So, having come from an image-processing background to put in 15 years of experience at audio processing companies, I've seen the xiph arguments, or something l

So, having come from an image-processing background to put in 15 years of experience at audio processing companies, I've seen the xiph arguments, or something like them, several times. Heck, I worked for Sony (where that precious 22KHz cliff was made) for close to ten years... in professional media... many of those years with Sony professional audio people. And the "44.1KHz is enough" or "48KHz" is enough" people are, sadly, kind of dumb. How do I know? Because I was dumb, too. Being a coding/math/audio/video badass, after a few years of industry experience, I rattled off some mouthy kid quip, saying "well, I don't know why we do 24/96, since nobody can hear above 20K anyway..." And a very talented, very knowledgable, and generally reserved engineer suddenly perked up one eyebrow and said, incredulously "because temporal and frequency response are inherently linked..." That look was in 2001, and I still remember that feeling of dread sinking in, realizing that I had no idea what he was talking about, no concept of why that would matter. I knew about Nyquist and could write a quick FFT, but he'd spent four years getting a degree in pure audio engineering at the most selective program in the country. That look, which I completely remember today, was like a deeply disappointed parent after a kid has just been bailed out of jail, from one of the nicest engineers I know. It was late, I was brash, and I pressed on, asking him what he meant. "What's a transient look like, spectrally?" he asked. He waited for my blinks to make audible sounds (due to the apparent hollowness of my head), then he asked "and how many channels of audio do we listen to?" He watched me stand there, like a doofus, for what seemed to me like several minutes (probably 5-10 seconds), then he went back to coding. It didn't hit me until weeks later, and I didn't really internalize until years later, that he was hinting at inter-ear phasing the other facilities of our auditory systems besides frequency response. Years later, I read up on Georg von Békésy's incredible work (including positional acuity), and I worked as a tech lead at Digidesign on the first-generation Venue live sound system which operated at 48KHz but with incredibly low latency (processing steps were 1, 2, or 16 samples) due to the requirements of, for instance, vocalists using in-ear monitors. Along the way, I ran across Microsoft engineers who thought that ~10-20ms inter-channel timing consistency would be okay in Windows Vista (it wasn't), conducted blind tests between 96KHz and 44.1KHz audio (for people who were shocked to immediately notice differences), came across plenty of hot-shot kids who said exactly the same kind of stuff I'd said, and saw postings from xiph making a mix of valid and grossly sophistic arguments ranging from "here's how waveform reconstruction from regularized samples work" (good) to "audio equipment can't even capture signals beyond this range" (dumb). At times, I thought about setting up refutation articles, then I realized, like many, that I had actual work to do. Von Békésy's work points to positional inter-ear phase fidelity of roughly 10µs. What's the sampling interval at 44.1KHz? >22µs? Good luck rebuilding that stereo field at 44.1... The trick is that there is a really serious diminishing return on audio sampling rate. 4KHz to 8KHz is enormous... 8KHz to 16KHz is transformative... 16KHz to 32KHz is eye-opening... 32KHz to 48KHz is appreciable... 48KHz to 96KHz is.... pretty minor, especially in the age of crappy $30 earbuds, streaming services delivering heavily compressed audio that will be crammed into a BT headset that may or may not be connected with additional compression, and all of the convenience that those changes bring. You may detect it in some audio if you're really listening, if you know what to listen for, and it may present advantages in system design (converters, processing, etc). From a data-rate perspective, the low-hanging fruit has already been picked. But people who smugly say that there is "no difference", that audiophiles are buying "snake oil", are letting their ignorance show, and that's including that kid that I was, 16 years ago. I've since moved out of pure pro media to consumer devices, where precision takes a back seat to the big picture a lot of the time. When discussing an audio fidelity multi-channel problem with a possible vendor last year, I expressed my concern about the inter-channel timing assurance slipping from 1µs to 50µs in product generations. "Depending on the sampling rate, that's several samples of audio", I said. A very senior engineer on our side (Director equivalent) quickly admonished me, saying "it's microseconds, not milliseconds", to which I said "I know... Which is why it's several samples, not several thousand..." From the look on his face, I'm 100% sure that he didn't understand me at the time, but I hope he put it together eventually. In the end, the industry has moved in the opposite direction of 24/192 for a long time. If we can get back to normalization of CD quality audio, I'll be happy.