Very skeptical of the thesis here. The practical limit is the sampling theorem and the bit resolution, not “computability.”
It is possible that there are persons who can resolve a 25th bit or be disturbed by aliasing associated with the sample rate chosen. But again, there is a bigger effect. It is well known that audiophiles like “warmth” which has often been attributed to analog saturation effects. But ultimately, what enters your ear is analog. It should be possible to design a digital system that reproduces this warmth at the analog ear, compensating for all harsh effects before detection.
There's another much, much larger issue, which is the complete lack of specification around the word "correct".
OK, so "a simple analog circuit like a passive 1-pole low-pass filter (see figure above) can have output values where digital emulations can't be sure if the result is correct". Sure, why not. But, stick another "passive 1-pole low-pass filter" next to it. Is it "correct"? Because it sure isn't "the same". That's the nature of analog circuitry like this.
"A person who has one analog circuit knows what the correct output is. A person who has two does not.", as the old saying about clocks goes.
The article is written from a mental perspective of there being a "correct", an exact, singular point that represents the one and only valid "correct" output for a digital simulation, and if the digital fails to capture that "correct" then it's just wrong. And it simply goes without question that if it is wrong obviously the music created with that wrongness has been destroyed or something, which is actually a pretty complicated topic of its own.
But "correctness" is actually a range in the world of analog... how are you going to prove that the digital simulation doesn't exist within the range of possibilities for a real circuit? Obviously there's plenty of "not even close" values. But digital getting to "close" can be hitting the bullseye because even if it doesn't match analog circuit A, it may well match analog circuit B with what is nominally the same design. (Which also suggests you can always expect that any digital simulation of a bit of analog gear you own will always not be quite exactly the same as what you have... but as long as it's in the range of the differences you can expect from another instance of your physical gear, which will also not be the same, it's basically "correct".)
Article also strikes me as written by someone who has at least some of the "Sampling fallacies and misconceptions" mentioned at https://people.xiph.org/~xiphmont/demo/neil-young.html . Doesn't come right out and say digital samples are jagged stairsteps but I think it's implied.
At standard pressure and temperature, a sound wave travels at 343 m/s, IIRC. At a distance of 1m, those soundwaves take about 3ms to get from a pair of speakers to my ears. I can set the latency of my soundcard to as low as 0.5ms, which would mean an additional distance of ~18cm. I easily move my head that much when playing standing up. I don't see how that additional latency is the issue.
That's because you're not describing audio production. You're describing audio reproduction.
In audio production you typically need to keep many audio sources in sync. It's not the latency of a single path that matters so much, it's that you need control of relative latency between different signal paths so that you don't introduce phasing effects.
It is possible that there are persons who can resolve a 25th bit or be disturbed by aliasing associated with the sample rate chosen. But again, there is a bigger effect. It is well known that audiophiles like “warmth” which has often been attributed to analog saturation effects. But ultimately, what enters your ear is analog. It should be possible to design a digital system that reproduces this warmth at the analog ear, compensating for all harsh effects before detection.