It does simulate HiRes playback.
The deception works, if you cannot hear the difference between HiRes and 48kHz.

It is not enforced in the sense that it can be removed by experienced Linux users.

Resampling is not a problem, the problem is that it is enforced upon ALSA users.
PipeWire can be removed, but resampling cannot be disabled in ALSA (you can only change the default resampler of ALSA).
It seems that ALSA users are treated like miserable slaves.

The only way to prevent resampling with ALSA is to install the fftrate ALSA plugin and configure it correctly
(but you may need to reconfigure the fftrate plugin for audio files of another format)

$ aplay audio_test_48kHz_16bit.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: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  48000 --> 48000 (J: 0.00%, T: None, W: Planar)

T: None  means "without resampling"

$ aplay -v audio_test_48kHz_16bit.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: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  48000 --> 48000 (J: 0.00%, T: None, W: Planar)
Ok.

Plug PCM: Rate conversion PCM (48000, sformat=S16_LE)
Converter: fftrate
Protocol version: 10003
Its setup is:
  stream       : PLAYBACK
  access       : RW_INTERLEAVED
  format       : S16_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 16
  buffer_size  : 7680
  period_size  : 1920
  period_time  : 40000
  tstamp_mode  : NONE
  tstamp_type  : MONOTONIC
  period_step  : 1
  avail_min    : 1920
  period_event : 0
  start_threshold  : 7680
  stop_threshold   : 7680
  silence_threshold: 0
  silence_size : 0
  boundary     : 8646911284551352320
Slave: Direct Stream Mixing PCM
Its setup is:
  stream       : PLAYBACK
  access       : MMAP_INTERLEAVED
  format       : S16_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 16
  buffer_size  : 7680
  period_size  : 1920
  period_time  : 40000
  tstamp_mode  : NONE
  tstamp_type  : MONOTONIC
  period_step  : 1
  avail_min    : 1920
  period_event : 0
  start_threshold  : 7680
  stop_threshold   : 7680
  silence_threshold: 0
  silence_size : 0
  boundary     : 8646911284551352320
Hardware PCM card 0 'iMic USB audio system' device 0 subdevice 0
Its setup is:
  stream       : PLAYBACK
  access       : MMAP_INTERLEAVED
  format       : S16_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 16
  buffer_size  : 7680
  period_size  : 1920
  period_time  : 40000
  tstamp_mode  : ENABLE
  tstamp_type  : MONOTONIC
  period_step  : 1
  avail_min    : 1920
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 8646911284551352320
  silence_threshold: 0
  silence_size : 8646911284551352320
  boundary     : 8646911284551352320
  appl_ptr     : 0
  hw_ptr       : 0

@zapper

Can your DAC play DXD waves?

Free DXD samples are available here:
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

It does not support 32bit Float 352.8 kHz audio format. It can be used to detect unwanted resampling.

Free DXD samples:
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

This one is 32bit Float 352.8 kHz:

32-bit: Track 8, "Funeral March Of A Marionette" 
https://spaces.hightail.com/receive/9oP4dCgNhs 

$ ls *.wav
'08-Faust - Funeral March Of A Marionette - 32bit.wav' 

$ mediainfo '08-Faust - Funeral March Of A Marionette - 32bit.wav' | grep Audio -A11
Audio
Format                                   : PCM
Format profile                           : Float
Codec ID                                 : 3
Codec ID/Hint                            : IEEE 
Duration                                 : 4 min 35 s
Bit rate mode                            : Constant
Bit rate                                 : 22.6 Mb/s
Channel(s)                               : 2 channels
Sampling rate                            : 352.8 kHz
Bit depth                                : 32 bits
Stream size                              : 743 MiB (100%) 

It is advisable to remove pulseaudio and pipewire.
They can be detected with these commands:

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

inxi -A

You may need to install inxi

sudo apt install inxi

If you are using "a DAC over SPIDF", you may not want, perhaps, your audio file to be resampled by crappy resamplers. The crappy resampling can be performed by pulseaudio, pipewire, ALSA, or by the player in use.

The easiest way to detect unwanted resampling is to play audio formats which are not supported by your DAC.
If they are played, a sort of resampling/conversion is involved.

If, for example, your DAC does not support DXD format, you can convert an audio file (.wav) to DXD format with Petrov's pcm_conv

pcm_conv -f 352800 -b 32f -T fft -v your_file.wav test_DXD.wav

You may also try:

pcm_conv -f 48000 -b 64f -T fft -v your_file.wav test_64bit_Float_48kHz.wav

pcm_conv -f 20000 -b 8 -T fft -v your_file.wav test_8bit_20kHz.wav

You can also create test audio files with Audacity:

sudo apt install audacity

Start Audacious with this command

audacious 2>&1

and try to play your test_DXD.wav with Audacious.

You can also debug Audacious with this command:

audacious -VV

You may need to install Audacious

sudo apt install audacious audacious-plugins

and configure it:

Audacious > File > Settings > Audio

Post the output of this command:

cat ~/.asoundrc

and the output of these two commands:

whereis arateconf

dpkg -l | grep alsa-config-utils

You were supposed to compile and install arateconf (along with other things), and use arateconf (in interactive mode) to generate ~/.asoundrc

The sound icon "beside the clock" is not a "mixer". It is a sound applet.
The ALSA software mixer is to be configured with the help of arateconf

A mixer is supposed to mix something, as the name suggests.
OSS4 has vmix. It is enabled by default and works out of the box.
ALSA has dmix. It is not enabled by default. It has to be configured by the user.

You may try to configure ALSA with arateconf
It is not difficult to compile: just copy and paste commands to terminal.

The user manual is here:
_https://dev1galaxy.org/viewtopic.php?id=6644

The updated instruction for compilation is here:
_https://dev1galaxy.org/viewtopic.php?id=7142

@g4sra

Are you paranoid about AI?

It is not so difficult to misuse math without any "bias". Usually, it is a sort of calculated deception. Take, for example, applied statistics
_https://en.wikipedia.org/wiki/Lies,_damned_lies,_and_statistics

Don't panic! The so-called "self-inflicted extinction" is just another hoax, a false warning about something very dangerous, an ancient method to fool and enslave naive people. This method was invented in the Stone Age, and, nevertheless, humanity persists to multiply.

@g4sra

It is not about math, it is about the method of thinking. Would you believe that at least half of pulseaudio users is above the average level of deafness?

The problem is that resampling of digital audio is unavoidable simply because it is digital. Moreover, the semi-deaf artists tend to downsample their recordings to CD format with crappy resamplers. The mass production of "lossless digital crap" is a big business. It is about money. The naive, and the ignorant, and other slaves of "knowledge" seem to be victims of a global deception.

It seems that you believe that you are able to hear "pure (not re-sampled) digital audio". However, to become audible a digital audio file should be upsampled to infinite sample rate (that is, to analog). If you believe that frequency and sample rate are the same thing, you should not believe your ears when hearing "pure analog", because infinite frequency cannot be audible, and, therefore, "pure analog" cannot be better than "pure digital" (although, you seem to believe the opposite).

The genuine "pure analog" is better, of course. For the same reason, HiRes audio is better than that of CD format. The only problem is that ALSA users are not likely to hear the difference either because they are semi-deaf or because progressive dementia makes ALSA difficult to configure. To fix this problem, they created a theory that human beings cannot hear the difference between HiRes and CD format.

To state the obvious (which might be not so obvious for the demented), resampling is not a problem, if you do not hear the difference. Imagine you have a real-time codec, which is exact in an absolute sense. Make it default resampler/converter, and the problem is solved.

If you do not hear the difference, you may try to believe that the resampler is very exact. If you are a sort of unbeliever, you can measure the exactitude of resampling with Petrov's pcm_mse.
_https://dev1galaxy.org/viewtopic.php?id=6722
However, if you measure the exactitude of resampling, you may discover that you are semi-deaf.

Is it so difficult to compile fftrate? It should not be difficult, provided that you are not completely demented.

If you install the fftrate plugin and configure it correctly, you may have a chance to understand what ALSA really is.
If you are not completely deaf, you may notice the difference in sound quality.
If you are not completely demented, you may start to think what is so special about the fftrate plugin.

However, to achieve enlightenment, you have to think without attempting to create a theory, or other sort of mythology. Otherwise, it may result in "shaking and foaming from the mouth" (as it was described by Francis Williams and Peter Worsley).

Although there is a conspiracy theory that ALSA does not have a secret documentation, it might be a real challenge to understand how it works. The purpose of the fftrate ALSA plugin was to provide a sort of ersatz for "exclusive mode" which is not supported by ALSA.

NOTE: If you do not know what "exclusive mode" is, you may try it with foobar2000 on MacOS

brew install --cask foobar2000 

Notice, however, that Homebrew for MacOS is already infected with PulseAudio
_https://formulae.brew.sh/formula/pulseaudio
To prevent it from being installed, you have to add a secret esoteric command to your ~/.zshrc
One may find it in a "classified documentation":

export HOMEBREW_FORBIDDEN_FORMULAE="pulseaudio" 

export HOMEBREW_FORBIDDEN_CASKS="cask1 cask2"  # block installation of specific casks

export HOMEBREW_FORBIDDEN_FORMULAE="formula1 formula2"  # block installation of specific formulae

export HOMEBREW_FORBIDDEN_TAPS="username/repo"  # block installation from or adding specific taps

export HOMEBREW_ALLOWED_TAPS="homebrew/core"    # only allow install of packages from homebrew/core

However, it seems to be impossible to disable resampling in Foobar2000. The sound is always rather strange. Perhaps, Foobar2000 has a sort of pulseaudio inside, which downsamples everything to  44100Hz. One may presume that Foobar2000 is a very special player for those audiophiles who are not only demented but also deaf.

Perhaps, the only safe way is OSS4 with Audacious. When exclusive mode is enabled in the setting of its OSS4 plugin, Audacious does not play audio formats, which are not supported by sound card. Another option is "ossplay -R", but it can only play waves.

There is, of course, a secret esoteric theory of audio formats. Notice that Digital eXtreme Definition (DXD) usually implies an extreme resolution by time (sample rate 352.8 kHz), while 32-bit Float means an extreme resolution by amplitude.
_https://en.wikipedia.org/wiki/Digital_eXtreme_Definition
"What Is 32-Bit Float Audio, and Should You Record In It?"
_https://www.wired.com/story/32-bit-float-audio-explained/

Mac's built-in hardware DAC supports the following audio formats:
44.100 Hz 2 ch 32-bit Float
48.000 Hz 2 ch 32-bit Float
88.200 Hz 2 ch 32-bit Float
96.000 Hz 2 ch 32-bit Float

There is a reason why Apple's audio engineers chose "96 kHz 32-bit Float" as a sort of reference audio format.

One may try a sort of esoteric audio test:

1. Convert a DXD wave to Apple's reference audio format with Petrov's pcm_conv:

pcm_conv -f 96000 -b 32f -T fft -v DXD.wav 32bitFloat96kHz.wav

2. Set Mac's real-time codec (resampler/converter) to "96.000 Hz 2 ch 32-bit Float" audio format with the "Audio Midi Setup" app (which is located in the "Utilities" subfolder of "Applications" folder).
_https://support.apple.com/en-gb/guide/audio-midi-setup/ams59f301fda/mac

3. Play your 32bitFloat96kHz.wav on Mac with "Music" player (former iTunes).

If you are not completely deaf, you may notice something about audio quality.

If you need a reference DXD, try a sort of "2L Recordings".

To solve a problem, you have to trust your ability of solving problems, your ability to think.

It is the first symptom of "Vailala Madness". The next to come is "shaking and foaming from the mouth".

See: Schwimmer, E. (ed) (1976) F. E. Williams: The Vailala Madness and Other Essays. London: C. Hurst and Company;
and Worsley, P. (1968) The Trumpet Shall Sound, 2nd edition. London: Granada