Includes a new GstVulkanHandlePool base class for pooling different
resources togther. The descriptor cache object is ported to
GstVulkanHandlePool with the exact same functionality.
A new GstVulkanFenceCache is also implemented for caching fences
which is used internally by GstVulkanDevice for creating or reusing
fences.
The existing GstVulkanTrashFenceList object now caches trash objects.
Part 1 is a base class (vkvideofilter) that handles instance, device,
queue retrieval and holding that has been moved to the library
Part 2 is a fullscreenrenderquad that is still in the plugin that
performs all of the previous vulkan-specific functionality.
Weak refs don't quite work here correctly as there is always a race with
taking the lock between find_view() and remove_view(). If find_view()
returns a view that is going to removed by remove_view() then we have an
interesting situation.
In theory, the number and type of views for an image are relatively
constant and should not change one they've been set up which means that
it is actually practical to perform pool-like reference counting here
where the image holds a pool of different views that it can give out
as necessary.
The whole `src_read()` function is a hot loop since the ringbuffer
thread is waiting on us, and printing to the console from inside it
can easily cause us to miss our deadline.
F.ex., if you had GST_DEBUG=3 and we accidentally missed a device
period, we'd trigger the "reported glitch" warning, which would cause
us to miss another device period, and so on. Let's reduce the log
level so that GST_DEBUG=3 is more usable, and only print buffer flag
info when it's actually relevant.
Some audio drivers return varying amounts of data per ::GetBuffer
call, instead of following the device period that they've told us
about in `src_prepare()`.
Previously, we would just drop those extra buffers hoping that the
extra buffers were temporary (f.ex., a startup 'burst' of audio data).
However, it seems that some audio drivers, particularly on older
Windows versions (such as Windows 10 1703 and older) consistently
return varying amounts of data.
Use GstAdapter to smooth that out, and hope that the audio driver is
locally varying but globally periodic.
Initially reported in https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/808
We were miscalculating the device period, i.e. the number of frames
we'll get from WASAPI in each IAudioClient::GetBuffer call, due to
a calculation mistake (truncate instead of round).
For example, on my machine when the aux input is set to 44.1KHz, the
reported device period is 101587, which comes out to 447.998 frames
per ::GetBuffer call. In reality we will, of course, get 448 frames
per call, but we were truncating, so we expected 447 and were
discarding one frame every time. This led to glitching, and skew over
time.
Interestingly, I can only see this with 44.1Khz. 48Khz/96Khz are fine,
because the device period is a more 'even' number.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/806
The for loop in gst_msdkdec_handle_frame is error prone
about how it manages surfaces. Because sometimes it sets
the surface variable to NULL and sometimes it needs to free
it right away. So better to print an error if surfaces are
leaked to help with any change around the loop.
msdk plugin is not used for sofware encode/decode as there are better
solutions available. Also, with MFX_IMPL_AUTO_ANY, if software decode
is not supported, the plugin will still load, but will then fail when trying to
run the (autoplugged) pipeline. With MFX_IMPL_HARDWARE_ANY,
the plugin fails and a better software decoder is auto-plugged.
GstNvBaseEnc::n_bufs was set from the previous encoding session
but it wasn't cleared after stop. That might result to invalid memory
access at the next start (no encoded data) and then stop sequence.
Instead of defining a variable for array length, use GArray::len
directly to avoid such confusion.
It broke after removal of usage of GTimeVal that was deprecated,
it requires seconds in this unix-based creation instead of microseconds.
The downside here is that it will create an extra object just to be
discarded in order to add the microsecond part to it.
It would end up segfaulting as it would return a NULL value
Based on Stream ID, the application can accept or reject the connection,
select the desired data stream, or set an appropriate passphrase for the
connection. Example usage:
srt://127.0.0.1:1234?streamid=mystream
Can be reproduced with:
videotestsrc ! x264enc key-int-max=$N ! \
h264parse ! msdkh264dec ! fakesink sync=1
It happens with any gop size but the smaller is the distance N
between key frames, the quicker it is leaking.
Fixes#1023
Current code would change any non-ok return from gst_pad_push to
GST_FLOW_ERROR, thus hiding meaningful returns such as GST_FLOW_EOS.
Tests also added.
Most of avtpcvfdepay messages are currently logged as warnings, which can
make some scenarios - such as receiving two AVTP streams on the same
pipeline - too verbose.
This patch tones those message down to INFO or DEBUG level - more in
sync with avtpaafdepay logging.
Allowing the UDP elements to bind on an interface is needed in more
complex networks where there are mutiple networks interfaces without
default gateway
From the documentation of gst_base_parse_set_infer_ts, it should be
disabled for non-audio data. Currently just disabling for all video
parsers that have reordered data: h264, h265, mpeg, mpeg4, vc1. Was
already disabled in h263.
Exposure mode property, extra colour tone values (aqua, emboss, sketch, neon), extra scene modes (backlight, flowers, AR, HDR).
Missing vmethods for exposure mode, analog gain, lens focus, colour temperature, min & max exposure time
Contribs by Mohammed Sameer <msameer@foolab.org>, Adam Pigg <adam@piggz.co.uk>
In case of pkg-config we need to create the include directories object
from the path using include_directories(). For INTELMEDIASDKROOT or
MFX_HOME we need to add the alternate include path ./include/mfx as
Intel MediaSDK now puts the headers there.
To allow curlhttpsrc to support DASH streams that use the on-demand
profile, it needs to support HTTP Range GETs. In GStreamer, the RANGE
is specified by issuing a GST_FORMAT_BYTES seek to set the start and
end of the range. curlhttpsrc needs to implement seek and set the
appropriate curl options to make it add the Range header to the
request.
Currently tsdemux timestamps only the PTS, and only issues the DTS if
it's different. In that case, parsers tend to estimate the next DTS
based on the previous DTS and the duration, which can accumulate
rounding errors.
