At a minimum, anything above 16/44.1 requires far more than just files: monitors, a treated room, listening position, DAC, etc... but most importantly - a trained ear. That last one is the most uncomfortable truth.
You need at least twice the frequency range for sample rate in order to represent the original signal. That's slightly misleading though, that's from the Nyquist-Shannon sampling theory and it's a mathematical fact but that is true for exact numerical samples, once you add in quantization that muddies the water a bit. Taken at the extreme, it's straightforward to see why a 1 bit quantization per sample at 44.1 kHz would not capture a perfect representation of some analog signal even if there's only a 1 kHz frequency component to the signal. If we instead decide to sample at 10 MHz but still one bit quantization, now that 1 kHz frequency component can be much more accurately represented even though we're still using the worst quantization possible. Don't think of quantization like a square wave or a step pattern, think of it as "the signal is closer to here than any other discrete value".
Now in terms of realistic audio encoding, 16 bit at 44.1 kHz is designed to be a faithful representation as far as human hearing is concerned. Can someone with a trained ear potentially tell the difference between that and 24 bit at 192 kHz? In a studio environment it's possible. Most audiophile claims are dubious and a blind A/B test catches them out on most of it but the Nyquist-Shannon sampling theorem does not directly apply to quantized samples, it's about exact samples and with quantization, sampling rate is intertwined somewhat with the quantization depth.
I don’t have great hearing, so I’m not sure I can really weigh in here (thanks punk concerts in my teens). I remember similar arguments around screens and 60Hz vs ‘the human eye’. I think a lot of people, myself included, can easily perceive the difference between 60Hz and something higher- given the right conditions. I would not be so quick to disregard claims of more sensitive hearing.
(I commented on this topic above/below in more detail.) Even with not-so-great hearing you would still be able to identify the difference (ie artifacts are pushed down, not up). Look up articles on the practical limitations of AD/DA converters and why the seemingly counter-intuitive claim that the difference between 44.1 kHz and above is noticeable, is actually a fully industry-accepted practical reality: aliasing, AD/DA lowpass filters, etc.
The human threshold-of-hearing curve intersects the threshold-of-pain curve at about 20 kHz.
Above that frequency (or thereabouts) the sound has to be so loud that it will literally instantly damage your hearing before you can hear it.
This has been replicated across many studies for more than 100 years.
Flicker threshold is completely different. You can’t damage your vision by increasing the FPS, and it has always been commercially desirable to use a lower frequency because that is cheaper.
Would you agree that a trained human could identify artifacts produced by an imperect conversion process? If you lean "yes", then that's your answer: AD/DA is not a Rust function perfectly implementing the Nyquist theorem, it's a collection of physical components many of which introduce artifacts into the audio path. This thread is not about the theory of human hearing, the electronic components are literally imperfect.
They're no more imperfect than the pickups on an electric guitar, the assembly inside the microphone, the circuit in the compressor and everything else in the analog signal chain that exists long before AD happens.
As I responded below, you are confusing math with physical reality. A true 44.1 kHz converter can't realistically capture frequencies ~18-20 kHz due to the limitations of filters used in the process. A perfect lowpass brick-wall filter just does not exist - they all introduce artifacts, which a trained ear can identify. You don't need to be a dog to hear the difference, just someone who does not assume that Nyquist theorem can be magically applied in the real world (and, ideally, someone who utilizes high quality converters with oversampling).
That extra 4.1 khz sample rate is for headroom for a low pass filter (and not necessarily a brick wall one). Leftovers or any such artifacts are below the noise floor, which is also an important part of the physical reality.
Would be happy to see an actual, real study to prove that humans can notice, but to my knowledge none exist that confirm they can. Not even any on teenagers or younger (the only group that can even hear close up 20khz).
Is there evidence that a trained ear can reliably perceive these artifacts in a blind test of converters? I'd be interested in reading those links since converters typically oversample into the mHz range. At 11.29 mHz (256x 44.1 mHz), Nyquist will be at 5.64 mHz. Even the cheapest consumer converters are performing this type of oversampling.
To draw a design parallel: pixel-perfect design isn't something we are born with, noticing tiny details is a developed skill.
And yes, you are on point: oversampling is used extensively, but this just points at the exact issue: Nyquist theorem gave us a math algorithm, we still need to account for the electronic component imperfections. And then we are entering a different space of quality/precision/psychoacoustics/perception/etc. Meaning, not all converters, not all pre-amps, not all mics "sound" the same, even when they use same types of components on paper.
Oh, dear, that AES 2014 paper from Meridian (which was trying to push its controversial proprietary MQA audiophile system the same year) was widely criticized on audio forums when it came out, ranging from the rectangular dithering method to the use of a hard metal tweeter that could cause IM.
I don't. Do you? I am not a researcher. Saying that, do you have a double-blind study handy on MP3 256 vs 320 actual audible differences? If not, can you yourself hear the difference? If you can - it might be an illusion.
Max representable frequency is half the sampling rate (nyquist-shannon theorem), which is still a bit above normal but IIRC the extra headroom has something to do with eliminating aliasing
The artifacts produced by pure 44.1 kHz convertion are aliased back down to lower frequencies. It's not about a theoretical human ear, it's about the actual physics of AD/DA conversion.
But the energies of the signal present above the Nyquist frequency (22050Hz in this case) are almost always incredibly weak, and double blind testing rarely shows any indication that humans can actually hear the aliasing.
Mixing process often involves hundreds of tracks, and if each introduces aliasing, this can become a problem. Some engineers do swear by "the final mix is 16/44.1 so why mix at a different resolution?" mantra - that's fine too.
Sure, but those are averages. I'm 30-ish, and my hearing doesn't cut out until somewhere in the 21kHz range. When I was younger, it was even higher. One of my roommates in college had one of those anti-rodent high-frequency noise generators, we almost came to blows over it.
If you want to hear the difference between an audio file recorded at 44.1 and 88.2kHZ, then you need slow the audio playback down. Otherwise, a trained ear cannot physically hear the difference.
44.1 is "enough" only in theory. This assumes a physically impossible steep filter. Realistically, frequencies around 20 kHz will create audible artifacts (aliasing). So yes, a trained ear can tell the diffrenece between 44.1 and even 48 kHz. Like many other commenters in this thread, you are mixing up math theory with physical limitations of AD/DA converters. Oversampling is a common way to address this limitation, but strictly speaking 44.1 kHz is not as obviously "enough" as it seems.
> Realistically, frequencies around 20 kHz will create audible artifacts (aliasing)
The energy of the signal components above the Nyquist is generally very low, and very few double blind tests have given any indication that humans can detect the resulting aliasing (even though many people claim to be able to do, almost always in non-double-blind environments).
Badly written digital synthesis can generate high energy signal components above 22kHz, but that's because they're badly written, not because the theory is wrong.
Genereally very low for a single track? What about 200 tracks? Badly written synthesis, or badly recorded live instruments, or bounced and re-bounced dozens of times... we are not talking about the quality-defining aspect here. You can produce an excellent mix on KRKs connected directly to a MacBook.
This space is not driven by a single precise formula. 48/96 kHz helps some engineers to produce better sounding mixes. Can everyone hear the extended range of Adam tweeters? Probably not. But some can, and they benefit from that. Even if there is no double-blind study to prove this in absolute terms.
If you recorded 200 tracks of the same instrument, so that the partials above Nyquist were all broadly the same, then sure, summing the tracks would include summing 200 copies of the aliasing results too.
But very little music is like that, and the energy profile above Nyquist will differ dramatically. Consequently, you're not summing a set of identical aliasing results, and in general, the results will still be undetectable to almost everyone.
Jacob Collier routinely works with 300+ tracks in Logic. He doesn't worry about this sort of thing, and neither do the Grammy voters who love what he does.
Got it. Grammy voters love Collier's mixes. What about Tony Maserati? He can clearly tell the difference between 44.1 and 88.2. If your argument is that these engineers can't hear the difference - you are going to be disappointed. They can. Even Dave Pensado who mixes at 16/44.1, does that because he rejects the idea, he can hear the difference according to him.
Do you have citations for this claim? The "golden ears" argument is often employed by audiophiles, but even the cheapest converters oversample by up to several hundred times as well as employ antialiasing filters.
The most impactful for noticing the difference? Again, I would argue it's the trained ear. If you have plenty of mixing experience then all these details add up, and a treated room becomes the most critical - agree with that.
Other than the top engineers in the industry. This is a discussion that always ends up in the "double-blind study" vs actual real engineers working in the industry.