That's one of those things that sounds true without putting thought into it but is ultimately false. A 44.1kHz sampled sounds can encode phases with offsets smaller than 22µs (1/44.1k). How much smaller? I'm not exactly sure, but I believe it's dependent on your sample size. My guess is 1/2^16 (assuming sample resolution of 16 bits), but I might be off by a factor of 2 (or indeed 100% wrong) there. Caveat emptor. If I'm right then that's 346 pico-seconds, which I'm betting is smaller than your brain can differentiate.

The formula, as far as I could find, is:

1 / (2 * pi * bandwidth * 2 ^ bitdepth)

So for full scale 44/16 signal it is about 0.1 nanosecond.

More here [1]. There are also some sample files for ABX tests there.

[1] - https://hydrogenaud.io/index.php/topic,108987.msg896449.html...

What you're sort of describing - having more of the higher frequencies resonate together - can actually cause intermodulation distortion (which itself can be in the audible range), which can make it sound worse...

I haven't had time to check the linked article but if it's the one I'm thinking of I'm pretty sure it goes into this.

In the end though, double blind testing (again, I think mentioned in the article) shows that there's a threshold (a bit below 48kHz, probably why 44.1kHz was chosen for CD audio) after which people cant distinguish between higher sample rates at any more accuracy than randomly guessing.

I think it's the intermodalation distortion or as I call it "the room sound" that your non-conscious mind perceives. The best example of this I've heard is when listening to identical masters on SACD/DSD vs PCM audio. Notably different sound when reproduced via loudspeakers.

Arrival time differences are essentially phase shifts and assuming infinite quantization you can encode phase shifts with infinite accuracy on a band limited sampled system.

So no.