ALSA without PulseAudio and PipeWire

igorzwx · 2024-05-30 03:06:21

It is not difficult to remove PulseAudio and PipeWire.
The problem is how to configure a software mixer for ALSA.

Such mixer might be needed, if one wants to use ALSA just like MacOS users are using the sound system on MacBooks: for video conferencing with Discord or similar app, or playing HiRes audio files without distortions produced by crappy resamplers. This implies that the desired software mixer should have a high quality real-time codec (resampler and format converter).

These problems can be solved through the help of the fftrate plugin
https://github.com/PetrovSE/fftrate

Debian packages:

libasound2-plugin-fftrate_1.6.3_amd64.deb
alsa-config-utils_1.6.3_amd64.deb
pcm-utils_1.6.3_amd64.deb

They provide:

libasound_module_rate_fftrate.so — ALSA plugin (FFT resampler/converter)
arateconf  —  utility to generate ~/.asoundrc  
pcm_conv  —  standalone FFT resampler/converter

How to compile fftrate plugin

Install "build-essential", etc.

sudo apt update
sudo apt-get install build-essential git debhelper
sudo apt install libasound2-dev libasound2-plugins swh-plugins

Install gcc-10 and g++-10 from Devuan's oldstable repositories.

$ cat /etc/apt/sources.list | grep oldstable
# oldstable repositories
deb http://deb.devuan.org/merged oldstable main contrib non-free
deb-src http://deb.devuan.org/merged oldstable main contrib non-free

Enable the oldstable repositories and update

sudo apt update

$ apt search g* | grep "gcc-10/\|g++-10/"
g++-10/oldstable,now 10.2.1-6 amd64 
gcc-10/oldstable,now 10.2.1-6 amd64

sudo apt install gcc-10 g++-10

$ whereis gcc-10 g++-10
gcc-10: /usr/bin/gcc-10 
g++-10: /usr/bin/g++-10

To compile the code you should configure "make" to use gcc-10 and g++-10, instead of the default gcc-12 and g++-12.
To build the deb packages, you have to specify "compatibility level 13" in debian/compat

$ man debhelper-compat-upgrade-checklist | grep "recommended mode"
          v13 This is the recommended mode of operation.

Compilation

mkdir petrov-fftrate
cd petrov-fftrate
git clone https://github.com/PetrovSE/fftrate.git

Now you have to edit two files inside the package "fftrate":

./fftrate/src/lib/makedef.mk
./fftrate/packets/debian/compat

Enable gcc-10 and g++-10:

$ cat ./fftrate/src/lib/makedef.mk | grep gcc -A1
CC      = gcc-10
CPP     = g++-10

Change compat to "13":

$ cat ./fftrate/packets/debian/compat
13

Now you can compile:

cd  ./fftrate/packets

$ ls -1
Makefile
debian
etc
mk_dpkg

Compilte the code:

make

Make debian packages:

./mk_dpkg

$ ls ../ | grep .deb
alsa-config-utils_1.6.3_amd64.deb
libasound2-plugin-fftrate_1.6.3_amd64.deb
pcm-utils_1.6.3_amd64.deb

How to install fftrate

Install dependencies (if they are not already installed):

sudo apt install libasound2 libasound2-plugins swh-plugins

Install fftrate:

sudo dpkg -i libasound2-plugin-fftrate_1.6.3_amd64.deb alsa-config-utils_1.6.3_amd64.deb pcm-utils_1.6.3_amd64.deb

Open /etc/fftrate.conf with a text editor:

sudo nano /etc/fftrate.conf

Enable fft resampler:

$ cat /etc/fftrate.conf | grep "Transform type" -A3
# Transform type
# Available: dct, fft (default: dct)
#transform = dct
transform = fft

Run arateconf (interactive mode) to generate ~/.asoundrc

arateconf

Press 0 to select the first sound card.
Press 1 to select the second sound card.
Press F to change format

For Intel HDA codec on motherboard, try 32bit 192000Hz
For USB headsets, try 16bit 48Hz

Run TEST (press T).
If all tests have been passed successfully, save the setting (press S) and exit (ESC).

EXAMPLE: Intel HDA

===============
|  Main menu  |
---------------

    Curr. | Used | Play (def) | Rec (def) | Available cards
----------|------|------------|-----------|-----------------------------------
0 - >>>>> |  *   |  *     *   |  *    *   | HDA-Intel - HDA Intel PCH
1 -       |  *   |  *         |  *        | USB-Audio - iMic USB audio system

U - Toggle used flag
P - Set this device as default player
R - Set this device as default recorder

O - Output device: PCH,0
I - Input device : PCH,0

F - Format: 192000 Hz, 2 ch, 'S32_LE'
C - Converter: fftrate

A - Show all plugins [ ]
M - Plug-ins:
 [X] Convert,  [ ] Expand, [X] Asym
 [ ] Play Vol, [X] Dmix
 [ ] Rec. Vol, [X] Dsnoop
 [ ] Phonon,   [ ] Normalizator

T - Test
S - Save to '/home/devuan/.asoundrc'
X - Delete '/home/devuan/.asoundrc'

ESC - Exit

EXAMPLE: iMic USB

===============
|  Main menu  |
---------------

    Curr. | Used | Play (def) | Rec (def) | Available cards
----------|------|------------|-----------|-----------------------------------
0 -       |  *   |  *     *   |  *    *   | HDA-Intel - HDA Intel PCH
1 - >>>>> |  *   |  *         |  *        | USB-Audio - iMic USB audio system

U - Toggle used flag
P - Set this device as default player
R - Set this device as default recorder

O - Output device: system,0
I - Input device : system,0

F - Format: 48000 Hz, 2 ch, 'S16_LE'
C - Converter: fftrate

A - Show all plugins [ ]
M - Plug-ins:
 [X] Convert,  [ ] Expand, [X] Asym
 [ ] Play Vol, [X] Dmix
 [ ] Rec. Vol, [X] Dsnoop
 [ ] Phonon,   [ ] Normalizator

T - Test
S - Save to '/home/devuan/.asoundrc'
X - Delete '/home/devuan/.asoundrc'

ESC - Exit

TEST

> t
Testing H/W compatibility ...

Device: PCH
Output:
 Open device "hw:PCH,0" ... Ok.
 Set rate 192000 Hz ... Ok.
 Set channels 2 ... Ok.
 Set format 'S32_LE' ... Ok.
 Set buffer size 30720 -> 30720
 Set period size 7680 -> 7680
Input:
 Open device "hw:PCH,0" ... Ok.
 Set rate 192000 Hz ... Ok.
 Set format 'S32_LE' ... Ok.
 Set buffer size 30720 -> 30720
 Set period size 7680 -> 7680

Device: system
Output:
 Open device "hw:system,0" ... Ok.
 Set rate 48000 Hz ... Ok.
 Set channels 2 ... Ok.
 Set format 'S16_LE' ... Ok.
 Set buffer size 7680 -> 7680
 Set period size 1920 -> 1920
Input:
 Open device "hw:system,0" ... Ok.
 Set rate 48000 Hz ... Ok.
 Set format 'S16_LE' ... Ok.
 Set buffer size 7680 -> 7680
 Set period size 1920 -> 1920

Press any key ...

S - Save to '/home/devuan/.asoundrc'

> s
Saving config file...
Test: 'default' ... Ok.
Test: 'default' ... Ok.
Ok.

Testing playback/recording

$ aplay *.wav
Playing WAVE 'audio_test_48kHz_16bit.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Input:  48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Output: 192000 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 7680
Rates:  48000 --> 192000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

$ arecord -V stereo -f S16_LE -r 48000 -c 2 1-fft-exp_rec_alsa.wav
Recording WAVE '1-fft-exp_rec_alsa.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Input:  192000 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 7680
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  192000 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

$ aplay 1-fft-exp_rec_alsa.wav
Playing WAVE '1-fft-exp_rec_alsa.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Input:  48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Output: 192000 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 7680
Rates:  48000 --> 192000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

Now you can install apulse and run discord

apulse discord

You may also need ALSA mixer

alsamixer

man alsamixer

Last edited by igorzwx (2024-05-31 00:40:14)

swanson · 2024-05-30 08:58:22

Thank you! Looks very interesting.

igorzwx · 2025-02-15 22:57:34

EDITED

BEFORE running "arateconf", you should close all audio applications.
The "TEST" function of "arateconf" will fail, if your "audio device" is already in use.
When pulseaudio and pipewire are removed, you can easily free your "audio devices" with "fuser".
For example:

$ fuser -av $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2261 F.... mate-settings-d
                     igor       2310 F.... mate-volume-con
/dev/snd/pcmC0D0c:
/dev/snd/pcmC0D0p:   igor       8248 F...m firefox-esr
/dev/snd/seq:
/dev/snd/timer:      igor       8248 f.... firefox-esr

"mate-settings-daemon" and "mate-volume-control-status-icon" need not to be killed.
But "firefox-esr" should be killed to free the "audio device":

$ fuser -ikv $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2261 F.... mate-settings-d
                     igor       2310 F.... mate-volume-con
Kill process 2261 ? (y/N) n
Kill process 2310 ? (y/N) n
/dev/snd/pcmC0D0p:   igor       8248 F...m firefox-esr
Kill process 8248 ? (y/N) y
/dev/snd/timer:      igor       8248 f.... firefox-esr
Kill process 8248 ? (y/N) y
Could not kill process 8248: No such process

$ fuser -av $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2261 F.... mate-settings-d
                     igor       2310 F.... mate-volume-con
/dev/snd/pcmC0D0c:
/dev/snd/pcmC0D0p:
/dev/snd/seq:
/dev/snd/timer:

Now, when audio devices are free, you can run "arateconf".

kapqa wrote:

if i press "T", there is "failed".
Input:
 Open device "hw:C1,0" ... Failed!

It failed to open "Input", because you configured your "Cambridge USB DAC" for recording.
"DAC" means "digital-to-analog converter",
see _https://en.wikipedia.org/wiki/Digital-to-analog_converter.
As the name suggests, it is not supposed to be used to record "analog sound".

For "Output", you also got an error message:

Set format 'S32_LE' ... Unsupported!

It means that your Cambridge DAC does not support 'S32_LE' (it is now default for playback),
or ALSA driver (snd_usb_audio) for your USB DAC does not support 32bit.
Try 'S16_LE', or else.
Read the manual again and change 'S32_LE' to 'S16_LE'.
Press "F" to change "format".

======================
|  Sound parameters  |
----------------------
S - Sample rate = 48000 Hz
C - Channels    = 2
F - Format: 'S32_LE'

A - Alignment buffer and period [X]
B - Set buffer multiplier = 1
M - Set period multiplier = 1

P - Play ampl. = 0 dB
R - Rec. ampl. = 0 dB

ESC - Return to main menu
> f

==============================
|  Available sample formats  |
------------------------------
0 - U8
1 - S16_LE
2 - S24_3LE
3 - S24_LE
4 - S32_LE
5 - FLOAT_LE
6 - FLOAT64_LE

ESC - Return to main menu
>

Your Intel HDA codec (Generic_1) does support 48kHz 32bit.
It should also support 192kHz 32bit.
This may provide much better sound quality than your "Cambridge DAC" (it is likely to be 48kHz 16bit and nothing more with ALSA).
You can test different "sample rates" and "formats" with "arateconf".

You can also check your "proc" (cards, hw_params), for example:

$ cat /proc/asound/cards
 0 [system         ]: USB-Audio - iMic USB audio system
                      Griffin Technology, Inc iMic USB audio system at usb-0000:00:1a.0-1.3.4, full s

$ cat /proc/asound/card*/pcm*p/sub*/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 1920
buffer_size: 7680

$ cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 1920
buffer_size: 7680

NOTE: "arateconf" is simply a tool to configure ALSA,
that is, a tool to generate an ALSA config file "~/.asoundrc".
It takes data from ALSA and from Linux (proc, etc.).

Notable Realtek products include ... audio codecs (AC'97 and Intel HD Audio ).
_https://en.wikipedia.org/wiki/Realtek#Notable_products

ALSA detected an Intel HDA codec on your motherboard. It has a "generic driver" for it.
It should work without problems.
You may select Intel HDA codec as default for recording.
If you connect new USB audio devices, you have to create a new ALSA config with "arateconf".

Last edited by igorzwx (2025-02-16 21:38:24)

unixuser · 2025-02-28 21:32:40

You won't have sounds on firefox, do you ?

greenjeans · 2025-03-01 00:41:44

You won't have sounds on firefox, do you ?

I don't use the OP's method to run pure alsa, but I can attest that FF works for me just fine with nothing but alsa. All the browsers i've tested do.

igorzwx · 2025-03-01 01:46:58

unixuser wrote:

You won't have sounds on firefox, do you ?

Playback in Firefox works without problems with ALSA (without pulseaudio and pipewire).
Sound quality is good, because fftrate ALSA plugin is installed and enabled (with ALSA config).

NOTE: If you want to use Firefox for video conferences, you have to run it with apulse:

$ apulse firefox

It works without problems, if ALSA software mixer is configured with arateconf.
If you are an ALSA expert, you may try to create ALSA config without the help of arateconf.

hacksenwerk · 2025-03-01 11:12:20

igozwx wrote:

NOTE: If you want to use Firefox for video conferences, you have to run it with apulse

So that is the problem!
wtf?!

firefox became poetterware 8 years ago as it seems:
https://www.mozilla.org/en-US/firefox/5 … easenotes/

igorzwx · 2025-03-03 21:52:11

@hacksenwerk

Some years ago, when ALSA backend was removed from Firefox, you had to
1. compile apulse,
2. configure ALSA mixer (with arateconf),
3. allow ALSA sound device in Firefox's sandbox config.

Today it is more simple. You have to
1. install apulse from Devuan repo,
2. configure ALSA mixer (with arateconf).
! You do not need to configure sandbox for apulse.

HYPOTHESIS: Perhaps, it might be possible to use microphone with Firefox's ALSA backend without apulse.
What might be needed is to set

security.sandbox.content.read_path_whitelist                /dev/snd/
security.sandbox.content.write_path_whitelist               /dev/snd/

in about:config

See, for example:
How to run FIREFOX without pulseaudio, by using "apulse" (Nov 22, 2017)
_https://forums.linuxmint.com/viewtopic.php?t=257818

Last edited by igorzwx (2025-03-04 00:19:00)

Khylsdrak · 2025-05-19 12:22:49

@igorzwx > I was only using Pulsaudio and with your advice I tried and managed to configure Alsa correctly.

aplay /usr/share/sounds/alsa/Side_Left.wav

works fine !

Then I tried a stupid

aplay -t ogg ~/MyStupidSound.ogg

and the sound was transformed into a WAV sound, horrible. I realized that aplay couldn't play Ogg/Vorbis files

and I removed pulseaudio !

Well I want to thank you, the sound is really good, everything works perfectly !

Last edited by Khylsdrak (2025-05-19 12:23:59)

igorzwx · 2025-05-20 06:20:30

Khylsdrak wrote:

I want to thank you, the sound is really good, everything works perfectly !

If you enjoy it, you may say thanks to the creator of the magic fftrate codec, Sergei Petrov.
_https://github.com/PetrovSE/fftrate

There is a belief that fftrate may cure deafness. It may not always work. If it does not, the victims of pulseaudio may not have any chance to regain hearing loss. It seems impossible to cure deafness: "there are no known treatments to restore hearing, because auditory hair cells in mammals, unlike those in birds or fish, do not regenerate once lost."
_https://www.hsci.harvard.edu/major-step-toward-treatment-leading-form-hearing-loss

Khylsdrak · 2025-05-20 17:37:44

I will say thank you to Sergei Petrov but you helped me a lot so I thank you both

igorzwx · 2025-05-22 07:35:30

UPDATE

The updated installation manual for Petrov's fftrate ALSA plugin is here:
_https://dev1galaxy.org/viewtopic.php?id=7142

Thanks to the magic patch provided by the Arch Linux community, the fftrate ALSA plugin can now be compiled with gcc-12 (Devuan 5) and newer versions of GCC available in Arch Linux and Debian repositories. You do not need to install gcc-10 from the oldstable repository.

kapqa · 2025-09-02 15:33:08

hello , my issue is with [color]arateconf[/color] it is saving but not saving any settings. since i am using mint/(which is based on ubuntu, in turn based on debian) am i at fault?

igorzwx · 2025-09-02 15:54:49

I tested the fftrate (and bluealsa) on the latest Linux Mint Mate (pipewire was removed).
It works without problems.

Please, start a new topic and post the output of these commands

fuser -av $(find /dev/snd -type c 2>/dev/null)

inxi -A

cat ~/.asoundrc

whereis arateconf

and complete output of arateconf

arateconf

See also:
_https://dev1galaxy.org/help.php

Last edited by igorzwx (2025-09-02 16:04:01)

kapqa · 2025-09-02 17:01:56

@igorzwx,

thanks!

https://dev1galaxy.org/viewtopic.php?pid=57772#p57772

bai4Iej2need · 2025-11-04 19:24:33

@igorzwx
what is your base ?
cat /etc/devuan_version ?
cat /etc/debian_version ?

uname -a ?

igorzwx · 2025-11-04 20:58:46

Devuan 5 only

$ inxi -Sxxx
System:
  Host: devuan Kernel: 6.1.0-40-amd64 arch: x86_64 bits: 64 compiler: gcc
    v: 12.2.0 Desktop: MATE v: 1.26.0 info: mate-panel wm: marco v: 1.26.1 vt: 7
    dm: LightDM v: 1.26.0 Distro: Devuan GNU/Linux 5 (daedalus)

kapqa · 2025-11-04 21:36:29

for devuan 6 this routine is recommended

https://dev1galaxy.org/viewtopic.php?id=7142

bai4Iej2need · Yesterday 13:07:28

from
https://dev1galaxy.org/viewtopic.php?pid=54682#p54682

I don't use the OP's method to run pure alsa, but I can attest that FF works for me just fine with nothing but alsa. All the browsers i've tested do.

@greenjeans
What is your Method?
OPs method seems very elaborate, at least. If there was a single script okay, but doing all the steps manually - tldr

Last edited by bai4Iej2need (Yesterday 13:13:33)

igorzwx · Yesterday 13:24:41

Have you already purged pulseaudio from libasound2-plugins?

kapqa · Yesterday 13:31:44

can attest this method proven and effective for using alsa on several distro
"exclulsely"

1. purge pulseaduio + pipewire as per instruction
2. install apulse
3. install arateconf as per instruction, enable fftrate
4. >> configure arateconf,
eventuallzy reboot.

now the setup is not complete by any means (firefox can be compiled as suggested) and lacks some feature (like fallback mechanism if preferred audio card in arateconf is not turned on or available) but this can be amended by more advanced user or use of additional configuration or programs i suppose.

igorzwx · Yesterday 14:32:11

reboot is needed after removal of pulseaudio/pipewire
reboot is not needed for arateconf

fftrate should be configured for the maximum sample rate which is supported by your soundcard.
For example, Intel DHA codec (motherboard) supports 32bit 192khz (max), and you set 32bit 192khz with arateconf.
Then run test with arateconf (all audio apps should be closed, when you run test).

The fftrate resampler is much better than the built-in HW resampler of your sound card.
That is why fftrate should be configured for the maximum sample rate which is supported by your soundcard.

Most important, remove pulseaudio/pipewire and purge pulseaudio from libasound2-plugins
_https://dev1galaxy.org/viewtopic.php?id=7523

bai4Iej2need · Yesterday 15:00:14

This looks like a rush job. When I start from your beginning:

LANG= aptitude show alsa-config-utils
E: Unable to locate package alsa-config-utils_1.6.3_amd64.deb
LANG= aptitude show libasound2-plugin-fftrate_1.6.3_amd64.deb
E: Unable to locate package libasound2-plugin-fftrate_1.6.3_amd64.deb
LANG= aptitude show pcm-utils_1.6.3_amd64.deb
E: Unable to locate package pcm-utils_1.6.3_amd64.deb

Maybe you state all the preconditions, pathes, settings and sources, before urging the forum visitors to debug your advice.

Last edited by bai4Iej2need (Yesterday 15:04:44)

igorzwx · Yesterday 15:35:27

Are you trying to instsall libasound2-plugin-fftrate from Devuan repositories?

You have to compile it and install with dpkg. Then you can find it with "apt show", for example

$ apt show libasound2-plugin-fftrate
Package: libasound2-plugin-fftrate
Version: 1.6.3
Status: install ok installed
Priority: optional
Section: libs
Maintainer: Petrov Sergey <petrovse@mail.ru>
Installed-Size: 172 kB
Depends: libasound2, libasound2-plugins
Download-Size: unknown
APT-Manual-Installed: yes
APT-Sources: /var/lib/dpkg/status
Description: ALSA library additional plugin
 This package contains plugin for the ALSA library that are
 not included in the main libasound2 package.
 .
 The following plugins are included:
   - fftrate: FFT based rate converter
 .
 ALSA is the Advanced Linux Sound Architecture.
 .
 (Repackaged on Tue, 04 Nov 2025 03:13:01 +0100 by dpkg-repack.)

@bai4Iej2need

1. You have not read the manual.
2. You claim that is it "a rush job".

Are you a sort of pulseaudio activist?

Last edited by igorzwx (Yesterday 16:02:56)

greenjeans · Yesterday 16:54:16

@greenjeans
What is your Method?
OPs method seems very elaborate, at least. If there was a single script okay, but doing all the steps manually - tldr

No method to it really. I simply don't install anything but alsa when i'm doing an initial net-install (Openbox), and if I do a Mate install I go in post-install and un-install Pulse (and Synaptic forces me to install Pipewire). Then I uninstall Pipewire and it doesn't try to force Pulse, so pure alsa. Then install any additional alsa plugins like the EQ package and such. In Openbox I then install AlsaTune for a nice mixer/EQ gui.

Mate has a built-in mixer that you can't hardly get rid of without losing functionality. Openbox does not, so you get pure un-filtered sound with as little software between the music and your ears as possible.

All the browsers i've tried work fine with alsa-only. No magic needed or secret sauce.

The officially official Devuan Forum!

#1 2024-05-30 03:06:21

ALSA without PulseAudio and PipeWire

#2 2024-05-30 08:58:22

Re: ALSA without PulseAudio and PipeWire

#3 2025-02-15 22:57:34

Re: ALSA without PulseAudio and PipeWire

#4 2025-02-28 21:32:40

Re: ALSA without PulseAudio and PipeWire

#5 2025-03-01 00:41:44

Re: ALSA without PulseAudio and PipeWire

#6 2025-03-01 01:46:58

Re: ALSA without PulseAudio and PipeWire

#7 2025-03-01 11:12:20

Re: ALSA without PulseAudio and PipeWire

#8 2025-03-03 21:52:11

Re: ALSA without PulseAudio and PipeWire

#9 2025-05-19 12:22:49

Re: ALSA without PulseAudio and PipeWire

#10 2025-05-20 06:20:30

Re: ALSA without PulseAudio and PipeWire

#11 2025-05-20 17:37:44

Re: ALSA without PulseAudio and PipeWire

#12 2025-05-22 07:35:30

Re: ALSA without PulseAudio and PipeWire

UPDATE

#13 2025-09-02 15:33:08

Re: ALSA without PulseAudio and PipeWire

#14 2025-09-02 15:54:49

Re: ALSA without PulseAudio and PipeWire

#15 2025-09-02 17:01:56

Re: ALSA without PulseAudio and PipeWire

#16 2025-11-04 19:24:33

Re: ALSA without PulseAudio and PipeWire

#17 2025-11-04 20:58:46

Re: ALSA without PulseAudio and PipeWire

#18 2025-11-04 21:36:29

Re: ALSA without PulseAudio and PipeWire

#19 Yesterday 13:07:28

Re: ALSA without PulseAudio and PipeWire

#20 Yesterday 13:24:41

Re: ALSA without PulseAudio and PipeWire

#21 Yesterday 13:31:44

Re: ALSA without PulseAudio and PipeWire

#22 Yesterday 14:32:11

Re: ALSA without PulseAudio and PipeWire

#23 Yesterday 15:00:14

Re: ALSA without PulseAudio and PipeWire

#24 Yesterday 15:35:27

Re: ALSA without PulseAudio and PipeWire

#25 Yesterday 16:54:16

Re: ALSA without PulseAudio and PipeWire

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