Fixes a number of warnings that look like this:
In file included from ../spa/include/spa/utils/result.h:37,
from ../spa/plugins/alsa/alsa-seq.c:35:
In function ‘set_timers’,
inlined from ‘do_reassign_follower’ at ../spa/plugins/alsa/alsa-seq.c:909:2:
../spa/include/spa/utils/defs.h:191:39: warning: ‘now.tv_sec’ may be used uninitialized [-Wmaybe-uninitialized]
191 | #define SPA_TIMESPEC_TO_NSEC(ts) ((ts)->tv_sec * SPA_NSEC_PER_SEC + (ts)->tv_nsec)
| ~~~~^~~~~~~~
../spa/plugins/alsa/alsa-seq.c:840:28: note: in expansion of macro ‘SPA_TIMESPEC_TO_NSEC’
840 | state->next_time = SPA_TIMESPEC_TO_NSEC(&now);
| ^~~~~~~~~~~~~~~~~~~~
../spa/plugins/alsa/alsa-seq.c: In function ‘do_reassign_follower’:
../spa/plugins/alsa/alsa-seq.c:836:25: note: ‘now’ declared here
836 | struct timespec now;
| ^~~
The reason for these warnings is that spa_system_clock_gettime() may
fail if a version check fails, but the code in question didn't check for
the possible fail. If it failed, then execution would continue, and the
arguments that were passed to the macro will be used uninitialized.
Fix this by checking whether function succeeded.
Move the code to check the position duration for changes to one
new method.
Also check for samplerate changes and adjust the resampler state
accordingly.
Strip the _alibpref from the device name, it contains a local counter
to identify the ucm card that should remain internal. We set a flag on
the device to notify of this.
Re-add the _alibpref of the local card to the device name if the
device was flagged.
See #1286
The _alibpref of the device was created in the session manager and
does not match our local _alibpref. Patch the device name with
the local _alibpref to make things match.
See #1286
The alibpref fallback does not contain the card number but it is
a local counter instead. Just check if it starts with something in
case the alsa library is not patched to return _alibpref.
These headers are designed for including in the project. So the user doesn't
need to install valgrind-devel and we don't have to worry about whether the
headers are available or not.
Use _alibpref to check if a device needs a UCM local config. Mark
the device as such and use this to set the OPEN_UCM property on
the device.
Open the UCM for a card when the device has the property set. Use the
same logic for loading the UCM as the acp code.
See #1251
UCM devices can require local data from use_case_mgr_open() but since
we do that in a separate process, make sure we reopen the use case
manager in case we are passed a UCM device so that the config is
available.
See #1251
When we enable a device, the node will be created and its software
volume will be set to 100%. Update the device volume with this as
well so that changing the volume has an effect.
Fixes#1198
According to the alsa-info.txt in the pipewire issues of #747 and #1206,
the Front Playback Volume is shared by Headphone and Lineout or
Headphone and Speaker, But Headphone, Lineout or Speaker they all have
independent Playback Switch, change to only use switch to mute the
Lineout or Speaker. This could resolve the issues of #747 and #1206.
See #1206 and #747
When we add a new listener to an object, it will emit the full state
of the object. For this it temporarily sets the change_mask to all
changes. Restore the previous state after this or else we might not
emit the right change_mask for the next listener.
Consider the case where one there are two listeners on an object.
The object emits a change and the first listener wants to enumerate the
changed params. For this is adds a new listener and then triggers the
enumeration. If we set the change_mask to 0 after adding the listener,
the second listener would get a 0 change_mask and fail to update
its state.
Motu M4 has four inputs (two line-in inputs, and two complete ones
with gain and XLR and whatnot), as well as four outputs (two monitor
pairs, and an unnaccounted headphone).
Sadly, like a few other interfaces, it wasn't being given an input
profile, since the matching code goes through default.conf testing
each config, and ends up selecting 'analog-surround-40', which does
not have input mapping. The inputs would fallback to 'multichannel-
input', which also doesn't have input paths.
Add input paths to all analog-surround-* mappings, and remove their
'direction=output' fields since they handle both out and in.
This replaces the manual check for "true" and some (inconsistent) return value
of atoi. All those instances now require either "true" or "1" to parse as
true, any other value (including NULL) is boolean false.
udev's ID_MODEL_ID and ID_VENDOR_ID are inconsistent: always 4-digit hex but
sound devices are prefixed with 0x, v4l devices are not. Depending on the
actual ID, the value will look like decimal (1234) or hex (a234).
pw-dump will then print those as either decimal integers (i.e. 0x1234 becomes
decimal 1234) or double (i.e. a234 becomes 41524.00).
Make this consistent by converting the string from hex do decimal where we
get it.
Reset the software volume when initializing the device and when the
hardware volume is updated.
The software volume was set to 0 by default and if the session manager
did not change the volume enough to cause a software volume change,
it would remain 0 and there would be silence.
Also improve the debug a little.
See #1160#1167#1164#1049#1117
When we start freewheeling, pause the device and resume when we
finish freewheel.
In freewheel mode, just discard samples in the sink and produce
silence in the source.
SPA_MEMBER is misleading, all we're doing here is pointer+offset and a
type-casting the result. Rename to SPA_PTROFF which is more expressive (and
has the same number of characters so we don't need to re-indent).
Make a new softVolume property that contains only the soft volume
to apply.
In the case of HW/SW volume, we pass the real volume in the
channelVolume and the leftover volume in softVolume. We don't
use the monitorVolume for this anymore because it is a completely
separate volume handled by the merger node.
This way, channelVolume always represents the effective volume
set on routes, channelmix and merger and only the softVolume (when
available) is applied as software volume by channelmix.
This makes things map a bit better to what is actually happening with
the real volume and leftover software volumes after applying the
hardware volumes in the device.
With this change, the volume on the monitor is not affected by the
sink volume anymore and we can use the monitorVolume for this later.
This also means that the monitor volume in pavucontrol of the sinks
does not change when the sink volume changes. PulseAudio is inconsistent
here: If the volume is HW, the monitor volume is not affected, if the
volume is SW, it is. In PipeWire there is an option in merger to
let the volume affect the monitor with monitor.channel-volumes = true.
This reverts commit 09106151d3.
We need the actual name and description of the mapping, not the
profile or else we end up with the full profile description in the
node description instead of the part of the profile that applies to
the device.
