GoXLR Mini has different numbers of channels actually available (21, 23,
or 25) depending on its firmware/etc, but its UCM profile specifies
always 23. The count can then be bigger or smaller than what is actually
available.
Fail a bit more gracefully in the case of too few channels: create all
the split devices specified by the profile. The channels that aren't
actually available in HW just won't get routed anywhere.
ALSA upstream IIUC is saying that the channel counts should be fixed, so
spew warnings that say the UCM profiles are wrong if they look wrong.
Although the two structs have same initial sequence, it's not really
correct to cast between their pointers. Alsa-lib also does this only
internally, but not in API.
Support also non-UMP IO with ALSA seq, in case either alsa-lib or the
kernel does not have UMP enabled.
Add configuration option "api.alsa.seq.ump" for optionally turning UMP
I/O off, for easier debugging.
Generally ALSA UCM profiles should all work as they're supposed to be
device-specific, so be more noisy when the profile fails to be supported
due to the PCM device failing to open.
Some logging on the probe outcome in failure case also makes
spa-acp-tool etc. log output easier to read.
In SplitPCM mode, Focusrite Scarlett Gen 4 (USB 1235:8218) UCM profile
specifies "CaptureChannels 2" for the Mic1/2 inputs, but
snd_pcm_hw_params_set_channels(2) fails for the HW device.
Fix by not requiring the channel count to be exact for SplitPCM, but
also allow larger numbers of channels than what UCM profile specifies.
When setting Route param, check that the route actually is part of the
active profile.
Also, check that the device given corresponds to the given route, before
setting properties. acp_device_set_port() also checks this, but we
shouldn't allow updating properties of Routes in non-active profiles.
Setting ports or applying props on devices not part of the profile can
do unexpected things e.g. alter mixer settings.
When api.alsa.split-enable=true for ACP device, instruct UCM to not
use alsa-lib plugins for SplitPCM devices.
Grab the information from UCM for the intended channel remapping, and
add the splitting information to the nodes emitted.
Session manager can then look at that, and load nodes to do the channel
splitting.
While initializing ALSA cards with UCM, we call pa_alsa_ucm_add_port()
for each UCM device for each UCM verb. This checks if a port has been
already added by name and skips port initialization if it is already
done. Different UCM verbs can have devices with the same name, which
means their port names end up being the same. So, this port creation
step currently is only done for one UCM verb for both UCM devices.
Volume control setup is also part of this process. The UCM devices only
know about the volume mixer information from the UCM verb that they are
defined in, so the volume control setup is done for one UCM verb at a
time. Skipping this setup when a same-named port exists means only the
profiles belonging to the first initialized UCM verb have working
hardware volume control.
Move the volume control setup out of the port initialization block so
that we try to do it every time, therefore for every UCM verb. However,
check that we don't run it twice for the same UCM verb for a port.
In theory, the PlaybackVolume etc. value can be different per-verb
for the same device, so we can't simplify this code to a single volume
setup per port.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/840
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Commit 88bb0bd7cc ("alsa: Allow to augment ucm port properties") adds
icon properties for some common port names, so that GUIs can show a
relevant icon to help disambiguate devices. However, these still do not
show up in pavucontrol, because it shows icons based on mappings'
properties. Add the relevant property to mappings as well.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/839
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
While trying to figure out device subsets that have aren't internally
contain conflicting devices, we walk through all possible subsets and
check each set if it satisfies ConflictingDevices/SupportedDevices
listed in UCM configuration. For a better user experience, we want to
skip subsets that are fully included in another valid subset we will
also generate.
The iterate_device_subsets() function that achieves the former is
intentionally in iterative form to avoid a stack overflow, since it will
walk through 2^n sets. However, the iterate_maximal_device_subsets()
function that skips incomplete sets is in recursive form, as I had
assumed tail-call optimization would take care of the potential problem.
Convert iterate_maximal_device_subsets() to an iterative form, because
the recursion seems to trigger a segfault with more than 16 devices on
PulseAudio. It doesn't seem to happen on PipeWire, but it's better to
not leave it to luck.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/838
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
The alsa/acp code already supports getting a user-friendly monitor name
using the EDID-Like Data (ELD) information available from cards that follow
the Intel HDA specification.
This patch adds support for also parsing the SAD fields of the ELD, and
exposing the results as a "iec958.codecs.detected" property on the
corresponding node, which should make it possible to provide more
user-friendly configuration UIs and defaults.
The default value will take effect if the session manager does not set a
different value.
Brief example:
test@test:~/checkouts/pipewire$ pw-dump | grep -E "(iec958.codecs.detected|iec958.codecs)\":"
"iec958.codecs": "[\"PCM\",\"AC3\",\"EAC3\",\"TrueHD\"]",
"iec958.codecs.detected": "[\"PCM\",\"AC3\",\"EAC3\",\"TrueHD\"]",
<after powering on my receiver>
test@test:~/checkouts/pipewire$ pw-dump | grep -E "(iec958.codecs.detected|iec958.codecs)\":"
"iec958.codecs": "[\"PCM\",\"DTS\",\"AC3\",\"EAC3\",\"TrueHD\",\"DTS-HD\"]",
"iec958.codecs.detected": "[\"PCM\",\"DTS\",\"AC3\",\"EAC3\",\"TrueHD\",\"DTS-HD\"]",
Big thanks to Pauli Virtanen <pav@iki.fi>, who also wrote large paths of the
code for this patch.
These shouldn't be enabled by default, because they cannot be
autodetected, and if the configuration is not a supported one, they
produce loud BRRRT at 100% volume and ALSA errors.
If the PCM is loaded directly using only api.alsa.path,
state->card_index may never be initialised, and then we end up opening
the wrong ctl device. Let's try a fallback for this case.
When the driver node is destroyed (like when unplugging the cable) it
will drop/pause all of the follower ALSA nodes. This is something that
happens internally because of how the ALSA nodes work together, on the
PipeWire level, the nodes are still running and they will just be moved to
another driver.
The problem is that nothing will then start the nodes again after moving
it to the new driver. Fix this by keeping track of the desired target
state of the ALSA node and restoring that state when we detect that we are
paused when moved to a new driver.
Fixes#4401
Add a new followerClock block in the profiler info. This is only set
when the follower could be a driver and it contains the clock info used
for following the driver, mostly the rate difference and delay.
Dump this info in pw-profiler -J
Make sure we always set the info in the clock, especially also when we
are following.
Keep a running average and variance of the error. Use this to
periodically update the DLL bandwidth. When the variance gets smaller,
we update the DLL more slowly to stay closer to the ideal rate.
This seems to improve the rate stability.
The properties of the card might overwrite those of the PCM.
For example, the cards's `alsa.id` will be set on the PCM too
since 37a51533e0 ("acp: add more properties for the card").
To avoid that, call `pa_alsa_init_proplist_card()` first
in `pa_alsa_init_proplist_pcm_info()` instead of last.
See #4135
Don't use the api.alsa.soft-mixer option to disable the path selection
but make a new api.alsa.disable-mixer-path.
Disabling the path selection might leave cards unusable after suspend,
so a separate option is a better idea.
See #4311
Use the helper instead of duplicating the same code.
Also add some helpers to parse a json array of uint32_t
Move some functions to convert between type name and id.
Add spa_json_begin_array/object to replace
spa_json_init+spa_json_begin_array/object
This function is better because it does not waste a useless spa_json
structure as an iterator. The relaxed versions also error out when the
container is mismatched because parsing a mismatched container is not
going to give any results anyway.
We have to unlink pcms when they are linked to a driver from
a different pcm.
When a playback and a capture pcm is linked and we start
the playback pcm and the capture pcm later this can leads
to a 'EPIPE' error on the capture device.
...
spa.alsa: hw:3,0c: snd_pcm_start: Broken pipe
...
The port.name should be something fairly unique and stable per node that
is also human readable.
Make sure we include the ALSA client name and port name in the
port.name but try to avoid double client names when the client name
is already in the port name.
There is also a api.alsa.disable-longname that is now set to true by
default. Setting this to false will include the unique client and port
id to the port.name.
This should make the midi port names much more presentable and more in
line with JACK1.
While the spec allows for 1ppm changes, our rate matching logic applies
these changes quite often, which can be spammy on USB. I haven't seen
hosts mind this, but it seems like it might be a problem at some point.
Additionally, if we also have bind ctls enabled, every pitch update is
also a wakeup for ourselves (whether or not we're listening for the
pitch ctls, since the mixer fd does not distinguish between ctls, those
are filtered after we wake up).
The 10ppm threshold is empirically tested as being not "too noisy" (i.e.
when updates happen, I can see them scroll by with `amixer events`).
If necessary, we can make this configurable in the future.
Use the new UMP alsa sequencer API to make it produce UMP packets.
Set the alsa sequencer to MIDI2.0, which will make it convert all
messages to MIDI-2.0 UMP automatically. We can copy this straight into
the control buffers.
This also solves some problems with large sysex messages that are now
nicely split into chunks with UMP.
