https://www.organism.earth/library/document/individual-and-the-world

pro-tip: if you toggle to "reader view" in firefox you can then fairly easily select all and cut/paste into a simple text document/file for offline reading(roughly a 150KB file / 24 page pdf)

more: the link referenced in the previous post is probably not the best offered at yt(commenters complained of numerous ads/commercials through the near 4hr audio-only "video" although i didn't personally sample said 4hrs)

another possible resource:

https://archive.org/details/alanwattscollection

If you are searching for intellectual entertainment, you may try
Alan Watts, The Individual & The World, Full Lecture
_https://youtu.be/Gr4Wo2OOJdY

EDIT:
The Brave Browser blocks ads & commercials in YouTube

To improve sound quality with ALSA, try to run it with apulse

apulse /usr/bin/brave-browser-stable

at your leisure/pleasure/satisfaction/etc

https://en.wikipedia.org/wiki/M%C3%BCnchhausen_trilemma

https://en.wikipedia.org/wiki/Karl_Popper

https://en.wikipedia.org/wiki/The_Open_Society_and_Its_Enemies

https://archive.org/details/dli.ministry.17322

hat-tip and kudos to @igorzwx and a heartfelt Endeavor to persevere!

Everything you do electronically or via software, changes the music from what the artist intended
it to sound like...
I use alsa precisely because it doesn't do those things by default, it doesn't re-sample...
But people who have never heard pure analog, tend to think digital is somehow better...

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.

Do you still believe that frequency and sample rate are the same thing, because both are measured in Hz?

I'm starting to have a theory

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

And in you, igorzwx, a symptom of delusional arrogance on steroids.

I'm starting to have a theory on this, hear me out: The real Igor died years ago, and some AI has taken over his personae. It would explain a lot.

off-topic

golinux wrote:

a symptom of delusional arrogance on steroids

It is not so simple. Think of Trotskyist dialectics: arrogance is modesty, knowledge is ignorance, "war is peace, freedom is slavery, ignorance is strength" (Orwell, 1984).
_https://en.wikipedia.org/wiki/Homage_to_Catalonia
_https://en.prolewiki.org/wiki/Dialectics

Notice that the fftrate package provides:
the fftrate ALSA plugin,
pcm_conv (standalone FFT resampler/converter),
and pcm_mse, which is to measure the exactitude of resampling.
_https://dev1galaxy.org/viewtopic.php?id=6722

The method of measurement is very simple:
1. downsample a HiRes wave 32bit 192kHz to CD format (16bit 44100Hz)
2. upsample the result to 32bit 192kHz format,
3. measure the difference with the original wave.

So that, you can measure the exactitude of resampling, and you may try to hear the difference.

If you fail to hear the difference, you may try to think why Intel HDA codecs (motherboard) support 32bit (fixed) 192kHz audio format, and Apple's MacBooks support 32bit (float) 96kHz audio format.

Since "ignorance is strength", one may to try to explain to Apple's audio engineers that the CD format is more than enough. It may help to find out whether Apple's audio engineers are always very polite.

Even with "perfect hearing" no one perfectly hears the alleged range of human hearing for 20-20k hz.

It seems that you cannot understand the difference between frequency and sample rate (i.e., resolution), simply because both are measured in Hz. It might be a symptom of dementia.

But still i'll humor you for the moment and your claims that my hearing (and half the Linux community),
is sub-par in some way (pretty presumptuous claim that).

My hearing though undoubtedly lessened in volume, is still quite accute after decades of
working in music. The legendary Tommy Ramone once said to me "I don't know how you made us
sound so good on that cheap PA system, but thank you!". That was during the South by Southwest
music festival in Austin one year, where I ran sound for 53 bands in 5 days.

I've installed sound systems in over 16,000 vehicles, boats, airplanes, homes, and businesses.
I have a rack of trophies for wins in audio competitions for sound quality. I was part of a team
that was IASCA grand national champions in sound for the 1001+ watts Pro class for several years
running. And not for nothing as they say, but I am a musician myself, and have put my time in
behind the PA as well as in front of it.

Everything you do electronically or via software, changes the music from what the artist intended
it to sound like, even small things like replaygain. I really dislike such attempts, Pulseaudio is
an excellent example of people flailing about with a keyboard and code and making things much worse.

I use alsa precisely because it doesn't do those things by default, it doesn't re-sample, and ffplay
doesn't either, allowing pass-through of the music somewhat less tampered with.

But people who have never heard pure analog, tend to think digital is somehow better, especially if they
do a ton of processing. But do yourself a favor, attend the symphony sometime, and actually listen to
what unprocessed real analog music sounds like, it will do you good.

Since the vast majority of Linux users are semi-deaf, semi-blind, and half-demented,

Well some of them are obviously...

FYI: resampling either does nothing, or it ****'s your music up. Period.

All software processing tends to do the same, and the more you heap on, the worse it gets. Having a keyboard and a grasp of coding does not make a person an expert on sound. And bulls**t you read on the internet (and copy/paste on forums) is mostly just that, means nothing in the real world.

Simple rule-of-thumb: The less electronics and software between you and your music is directly proportional to the quality of your listening experience. The digital age is the worst thing that ever happened to the quality of music.