when playing some codec such as matroska with vp9 codec,
demuxer will save information like video_mastering_display_info
and video_content_light_level in caps that decoder need,
v4l2videodecoder can use it by calling V4L2_CTRL_CLASS_COLORIMETRY
ioctl.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4403>
If decoder notify a source change event when the capture format is
changed, not the resolution changed.
then gst_v4l2_object_acquire_format will retuen false due to
unsupported format.
we need to clear the format lists in the source change flow,
and reenumerate format list
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5218>
This elements pass RTP packets along unchanged and appear as a RTP
payloader element.
This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
Imported dmabuf are not being duped, so they should never be closed. Instead,
we ensure their live time by having strong reference on their original
buffer. This should fix potential flickering due to dmabuf being closed
too early.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5101>
Now that we can split GStreamer buffers over multiple v4l2 buffer, we may
endup waiting for these buffers to be processed. Avoid waiting for any of
the parts being processed. As a side effect, the pool will now try to
grow if the number of buffers is not sufficient, and will fail
otherwise.
This fixes a hang if the very first frame did not fit. In this case, the
driver will retrain that buffer until the capture is setup, but
GStreamer won't setup the capture until process() function have
returned.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5100
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5143>
Fix warnings from bindings changes in various plugin
examples
Fix the python mixer plugin by ensuring that PIL
is not holding a reference to mapped GstBuffer memory.
Port the filesrc example from old_examples
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5187>
The hack enforcing strictly increasing timestamps was, according to the
code comments, because librtmp was confused with backwards timestamps.
rtmp2sink is not using librtmp as rtmpsink did, so this is no longer
required.
Also changing the timestamps is causing audio glitches when streaming to
Youtube.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5212>
Add gst_audio_ring_buffer_set_errored() that will mark the
ringbuffer as errored only if it is currently started or paused,
so gst_audio_ringbuffer_stop() can be sure that the error
state means that the ringbuffer was started and needs stop called.
Fixes a crash with osxaudiosrc if the source element posts
an error, because the ringbuffer would not get stopped and CoreAudio
would continue trying to do callbacks.
Also, anywhere that modifies the ringbuffer state, make sure to
use atomic operations, to guarantee their visibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5205>
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.
By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:
http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc
This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
By default, macOS attempts to run lldb against a misbehaving process to handle the crash. This does not play well
with the SISEGV/SIGQUIT handler we add in gst-launch/gst-validate. The 'spinning' mechanism causes the lldb
and debugserver processes ran by macOS to misbehave, taking 100% CPU and rendering both themselves and the GStreamer
instance frozen and very hard to effectively kill. macOS's Activity Monitor is also unusable while this is happening.
This patch takes the quickest possible solution of just disabling those signal handlers entirely on macOS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5190>
Adds gst_queue_array_sort for sorting and gst_queue_array_push_sorted{,struct} for pushing in a sorted order.
All three functions accept a comparison GCompareDataFunc along with optional user_data to pass to it.
In gst_queue_array_sort a small workaround was needed to correctly sort non-struct arrays. Like what _find() already
does, we need to dereference our pointers first, to make sure we can use the same comparison functions everywhere.
This is done via a small wrapper around the provided comparison function.
The array can also wrap around (tail ends up 'before' the head), in which case we have to reorder the array (similar to
what do_expand() does) to then be able to use an existing sorting function, like g_qsort_with_data().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5112>
If a depayloader aggregates multiple RTP buffers into one buffer only
the last RTP buffer was checked for header extensions. Now the
depayloader remembers all RTP packets pushed before a output buffer is
pushed and checks all RTP buffers for header extensions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
Don't call wait_event() at all for gap events, as basesink will
end up waiting for the time that the gap event would be rendered
out at the audio device. There's no need to render it at all,
just treat it as a handy point to resync the audio if needed,
let the ringbuffer render silence, and place the next buffer
into the ringbuffer where it belongs.
The only thing we really need to do is make sure the ringbuffer
and clock are running, and wait for preroll.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2749
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5178>
Adding cudaipc{src,sink} element for CUDA IPC support.
Implementation note:
* For the communication between end points, Win32 named-pipe
and unix domain socket will be used on Windows and Linux respectively.
* cudaipcsink behaves as a server, and all GPU resources will be owned by
the server process and exported for other processes, then cudaipcsrc
(client) will import each exported handle.
* User can select IPC mode via "ipc-mode" property of cudaipcsink.
There are two IPC mode, one is "legacy" which uses legacy CUDA IPC
method and the other is "mmap" which uses CUDA virtual memory API
with OS's resource handle sharing method such as DuplicateHandle()
on Windows. The "mmap" mode might be better than "legacy" in terms
of stability since it relies on OS's resource management but
it would consume more GPU memory than "legacy" mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4510>
If glyphrun unit is changed in a single line, there could be
overlapped background area which result in drawing background
twice. Adding geometry combine so that background geometry objects
with the same color can be merged and rendered at once
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5179>
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2900
The `reports` list was being copied as a reference, therefore, copies of
a test ended up inadvertedly sharing the same list of reports. Reports
added by one instance of the test would be reflected in all instances.
This caused a race condition where, if a test was run on repeat with
gst-validate-launcher -f, very often wrong log file was shown to the
user. For instance, gst-validate-launcher would say "test failed, see
log for iteration7", but iteration7 would contain "TEST PASSED".
Worse, the runner would add the report to that incorrect log file,
mixing problems between different executions of the tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5177>
Latest MSYS2 MinGW provides these now, so we don't need to define them
if they're already present in the header.
The AudioClient3 GUID requires the Windows 10 SDK, so it's only
available in the latest MinGW, and the MinGW in Cerbero is too old.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5155>
VA decoders implementation has been verified from 1.18 through 1.22
development cycles and also via the Fluster test framework. Similar
to other cases, we can prefer hardware over software in most cases.
At the same time, GStreamer-VAAPI decoders are demoted to NONE to
avoid collisions. The first step to their deprecation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2312>
These 10bit formats are identical to NV12_16L32S, but 64bytes of data is being
prefixed with 16bytes data with four pixels of lower 2bits per byte. For
MT2110T, the lower two bits set so each bytes contains a column of 4 pixels,
also describe as tiled lower 2 bits. MT2110T has been chosen as a name to match
the vendor chosen name. This format is unlikely to exist for other vendors.
For MT2110R, the 2 low bits are in raster order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3444>
When advancing the ringbuffer, store the processed CoreAudio sample
time, then interpolate the clock in the _get_delay() calls to smooth
the clock. CoreAudio's "latency" report is always a constant and
otherwise leads to the clock generating a latency-time staircase.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>
Set the BufferFrame size in CoreAudio so it will deliver data
in ringbuffer segment units when recording. Otherwise, CoreAudio
will provide data in whatever granularity it wants, with no
relationship to the requested latency-time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>
The current limit is `x10`, which allows just `+20 dB` of gain.
While it may seem sufficient, this came up as a problem
in a real-world, non-specially-engineered situation,
in strawberry's EBU R 128 loudness normalization.
(https://github.com/strawberrymusicplayer/strawberry/pull/1216)
There is an audio track (that was not intentionally engineered that way),
that has integrated loudness of `-38 LUFS`,
and if we want to normalize it's loudness to e.g. `-16 LUFS`,
which is a very reasonable thing to do,
we need to apply gain of `+22 dB`,
which is larger than `+20 dB`, and we fail...
I think it should allow at least `+96 dB` of gain,
and therefore should be at `10^(96/20) ~= 63096`.
But, i don't see why we need to put any specific restriction
on that parameter in the first place, other than the fact
that the fixed-point multiplication scheme does not support volume
larger than 15x-ish.
So let's just implement a floating-point fall-back path
that does not involve fixed-point multiplication
and lift the restriction altogether?
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5063>
Issue was that Qt was caching multiple different shadersbased on a static
variable location. This static variable location was the same no matter
the input video format and so if an item had an input video format of
RGB and another of RGBA, they would eventually end up using the same
GL shader leading to incorrect colours.
Fix by providing different static variable locations for each of the
different shaders that will be produced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5145>
