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#1 Re: Desktop and Multimedia » Output lag routing OBS through apulse » Today 00:41:26

stultumanto wrote:

The Firefox packaged with mainline Debian no longer supports ALSA directly.

It is not true. Firefox-esr works with ALSA. To reduce latency, you have to disable pulse-rust backend and recompile libasound2-plugins with --disable-pulseaudio
_https://dev1galaxy.org/viewtopic.php?id=7523

It is not difficult to compile Firefox without pulseaudio.

Firefox uses 32-bit floating-point audio format by default. If your sound card does not natively support this format, direct hw:device access will not work. You must use the ALSA plug plugin for format conversion. Configure your ALSA default device to use type plug with slave.pcm "hw:X,Y" for automatic format conversion.

#2 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » Yesterday 21:31:43

Phonon should be enabled in KDE settings.

There is also:

$ apt-file find /usr/bin/phononsettings
phonon4qt5settings: /usr/bin/phononsettings

I am not a KDE user, and KDE has nothing to do with the topic.
Start a new topic in Multimedia.

#3 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » Yesterday 20:00:42

Do you have Phonon installed?

phonon
-https://tracker.debian.org/pkg/phonon

binaries:

ibphonon-l10n
libphonon4qt5-4t64
libphonon4qt5-data
libphonon4qt5-dev
libphonon4qt5experimental-dev
libphonon4qt5experimental4t64
libphonon4qt6-4t64
libphonon4qt6-dev
libphonon4qt6experimental-dev
libphonon4qt6experimental4t64
phonon4qt5
phonon4qt5-backend-null
phonon4qt5settings
phonon4qt6
phonon4qt6-backend-null
phonon4qt5-backend-vlc/oldstable 0.11.3-1 amd64
  Phonon4Qt5 VLC backend

phonon4qt5-backend-gstreamer/oldstable 4:4.10.0-1 amd64
  Phonon Qt5 GStreamer 1.0 backend

#4 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » Yesterday 18:38:37

If you have a solid evidence (+log of compilation), file a bug.

AMAROK with Phonon

amarok 2>&1 Chris\ Rea\ \'And\ You\ My\ Love\'\ by\ Mila\ Gee\ \(HD\).mp3 
**********************************************************************************************
** AMAROK WAS STARTED IN NORMAL MODE. IF YOU WANT TO SEE DEBUGGING INFORMATION, PLEASE USE: **
** amarok --debug                                                                           **
**********************************************************************************************
[0000558a4e898f80] vlcpulse audio output error: PulseAudio server connection failure: Connection refused
[0000558a4e898f80] vlcpulse audio output error: PulseAudio server connection failure: Connection refused
[0000558a4e898f80] main audio output error: no suitable audio output module
QObject::connect: No such signal Phonon::VLC::MediaObject::angleChanged(int)
QObject::connect: No such signal Phonon::VLC::MediaObject::availableAnglesChanged(int)
WARNING: Phonon::createPath: Cannot connect  Phonon::MediaObject ( no objectName ) to  Phonon::AudioDataOutput ( no objectName ).

Enable Phonon plugin with arateconf

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

and try to run  AMAROK without apulse

But why did you post your AMAROK to "How to compile Audacious" topic?

#5 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » Yesterday 16:18:16

Does it help to solve the problem?

To solve such problems, one my try a scientific method proposed by Feynman

The first principle is that you must not fool yourself - and you are the easiest person to fool. So you have to be very careful about that. After you've not fooled yourself, it's easy not to fool other scientists. You just have to be honest in a conventional way after that.

Feynman, Richard P. (June 1974). "Cargo Cult Science" (PDF). California Institute of Technology.
_http://calteches.library.caltech.edu/51/2/CargoCult.pdf
_https://ghostarchive.org/archive/20221009/http://calteches.library.caltech.edu/51/2/CargoCult.pdf
_https://paulsteinhardt.org/wp-content/uploads/2020/10/CargoCult.pdf

A hypothesis is a statement, which can be verified. Otherwise, it is a myth.
Scientific problems are not easy to solve. You may try, perhaps, a clean install of Devuan, and document your experiments.

#6 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-10 15:41:48

A simple script to toggle ALSA configs:
_https://dev1galaxy.org/viewtopic.php?pid=56780#p56780

Remove pulse, pipewire, and pulse plugin
_https://dev1galaxy.org/viewtopic.php?id=7523

Install fftrate
_https://dev1galaxy.org/viewtopic.php?id=7142

Run arateconf to configure ALSA

$ cat ~/.asoundrc
# ALSA library configuration file managed by arateconf.
#
# MANUAL CHANGES TO THIS FILE WILL BE OVERWRITTEN!
#
# Manual changes to the ALSA library configuration should be implemented
# by editing the ~/.asoundrc file, not by editing this file.

#=====================================================
# Configuration for system
#-----------------------------------------------------

# Perform dmixer
pcm.dmixer_system
{
	type			dmix
	ipc_key			1024
	ipc_perm		0666

	hint
	{
		show		off
		description	"Direct mixing of multiple audio streams (system)"
	}

	slave
	{
		pcm		"hw:system,0"

		rate		48000
		channels	2
		format		S16_LE

		period_size	1920
		buffer_size	7680
	}
}

# Perform dsnooper
pcm.dsnooper_system
{
	type			dsnoop
	ipc_key			1025
	ipc_perm		0666

	hint
	{
		show		off
		description	"Recording from the same device for several applications simultaneously (system)"
	}

	slave
	{
		pcm		"hw:system,0"

		rate		48000
		format		S16_LE

		period_size	1920
		buffer_size	7680
	}
}

# Perform duplex
pcm.duplex_system
{
	type			asym
	playback.pcm		"dmixer_system"
	capture.pcm		"dsnooper_system"

	hint
	{
		show		off
		description	"Full duplex for simultaneous playback and recording (system)"
	}
}

# Perform convert
pcm.convert_system
{
	type			rate
	converter		fftrate

	hint
	{
		show		off
		description	"Sample rate converter (system)"
	}

	slave
	{
		pcm	"duplex_system"
		rate	48000
		format	S16_LE
	}
}

# Perform plug device
pcm.primary_system
{
	type			plug
	slave.pcm		"convert_system"
	hint.description	"Default device (system)"
}

#=====================================================
# Configuration for PCH
#-----------------------------------------------------

# Perform dmixer
pcm.dmixer_PCH
{
	type			dmix
	ipc_key			1026
	ipc_perm		0666

	hint
	{
		show		off
		description	"Direct mixing of multiple audio streams (PCH)"
	}

	slave
	{
		pcm		"hw:PCH,0"

		rate		192000
		channels	2
		format		S32_LE

		period_size	7680
		buffer_size	30720
	}
}

# Perform dsnooper
pcm.dsnooper_PCH
{
	type			dsnoop
	ipc_key			1027
	ipc_perm		0666

	hint
	{
		show		off
		description	"Recording from the same device for several applications simultaneously (PCH)"
	}

	slave
	{
		pcm		"hw:PCH,0"

		rate		192000
		format		S32_LE

		period_size	7680
		buffer_size	30720
	}
}

# Perform duplex
pcm.duplex_PCH
{
	type			asym
	playback.pcm		"dmixer_PCH"
	capture.pcm		"dsnooper_PCH"

	hint
	{
		show		off
		description	"Full duplex for simultaneous playback and recording (PCH)"
	}
}

# Perform convert
pcm.convert_PCH
{
	type			rate
	converter		fftrate

	hint
	{
		show		off
		description	"Sample rate converter (PCH)"
	}

	slave
	{
		pcm	"duplex_PCH"
		rate	192000
		format	S32_LE
	}
}

#=====================================================
# Configuration for default audio device
#-----------------------------------------------------

# Perform plug device
pcm.!default
{
	type			plug
	slave.pcm		"convert_PCH"
	hint.description	"Default device"
}

#7 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-10 15:12:43

The secret configs are here: /usr/share/alsa/

$ grep -r "defaults.pcm.dmix.rate" /usr/share/alsa/
/usr/share/alsa/pcm/dsnoop.conf:			name defaults.pcm.dmix.rate
/usr/share/alsa/pcm/dmix.conf:			name defaults.pcm.dmix.rate
/usr/share/alsa/alsa.conf:defaults.pcm.dmix.rate 48000

Debian Wiki:
Advanced features such as mixing should already be configured with sane defaults.
_https://wiki.debian.org/ALSA#Configuration

#8 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-10 14:17:14

It seems to be a sort of "secret software mixer". It is not difficult to prove that it does exist, and it is enabled. See: _https://dev1galaxy.org/viewtopic.php?id=7538

You may also try secret esoteric commands like these:

$ echo "Debian Default Sample Rate:  $(grep -r "defaults.pcm.dmix.rate" /usr/share/alsa/ | grep ":defaults" | cut -d\  -f2-) Hz"
Debian Default Sample Rate:  48000 Hz
$ grep -rE "defaults.pcm.dmix.rate|defaults.pcm.card|defaults.pcm.device" /usr/share/alsa/ | grep -E ":defaults.pcm.dmix.rate|:defaults.pcm.card|:defaults.pcm.device" | cut -d: -f2-
defaults.pcm.card 0
defaults.pcm.device 0
defaults.pcm.dmix.rate 48000

#9 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-09 16:35:57

@Danielsan

Don't you know that Debian/Devuan has already a very advanced dmix "configured with sane defaults"?

If you propose an alternative config, you may try to explain why your config is better than the default dmix config of Debian.

Debian/Devuan Defaults:

defaults.pcm.dmix.rate 48000
defaults.pcm.card 0
defaults.pcm.device 0

If you need 44.1 kHz sample rate and "card 1", you can set them in  ~/.asoundrc

$ cat ~/.asoundrc
defaults.pcm.dmix.rate 44100
defaults.pcm.card 1

These two lines are enough to set "default card" and default sample rate. A self-made dmix config is not needed.

Gentoo Wiki: ALSA: Configuration

When multiple sound cards are in use, the device numbers could be reordered across boots, such that using a name is advantageous.

If the correct name is unclear, a list of valid names can be easily obtained with:

cat /sys/class/sound/card*/id

Here is output from a developer's system that has multiple sound cards:

$ cat /sys/class/sound/card*/id
Q1U
HDMI
PCH
C930e

Here we have the Q1U microphone as Q1U, the builtin HDMI as HDMI, the analog audio jacks as PCH and a webcam's builtin microphone as C930e. Any of these are valid names for the card.

! Warning
Specifying numbers instead of names when multiple sound cards are used can result in device reordering across boots, which will prevent sound from working properly until the configuration file is edited to use the new number.

_https://wiki.gentoo.org/wiki/ALSA#Configuration

#10 Freedom Hacks » How to detect hidden ALSA resampling » 2025-11-07 16:07:53

igorzwx
Replies: 0

1. Make fftrate the default ALSA resampler

$ cat ~/.asoundrc
defaults.pcm.rate_converter "fftrate"
$ file 'rudra veena and pakhawaj.flac'
rudra veena and pakhawaj.flac: FLAC audio bitstream data, 16 bit, stereo, 44.1 kHz, 49123284 samples
$ file audio_test_48kHz_16bit.wav
audio_test_48kHz_16bit.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 48000 Hz

2. Run media players with debug 2>&1

$ audacious 2>&1 'rudra veena and pakhawaj.flac'
Input:  44100 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 940
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1024
Rates:  30080 --> 32768 (J: 0.09%, T: FFT, W: Vorbis)
Ok.
$ /usr/bin/totem 2>&1 'rudra veena and pakhawaj.flac'
Input:  44100 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 940
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1024
Rates:  30080 --> 32768 (J: 0.09%, T: FFT, W: Vorbis)
Ok.
$ mpv 2>&1 'rudra veena and pakhawaj.flac'
 (+) Audio --aid=1 (flac 2ch 44100Hz)
AO: [alsa] 48000Hz stereo 2ch s16
$ mpv 2>&1 audio_test_48kHz_16bit.wav
 (+) Audio --aid=1 (pcm_s16le 2ch 48000Hz)
AO: [alsa] 48000Hz stereo 2ch s16

3. Check hw_params

$ 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/system/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 1024
buffer_size: 16384

This means that the default sample rate is 48kHz. Let us change it. Presumably, there is already a sort of invisible dmix, so that we can set defaults.pcm.dmix.rate in ALSA config.

$ cat ~/.asoundrc
defaults.pcm.rate_converter "fftrate"
defaults.pcm.dmix.rate 44100
$ /usr/bin/totem 2>&1 audio_test_48kHz_16bit.wav
Input:  48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 480
Output: 44100 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 441
Rates:  48000 --> 44100 (J: 0.00%, T: FFT, W: Vorbis)
Ok.
$ mpv 2>&1 audio_test_48kHz_16bit.wav
 (+) Audio --aid=1 (pcm_s16le 2ch 48000Hz)
AO: [alsa] 44100Hz stereo 2ch s16
$ mpv 2>&1 'rudra veena and pakhawaj.flac'
 (+) Audio --aid=1 (flac 2ch 44100Hz)
AO: [alsa] 44100Hz stereo 2ch s16
$ cat /proc/asound/system/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 2734
buffer_size: 11026

It works because dmix is already enabled.

Debian Wiki:
Advanced features such as mixing should already be configured with sane defaults.
_https://wiki.debian.org/ALSA#Configuration

This means that there is already a very advanced software mixer with dmix and other plugins configured for pulseaudio. It might be obvious that this strange construction was created to imitate "bit perfect" playback of audiophile apps for macOS: media players can easily change the default sample rate of the software mixer to avoid software resampling. Try "bit perfect" mode of Audacious for macOS. It makes sense for macOS, because the built-in HW resampler of the DAC is better than the software resampler of macOS.

Why do we need this Stone Age technology? Configure a normal mixer with arateconf and forget about problems with sound quality. The so-called "bit perfect" is not needed, because the fftrate resampler is much better than the built-in HW resampler of your DAC. You can safely configure fftrate for the maximal sample rate supported by your DAC.

NOTE: When software mixer configured by arateconf, mpv does not resample anything,

$ mpv 'rudra veena and pakhawaj.flac'
 (+) Audio --aid=1 (flac 2ch 44100Hz)
Input:  44100 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1764
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  44100 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.
AO: [alsa] 44100Hz stereo 2ch s16

Explanation:

$ mpv 'rudra veena and pakhawaj.flac'
 (+) Audio --aid=1 (flac 2ch 44100Hz)       # mpv Input 
AO: [alsa] 44100Hz stereo 2ch s16           # mpv Output --> ALSA software mixer

# ALSA software mixer: fftrate
Input:  44100 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1764
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  44100 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

#11 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-06 21:06:30

$ audacious 2>&1 '08-Faust - Funeral March Of A Marionette [Tuncated, No Dither].flac'
Input:  352800 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 14112
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  352800 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.
$ audacious --version
Audacious 4.5-devel (Devuan 5 Daedalus)
$ 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)

Yesterday 23:15:26
I had already explained you that 24bit means 32bit in this particular case

The 32-bit representation serves as a larger, more flexible "container" to preserve the higher precision of the 24-bit data and prevent clipping during internal mixing or editing, especially in professional digital audio workstations (DAWs). The 24-bit data is stored within the 32-bit format, ensuring its precision is maintained without needing resampling.

Why 32-bit is Used as a Container
DAWs and audio servers use 32-bit internally to avoid the complexities of dithering and exporting to fixed-point formats like 24-bit...
When 24-bit integer data is stored in a 32-bit container, the 24-bit precision is preserved within the larger format, similar to storing a number like 0100 in a longer string of zeros, like 00000100.

...the data buffers which these descriptors define will contain the actual sound samples (or have samples written into them) structured like the content of .wav files (though 20 and 24-bit samples must be padded out with zeros at the LSB end to make them all 32-bits long).
_https://wiki.osdev.org/Intel_High_Definition_Audio

#12 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-06 20:23:49

If believe that the PulseAudio ALSA plugin is harmless, you may also try believe that you have a "bit perfect" playback, unless you've explicitly enabled resampling in a conf file.

#13 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-06 14:07:31

It has nothing to do with jack

Try to debug mpv and other players with 2>&1
For example

mpv 2>&1 *.mp3

Arch Linux: A list of common issues and solutions for Podman
_https://man.archlinux.org/man/extra/podman/podman-troubleshooting.7.en

Container permission denied: How to diagnose this error
_https://www.redhat.com/en/blog/container-permission-denied-errors

#14 Re: Desktop and Multimedia » [SOLVED] (alsa) arateconf not saving properly » 2025-11-06 03:14:17

There is already arateconf utility to generate ALSA configs. It works in interactive mode. It is self-explanatory. The problem is that semi-deaf artists cannot use it because of dementia.

#15 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-06 02:07:23

If pulseaudio is removed, the PulseAudio ALSA plugin sits at the ALSA library level, intercepting all audio calls system-wide. This means every application that tries to use ALSA will encounter the broken plugin trying to route audio through the non-existent PulseAudio server. The leftover plugin blocks exclusive mode access and causes audio failures, making it impossible to achieve the low-latency, direct hardware access that removing PulseAudio was meant to enable.

This does affect sound quality, but semi-deaf artists might be perfectly happy with such a "fine ALSA".

#16 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-05 23:12:17

you have a container installed in your home folder, which you are not allowed to read

chown: cannot read directory '/home/rich/.local/share/containers/storage/overlay/.../work/work': Permission denied

Install mc, navigate to that container and check who is owner, which permissions

➤ apt show mc
Package: mc
...
Homepage: https://www.midnight-commander.org
...
Description: Midnight Commander - a powerful file manager
...

Do you have a sort of Podman installed?

#17 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-05 21:15:26

Everything works, fftrate can resample everything

I had already explained you that 24bit means 32bit in this particular case

The 32-bit representation serves as a larger, more flexible "container" to preserve the higher precision of the 24-bit data and prevent clipping during internal mixing or editing, especially in professional digital audio workstations (DAWs). The 24-bit data is stored within the 32-bit format, ensuring its precision is maintained without needing resampling.

Why 32-bit is Used as a Container
DAWs and audio servers use 32-bit internally to avoid the complexities of dithering and exporting to fixed-point formats like 24-bit...
When 24-bit integer data is stored in a 32-bit container, the 24-bit precision is preserved within the larger format, similar to storing a number like 0100 in a longer string of zeros, like 00000100.

...the data buffers which these descriptors define will contain the actual sound samples (or have samples written into them) structured like the content of .wav files (though 20 and 24-bit samples must be padded out with zeros at the LSB end to make them all 32-bits long).
_https://wiki.osdev.org/Intel_High_Definition_Audio

Everything works, fftrate can resample everything

You still have a problem with pulseaudio:

ERROR ../audacious-plugins/src/alsa/config.cc:238 [guess_element]: No suitable mixer element found.

Read this thread
_https://www.linuxquestions.org/questions/linux-software-2/alsa-error-no-suitable-mixer-element-found-and-can%27t-connect-to-pulseaudio-4175623535/

_https://www.linuxquestions.org/questions/linux-software-2/alsa-error-no-suitable-mixer-element-found-and-can%27t-connect-to-pulseaudio-4175623535/#post5818488
The issue was permission-based. I logged into x as root briefly and pavucontrol opened with no problem, as did Audacious.

So I then went to my user account and tried, as my user name

chown -R lysander /home/lysander

It told me that there are quite a few folders not owned by me, but by root. One of them was /home/lysander/.config/pulse, whose ownership changed to root around the time I was doing the install of alsaequal.

I changed the ownership to my user whilst inside the folder with

#chown -R lysander pulse

Now pavucontrol opens fine and Audacious doesn't show any errors.

I should be careful in the future with installions and using root.

lysander - is the username of that person,

chown means change ownership

#18 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-05 20:11:50

Try to play .dsf with Audacious

Free samples are available
_https://dev1galaxy.org/viewtopic.php?id=6716

$ audacious 2>&1 *.dsf
Input:  1411200 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 56448
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  1411200 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

#19 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-05 17:41:00

Because of your personal Fletcher-Munson curve you may not hear the difference between your ALSA and pure ALSA

#20 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-05 17:29:59

libaudcore is inside Audacious
libaudcore is searching for "registered" plugins and cannot find them.

debdir/usr/lib/libaudcore.so.5.5.0

#21 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-05 17:06:33

Your ALSA is not pure, To get pure ALSA, you have to purge pulseaudio from libasound2-plugins
_https://dev1galaxy.org/viewtopic.php?id=7523

It is not enough to remove pulseaudio, you have to remove other crap to get "pure alsa".

Mate sound applet works with ALSA without problems.

EQ is needed for semi-deaf users to enhance high frequencies.

#22 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-05 16:27:13

If you want to try OSS4, you may better try it on ArchLinux.

Arch Linux AUR repository
_https://aur.archlinux.org/packages/oss

Package Details: oss 4.2_2020-2
Package Actions
View PKGBUILD / View Changes
Download snapshot
Search wiki
Git Clone URL:    https://aur.archlinux.org/oss.git (read-only, click to copy)
Package Base:    oss
Description:    Open Sound System UNIX audio architecture
Upstream URL:    http://developer.opensound.com/
Keywords:    oss
Licenses:    GPL2
Conflicts:    libflashsupport-oss-git, libflashsupport-oss-nonfree, oss-git, oss-nonfree
Submitter:    keenerd
Maintainer:    alexdw
Last Packager:    alexdw

Better oss-git:

_https://aur.archlinux.org/packages/oss-git
Package Details: oss-git 5693e1e-2
Package Actions
View PKGBUILD / View Changes
Download snapshot
Search wiki
Git Clone URL:    https://aur.archlinux.org/oss-git.git (read-only, click to copy)
Package Base:    oss-git
Description:    Open Sound System UNIX audio architecture
Upstream URL:    http://developer.opensound.com/
Keywords:    oss
Licenses:    GPL2
Conflicts:    libflashsupport-oss, libflashsupport-oss-nonfree, oss, oss-nonfree
Provides:    oss
Submitter:    Nowaker
Maintainer:    seawright
Last Packager:    seawright

I used PKGBUILD of oss-git package.

maybe therefore the error?

Error was because they both should be recompiled correctly.

#23 Re: Freedom Hacks » [HowTo] audacious_4.5-devel-1_amd64.deb (meson+samu and muon+samu) » 2025-11-05 16:09:34

It cannot find audacious-plugins, because they were compiled with another "audacions"

INFO ../src/libaudcore/plugin-registry.cc:431 [operator()]: Plugin not found: oss4

OSS4 plugin will not be compiled, because you do not have OSS4 installed. It will newer find it.

The method:
1. Compile Audacious
2. Install Audacious
3, Compile Audacious-plugins.
4. Install them

Do not enable OSS4 in the config, because you do not have it installed.

#24 Re: Freedom Hacks » ALSA without PulseAudio and PipeWire » 2025-11-05 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?

#25 Re: Desktop and Multimedia » [SOLVED] (alsa) arateconf not saving properly » 2025-11-05 15:03:58

On the contrary, that would be a regress. If the brain is not trained it regresses.
The brains of semi-deaf, semi-blind, and half-demented victims of pulseaudio do need regular training.

The brain experiences cognitive decline if not mentally stimulated, a process often described as "use it or lose it". Just as a muscle weakens from inactivity, a lack of mental training can lead to reduced thinking skills, memory problems, and a decline in creativity. This is because the brain can be thought of as a muscle that strengthens and maintains its function through regular use and challenges.

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