Change the logic to skip only devices which have "Input" as their IOID. The ALSA
Input/Output identifier (IOID) it may be either "Input", "Output", or NULL; with
the latter meaning that the device supports both input and output.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9534>
plugin_init() will not get called if element/feature registration
happens manually, such as when using linking only specific plugin
features with gstreamer-full. That is possible when plugins contain
static features.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9496>
The download element relied on a fuzzy translation from GStreamer format to a
DRM fourcc, and then all supported modifiers for that fourcc. Since !9306 this
was fixed to only enumerate that way when direct import is used.
Flag direct upload to the transform caps helper, so that we now enumerate all
non-external formats again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9339>
Add support to the ALSA device provider to enumerate PCM outputs that do
not correspond to a physical sound device i.e. they are "virtual" sinks,
like the plug, dmix, or softvol PCM outputs that can be setup in the ALSA
configuration files.
The main use-case for this is allowing usage of GstDeviceMonitor in setups
where there is no audio server and have custom ALSA audio configurations.
As those are likely to be uncommon, the feature is opt-in: a list of device
names and wildcard patterns separated by semicolons must be assigned to the
GST_ALSA_PCM_ALLOW environment variable before such PCM outputs will be
enumerated by the ALSA device provider. This allows either scanning all
PCM outputs, listing individual outputs, providing simple patterns with
'*' wildcards (which match only at the start or end of the name), or
a combination of them:
GST_ALSA_PCM_ALLOW=1 # Enable listing PCM outputs.
GST_ALSA_PCM_ALLOW='*' # Same, using a wildcard.
GST_ALSA_PCM_ALLOW='out_1;out_1' # Exact listing.
GST_ALSA_PCM_ALLOW='out_*' # Using a wildcard.
GST_ALSA_PCM_ALLOW='out_*;other_*;line_out' # Multiple items.
The main motivation for this patch is supporting enumeration of PCM outputs
in the WebKit GTK and WPE ports, which use GstDeviceMonitor to determine
which devices may be chosen for sound output. While on desktops typically
PulseAudio or PipeWire are used nowadays, on embedded devices it is often
desirable to avoid them and use custom configurations that perform audio
routing and processing using only ALSA.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8831>
Stop testing DSD rates in gst_alsa_detect_dsd_rates() if the rate becomes zero
or negative. This avoids an infinite loop if gst_alsa_probe_supported_formats()
is used on a PCM sink defined like the following in the ALSA configuration file:
pcm.buggy {
type plug
slave.pcm "buggy_volume"
hint.description "Causes an infinite loop in GStreamer"
}
pcm.buggy_volume {
type softvol
slave.pcm "buggy_dmix"
control.name "buggy_volume"
}
pcm.buggy_dmix {
type dmix
ipc_key 12345
slave {
pcm "hw:0,0"
period_size 1024
buffer_size 4096
}
}
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8985>
This pool isn't reusing its buffers, which makes it pointless to enable
the cache
Holding an extra buffer in free queue can also lead to a deadlock when
the pool's max buffer count is configured low (commonly 2).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8939>
Wrap dependencies add a ton of warnings with the latest GCC in Fedora
42. Squelch them by specifying that these dependencies are not
a part of the gstreamer project, and should be treated as system deps.
libsoup needs some porting work for the bump, and vorbis/lame are
already at their latest releases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8753>
There was a race condition where the demuxer would seek back to beginning after
determining the duration and while that seek was in progress one pad would
attempt to push a new buffer downstream, leading to a critical warning in
gst_pad_push().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8785>
Sometimes the seek to the end of file to determine the duration would trigger a
reset of the source pads, that would confuse the decoder downstream and trigger
an error. So the proposed fix is to not reset pads when the segment event being
processed is the consequence of a seek performed to determine the duration.
Fixes#4212
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/937>
After calculating a correct duration the oggdemux in some cases sets the duration to GST_CLOCK_TIME_NONE.
After that any seek will fail due to the oggdemux calculating a target time after the actual duration.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8296>
The check is exactly the same, but more explicit.
Original commit that introduced the check is
20fb58be198e7d76ece4e6c635cda7c919fad6d2:
---
vorbisdec: don't put invalid bitrate values into the taglist
Bitrates are stored as 32-bit signed integers in the vorbis
identification headers, but seem to be read incorrectly,
namely as unsigned 32-bit integers, into the vorbis structure
members which are of type long, which makes our check for
values <= 0 fail with files that put -1 in there for unset
values.
---
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>
* if downstream provides us with a pool that can allocate DMA buffers,
gldownload wraps it into a GL buffer pool that uses
glEGLImageTargetTexture2DOES() to import dmabuf into textures
* upstream GL elements can allocate from that pool
* gldownload unwraps DMA buffers from incoming GL buffers and passes
them downstream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>
