"Audiophile" products get me so mad, I don't even want to look at them to mock them. It's blatant anti-science, willful ignorance. To the chagrin of people's urge to keep optimizing, digital audio has been solved decades ago: CD audio can perfectly represent audio containing tones up to 22kHz, with a 96dB signal to noise ratio. Those are both figures that seem to exceed the hearing of even the best listener on earth.<p>For <i>studio use</i>, 24 bits offer some benefits for editing, but <i>not</i> because it sounds better, because we cannot <i>hear</i> better. Instead, if you mix audio it's useful to have some headroom over the dynamic range of your music (headroom you cannot hear), before you mix it back down to 16 bits for listening. Think about pasting pictures into a bigger canvas before cutting it back down.<p>Things like 96kHz or even 192kHz audio can actually be <i>harmful</i> for listening: Not because you can fundamentally hear a difference (you can't), but because if you have any nonlinearities in your audio playback chain, those inaudible frequencies above ~20kHz will cause intermodulation distortion in your audible range. Congrats, you've taken some useless crap over your hearing range and reflected it back as ugly crap into your hearing range.<p>Anyone who has invested time into understanding Fourier transforms and digital sampling on a fundamental mathematical level realizes that, and just gets angry at the lies. The Fourier transform and the sampling theorem don't know what an "open sound stage" or an "airier voice" is, if you think changes in sample rates and jitter translate into such human concepts instead of <i>plain noise</i>, you live in some phantasy non-physical world.<p>What is <i>not</i> solved is speakers and room acoustics. That remains notoriously hard, because a room easily can (and most often does) reflect parts of the spectrum such that it cancels itself out. No amount of equalizing can fix that (it's mathematically impossible), you need to change your room or speaker arrangement.<p>If you want to optimize audio, nerd out about that. Or if you absolute must, argue about <i>lossy</i> encoding formats (e.g. AAC vs. MP3) and bitrate. There is much less hard math in your way there, as they strongly rely on psychoacoustics. But anyone who claims that they hear a difference with 192kHz (on a provably linear audio chain) or an "audiophile Ethernet switch" shows that they're talking bullshit to begin with, so why would I trust their opinion on some potential real difference.