideoToolbox.
Unfortunately, the initial buffer size of 15 is too short when running
in a VM - the CPU brand string there looks like "Apple M1 Pro (Virtual)",
which due to its length causes sysctlbyname to return -1, resulting in
broken formats still showing up in the caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4082>

ffmpeg: avauddec/avviddec: Free packet side data after usage

As we don't use proper refcounting with AVPacket we have to manage this
ourselves.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2006

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4078>

tools: Make sure UTF-8 encoded command line arguments on Windows

On Windows, arguments passed in main() are system codepage
encoded and might not be valid UTF-8 string.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4072>

nvencoder: Fix b-frame encoding on Linux

On Windows, Win32 event handle is used to wait for encoded output,
but it's not available on Linux. We should delay bitstream locking
if encoder returns "need-more-input"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4008>

ci: prepare_cerbero_env: Use predictable git clone path

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4069>

meson: Switch dav1d wrap to a tarball

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4069>

h265parse: Always set profile on src caps

h265parse should provide profile for autoplugging

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4068>

buffer: fix copy meta reference debug log formatting

Fixes the following valgrind error:
==616== Conditional jump or move depends on uninitialised value(s)
==616==    at 0x4900E34: gst_debug_print_object (gstinfo.c:1143)
==616==    by 0x49010B6: gst_info_printf_pointer_extension_func (gstinfo.c:1215)
==616==    by 0x4959FDB: __gst_printf_pointer_extension_serialize (printf-extension.c:47)
==616==    by 0x495A487: printf_postprocess_args (vasnprintf.c:258)
==616==    by 0x495A52C: __gst_vasnprintf (vasnprintf.c:290)
==616==    by 0x4959F8F: __gst_vasprintf (printf.c:154)
==616==    by 0x4901C1F: gst_debug_message_get (gstinfo.c:791)
==616==    by 0x4901C75: _gst_debug_log_preamble (gstinfo.c:1431)
==616==    by 0x4903208: gst_debug_log_default (gstinfo.c:1575)
==616==    by 0x49020BA: gst_debug_log_full_valist (gstinfo.c:624)
==616==    by 0x490211D: gst_debug_log_valist (gstinfo.c:656)
==616==    by 0x49021AD: gst_debug_log (gstinfo.c:533)
==616==    by 0x48DDC11: gst_buffer_copy_into (gstbuffer.c:693)
==616==    by 0x48DF5F1: gst_buffer_copy_with_flags (gstbuffer.c:727)
==616==    by 0x48DF640: gst_buffer_copy_deep (gstbuffer.c:756)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4038>

jpegdecoder: fail early if no input caps have been provided

The jpegdecoder class does not implement the ->parse() virtual function,
and we always need to add the jpegparse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1829
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>

av1decoder: fail early if no input caps have been provided

The av1decoder class does not implement the ->parse() virtual function,
and we always need to add the av1parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>

vp9decoder: fail early if no input caps have been provided

The vp9decoder class does not implement the ->parse() virtual function,
and we always need to add the vp9parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>

vp8decoder: fail early if no input caps have been provided

The vp8decoder class does not implement the ->parse() virtual function,
it can only accepts frame aligned data. If some element such as filesrc
feed it with unaligned data, the behaviour is undecided. So we should
set_needs_format of the decoder to TRUE, then it can fail with a
"not-negotiated" error early, rather than go on and generate unexpected
error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>

mpeg2decoder: fail early if no input caps have been provided

The mpeg2decoder class does not implement the ->parse() virtual function,
and we always need to add the mpegvideoparse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>

h264decoder: fail early if no input caps have been provided

The h264decoder class does not implement the ->parse() virtual function,
and we always need to add the h264parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>

h265decoder: fail early if no input caps have been provided

The h265decoder class does not implement the ->parse() virtual function,
and we always need to add the h265parse element before it. So we should
set_needs_format of the decoder to TRUE, then if no parse before it, it
can fail with a "not-negotiated" error early, rather than go on and
generate unexpected error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4066>

curlhttpsrc: Add curl anyauth option

Add curl anyauth option to support http request to endpoints not using
only basic authentication (as default in curl). Also aligning with
curlhttpsink that already uses this option.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4065>

rtspconnection: Annotate RTSP message parameters correctly

These parameters are not actually `out` parameters but must
be allocated and zero-initialized by the calling function.
Marking them as `out caller-allocates` will cause memory
corruptions when calling these APIs from e.g., Python code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4061>

rtspsrc: Use the correct vfunc for the `push-backchannel-sample` action signal

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/446

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4056>

qtmux: Fix assertion on caps update

GstQTMuxPad.configured_caps should be protected since it's
updated from streaming thread and accessed in aggregate thread

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4055>

gstglwindow_x11.c: Fix colormap leak

This fixes an X server side memory leak, this is normally not severe
but in some circumstances where the glwindow is recreated frequently
it can result in a very bad memory leak.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4054>

d3d11overlaycompositor: fix texture width and height

The dimension of the overlay texture directly corresponds to the size of the overlay **buffer** which is given by its video meta.
The dimension at which the overlay should be displayed directly correspond to the overlay `render_width`and `render_height`.

This match the behavior of glimagesink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4053>

gst-plugins-bad: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4041>

gst-plugins-good: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4041>

gst-plugins-base: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4041>

gstreamer: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4041>

vaallocator: Check return value from va_sync_surface

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4007>

va: Fix some code defects

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4007>

gst-play: update translated string

Make it more consistent with the other option descriptions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4033>

qtmux: Implement writing of `av1C` version 1 box

Version 0 is ancient and not specified in any documents. Take it
directly from the `codec_data` if presents or otherwise try to construct
a reasonably looking `av1C` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4027>

qtdemux: Drop av1C version 0 parsing and implement version 1 parsing

The av1C box is optional so dropping parsing does not break anything
fundamentally, and there seems to be no historical record how version 0
even looks like while the comments and the parsing disagreed with each
other.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4027>

wpe: Logging fixes for the WebExtension

Using logging macros without a `GST_CAT_DEFAULT` in scope leads to critical
warnings.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4022>

appsrc: Don't chain up `BaseSrc::negotiate()`

If we have caps then we can only set exactly those caps, if we have no
caps yet then negotiating anything is not very meaningful because the
caps are defined by the application and not downstream.

Avoids, among other things, an unnecessary allocation query and spurious
useless caps being set before the first buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4020>

gst-play: Don't force accurate seeking

This crept in several years ago sadly :(

The usage of accurate seeking should be reserved to use-cases where it is
essential that we seek to that position. This should not be the default.

There is a new option `--acurate-seeks/-a` to be able to force that.

Furthermore, if accurate seeks aren't required, a player should be using the
GST_SEEK_FLAG_KEY_UNIT flag to seek to the closest keyframe and provide the most
reactive experience.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4017>

uridecodebin: Set source element to READY before querying it

Generating the source element is done when uridecodebin is doing the
READY to PAUSED state change, so it is reasonable to set the new source
element to that state.

This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4016>

ccconverter: don't debug a potentially freed filter caps

Fixes a use-after-free

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4013>

d3d11videosink: Fix rendering on external handle

Partial revert of the commit 068a5c1053ae05b3c7747243948447e764d25aa6.
That introduced size mismatch between internal and external HWND

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4010>

vah265enc: Use helper to update properties.

This is a continuation of the original patch by
 Mengkejiergeli Ba <mengkejiergeli.ba@intel.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4009>

av1parser, h265parser: Fix some code defects

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4003>

webrtc: Calculate the jitter for remote-inbound-rtp stats

Populate the clock-rate in the internal stats structure, so
it can be used by the _get_stats_from_remote_rtp_source_stats()
method to calculate remote receivers' jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3994>

webrtcbin: Report full codec-stats for source pads

Use the current caps for webrtcbin srcpads, as received_caps
are only stored for sink pads based on incoming caps events.

Makes it so that webrtcbin stats reports contain fuller
codec information.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3994>

msdkav1enc: fix the category for msdkav1enc debug

The msdkav1enc debug initialized with gst_msdkav1dec_debug,
fix this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3991>

qtdemux: Don't emit GstSegment correcting start time when in MSE mode

When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).

Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:

ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it

This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.

Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.

Co-authored by: Alicia Boya García <ntrrgc@gmail.com>

...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467

[1] https://github.com/rdkcentral/mvt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3990>

tests: Move srtp test known issue def to the 'check' testuite

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3986>

validate:launcher: try to bne more explicit about where to add known failure

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3986>

validate:launcher: Handle known return code issue in none validate based tests

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3986>

codectimestamper: Fix timestamping on sequence update

... and enhance debug logging.

Keep internal timestamp offsets on drain, no reason to reset them

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3987>

codecparsers: {h264,h265}bitwriter: Remove redundant condition checks

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3978>

validate: add elements_srtp.test_play_key_error to flaky test list

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1777

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3970>