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The real question is whether resampling will make any detectable difference.
The real question is whether your trust your ears.
If you do not trust your ears, you can easily fool yourself with "objective tests", and enjoy any "digital crap".
If you do not trust your ears, how can you trust your eyes, your brain, your thoughts, and your opinions, and your "objective tests"?
If you do not trust your ears, how can you trust your eyes, your brain, your thoughts, and your opinions?
Of course they aren't.
Do you trust your eyes?
Do you trust your brain?
Do think that your brain can be deceptive?
@steve_v
Do you think your ears are not a reliable instrument?
I trust objective measurement
If you do not know how to fool yourself with "objective measurement", try to read Richard Feynman
Feynman, Richard P. (June 1974). "Cargo Cult Science" (PDF). California Institute of Technology. Archived (PDF) from the original on 2022-10-09. Retrieved 2015-10-25.
_http://calteches.library.caltech.edu/51/2/CargoCult.pdf
If you do not trust your ears, you might be perfectly happy with pipewire.
people having strange issues with the system bell before
Arch Linux users had such problem many years ago, when systemd was enforced.
The "beeping driver" was in the boot image.
It was not enough to blacklist it, one had to rebuild the boot image (Initramfs).
@steve_v
It seems that you trust your scope and terminal output rather than your ears.
It is not so difficult to install fftrate, configure is properly, and check whether you hear the difference.
If you don't hear the difference, you might be perfectly happy with pipewire.
It does simulate HiRes playback.
The deception works, if you cannot hear the difference between HiRes and 48kHz.
Resampling is forced if you use dmix
You are mistaken. Resampling cannot be disabled in ALSA.
With or without dmix, ALSA is always resampling everything to 48kHz.
To fool semi-deaf users, it does simulate HiRes playback with Intel HDA codec (and other HiRes DACs), but it sounds crappy.
The deception is justified by the theory that human beings cannot hear the difference between HiRes and CD format.
The only way to prevent ALSA resampling, is to install the fftrate codec and configure it correctly.
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 : 0Since the fftrate real-time codec is likely to provide much better quality of resampling than your DAC built-in hardware resampler, it is recommended to set the fftrate resampler to maximum sample rate supported by your DAC (digital-to-analog converter).
If pipewire is so good, why is it enforced upon Linux users?
@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/
Digital eXtreme Definition (DXD) is a digital audio format
_https://en.wikipedia.org/wiki/Digital_eXtreme_Definition
It is a wave file. For example:
24bit_352.8kHz.wav
32bitFloat_352.8kHz.wav
32bit_384kHz.wav
and the like.
"Unwanted resampling" means that your audio file is resampled by crappy resamplers, and your DAC is playing digital crap.
If your DAC is playing 32bitFloat_352.8kHz.wav, it is playing digital crap (this is how you detect unwanted resampling).
I does not make any sense to buy a DAC, if you are going to play digital crap.
If you do not want to hear digital crap, you have to remove pulseaudio and pipewire, install the fftrate ALSA plugin and configure it correctly.
The manual is here: _https://dev1galaxy.org/viewtopic.php?id=6644
EDIT:
It is not always easy to provide a polite answer to a question like this: "Why should I care that 2+2=4, if I want to believe that 2+2=5?"
One may better learn how to configure the sound system, and then think whether he need a DAC.
...good audio wisdom, suggesting a practical approach to improving your sound system by first optimizing what you already have and understanding your current setup before deciding if a DAC is a necessary upgrade for your needs. Configuring your system well and listening to it for a significant period can reveal if a dedicated external DAC would genuinely improve your audio experience or if your existing system's built-in DAC is sufficient.
Digital input word widths supported
16-24bit
Digital input sampling frequencies supported
32kHz, 44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz**, 192kHz
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%) Post your DAC spec.
By default, macOS resamples everything to 48kHz.
You can check MAC's audio settings with the "Audio Midi Setup" app (which is located in the "Utilities" subfolder of "Applications" folder).
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 -AYou may need to install inxi
sudo apt install inxiIf 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.wavYou may also try:
pcm_conv -f 48000 -b 64f -T fft -v your_file.wav test_64bit_Float_48kHz.wavpcm_conv -f 20000 -b 8 -T fft -v your_file.wav test_8bit_20kHz.wavYou can also create test audio files with Audacity:
sudo apt install audacityFree DXD samples are available here:
What We Hear With DXD 32-bit Files (Free Sample Downloads)
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/
Start Audacious with this command
audacious 2>&1and try to play your test_DXD.wav with Audacious.
You can also debug Audacious with this command:
audacious -VVYou may need to install Audacious
sudo apt install audacious audacious-pluginsand configure it:
Audacious > File > Settings > AudioWhy don't you ask that person who installed Devuan on your computer? He might be able to compile arateconf and configure ALSA.
Post the output of this command:
cat ~/.asoundrcand the output of these two commands:
whereis arateconfdpkg -l | grep alsa-config-utilsYou 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.
Don't panic! If you have an interview tomorrow, you may better try to compile.
Open: _https://dev1galaxy.org/viewtopic.php?id=7142
Scroll to the blue line:
How to compile Petrov's fftrate ALSA plugin on Devuan 5 Daedalus
Install "build-essential", etc.
sudo apt update
sudo apt-get install build-essential git debhelper
sudo apt install libasound2-dev libasound2-plugins swh-plugins and so on.
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
Post the output of these commands:
fuser -av $(find /dev/snd -type c 2>/dev/null)inxi -AYou may need to install inxi
sudo apt install inxi@g4sra
Are you paranoid about AI?
Your "angels" seem as hypothetical as "self-inflicted extinction" in your apocalyptic speeches.
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
@g4sra
There is math, and there are many methods to misuse it. It depends on the method of thinking.