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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.
Phonon should be enabled in KDE settings.
There is also:
$ apt-file find /usr/bin/phononsettings
phonon4qt5settings: /usr/bin/phononsettingsI am not a KDE user, and KDE has nothing to do with the topic.
Start a new topic in Multimedia.
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-nullphonon4qt5-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 backendIf 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, [ ] Normalizatorand try to run AMAROK without apulse
But why did you post your AMAROK to "How to compile Audacious" topic?
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.
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"
}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 48000Debian Wiki:
Advanced features such as mixing should already be configured with sane defaults.
_https://wiki.debian.org/ALSA#Configuration
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@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 0If 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 1These 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*/idHere is output from a developer's system that has multiple sound cards:
$ cat /sys/class/sound/card*/id Q1U HDMI PCH C930eHere 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
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 Hz2. 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 s163. 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: 16384This 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: 11026It 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 s16Explanation:
$ 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.$ 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 caseThe 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
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.
It has nothing to do with jack
Try to debug mpv and other players with 2>&1
For example
mpv 2>&1 *.mp3Arch 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
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.
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".
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 deniedInstall 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?
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/lysanderIt 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 pulseNow 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
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.Because of your personal Fletcher-Munson curve you may not hear the difference between your ALSA and pure ALSA
libaudcore is inside Audacious
libaudcore is searching for "registered" plugins and cannot find them.
debdir/usr/lib/libaudcore.so.5.5.0Your 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.
If you want to try OSS4, you may better try it on ArchLinux.
Arch Linux AUR repository
_https://aur.archlinux.org/packages/ossPackage 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.
It cannot find audacious-plugins, because they were compiled with another "audacions"
INFO ../src/libaudcore/plugin-registry.cc:431 [operator()]: Plugin not found: oss4OSS4 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.
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?
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.