Use the NEAREST flag when setting a format. This only works for raw
formats and will update the format with the nearest accepted rate
or channels. We can then query the real configured format and use that
for the converter.
This makes things work when a driver tells us it can do 44100Hz but then
refuses and changes the rate to 48000.
See #2197, #2457, #2455, rhbz#2096193
Pass MIDI events as they are.
JACK requires NoteOn 0-velocity midi events to be patched to NoteOff
events for compatibility with LV2 plugins. Let's do this patchup in
the JACK layer then and add an option to disable it.
It's best to pass the midi messages unmodified and then patch them up
wherever they need patching up.
Assume that capture and playback nodes from a device have different
clocks. This enables the adative resampler to match them. A lot of devices
actually have slightly different rates and would work out of the box
with this fix.
Make an exception when the card is configured in the pro audio profile.
Then we force the same clock on all device nodes and avoid resampling
and rate matching. This can still be changed with a session manager
override.
When filling up the channels, either fill up the positions with one
of the know layouts or use AUX channels, never try to mix them.
This avoid cards with a large number channels to show a strange mix
of surround and AUX channels.
./spa/plugins/alsa/test-timer.c: In function ‘main’:
../spa/plugins/alsa/test-timer.c:224:79: warning: comparison is always true due to limited range of data type [-Wtype-limits]
224 | while ((c = getopt_long(argc, argv, "hD:f:r:c:", long_options, NULL)) != -1) {
|
The PropInfo either has a registered id (and then also a name from the
type-info) or a custom name as a string.
In all cases, the description contains a free form text that clarifies
the property.
Use the description in the stream controls name.
Use the max error to do a resync. Don't reset the dll, there is no
reason for that.
Don't use _rewind, but instead limit the amount of samples we read and
write
Should keep more stable sync in most cases.
If card busy check fails due to error, just log info message and
consider the card not busy.
For kernels with CONFIG_SND_PROCFS=n, /proc/asound is not present, and
we have to handle that. It's also better to fail open here, rather than
end up with missing devices.
Pass the current time around in various functions.
Make a higher precission htimestamp based get_delay() function. Seems to
work fine for playback but not for capturee.
When we get a commit error, try to resync our pointers. This fixes a
problem of endless commit errors after a quantum change because it never
manages to resync properly.
For kernels compiled with CONFIG_SND_VERBOSE_PROCFS=n, the pcmXX
/proc/asound entries do not exist. In that case, the "device busy"
check cannot be done, but we should still check existence of PCM devices
correctly.
Count the number of PCM devices from /dev/snd, which should work also
without /proc/asound or /sysfs/class/sound.
Alsa device acp probe results to missing profiles if some PCM devices
are busy. Currently, we retry based on a timeout and give up after some
retries. However, exposing cards with missing profiles is never
useful.
Never expose cards if some PCM devices are busy. Instead, retry adding
device on inotify fd close events, which arrive when some process has
closed a PCM device.
When probing for devices in alsa-udev, check via /proc to avoid inotify
busy loop.
There is a race condition on udev permission changes (e.g. switching
VTs), when pipewire from one user closes devices, and the other process
from second user tries to open them. The close/open are not ordered, so
opening a device may fail in alsa-acp-device.c resulting to missing
devices/profiles.
Address this by trying to open devices already in alsa-udev.c. If some
devices are busy, do not emit the device info yet, but retry after a
timeout. If it still fails, go ahead emitting the device, even if some
profiles may be missing. The retry with delay should be enough to solve
the race almost always.
So that we can catch the SOUND_INITIALIZED update between enumerating
devices and starting the monitor.This fixes a races in detecting devices.
Thanks to Matthias Fend for finding this.
Fixes#2046
