The order of attribute changes is random, so it's possible that controlCX is
accessible before the other devices, which marks the device as available but it
actually fails to open. Only consider the device accessible if both control and
PCM devices can be accessed.
This requires reacting to ATTRIB changes of pcm devices as well now.
Fixes#2534
We need to check the last offset against the size of the buffer, not the
remaining size in the buffer.
When the writing is split, this could cause the buffer to be reused
wrongly.
See #2536
PipeWire v0.3.7 or later hits assertion at alsa-lib mixer API due to
wrong handling of removal event for mixer element.
wireplumber: mixer.c:149: hctl_elem_event_handler: Assertion `bag_empty(bag)' failed.
The removal event is defined as '~0U', thus it's not distinguished from
the other type of event just by bitwise operator.
At the removal event, class implementator for mixer API should detach
mixer element from hcontrol element in callback handler since alsa-lib
has assertion to check the list of mixer elements for a hcontrol element
is empty or not after calling all of handlers. In detail, please refer to
MR to alsa-lib:
* https://github.com/alsa-project/alsa-lib/pull/244
This commit fixes the above two issues. The issue can be regenerated by
`samples/ctl` Python 3 script of alsa-gobject.
* https://github.com/alsa-project/alsa-gobject/
It adds some user-defined elements into sound card 0. When terminated by
SIGINT signal, it removes the elements. Then PulseAudio dies due to the
assertion.
Fixes: 1612f5e4d2 ("alsa-acp: Add libacp based card device")
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.
