speakers for 24/96 audio

Kijanki, your example of a 15KHz sine wave and "three points", implying poor reproduction, isn't the case, according to the well-proven Nyquist Theorem. 20KHz is reproduced as accurately as 1KHz with a 44KHz sampling rate. As for the phase shifts from steep digital filters, this is what over-sampling was invented to address, and eases the difficulty of designing a good anti-aliasing filter. For playback 16/44 is probably better than your audio system. (As I mentioned, for recording you might want the headroom of a longer word-length.)

Bob R is correct about some amps being a limiting factor in hearing the better s/n ratios of longer word-lengths. Most high-end amps are rated as having noise levels about 100db below full power. Since amps usually have about 25-27db of gain that means that their s/n ratio is only about -75db at 1 watt, or well inside the capabilities of 16/44. Krell amps, for example, are about the best, and they have s/n ratios in the mid-high -80db range at 1 watt.

Irvrobinson - 20kHz frequency is reproduced accurately (no aliasing). As for the amplitude, theorem assumes infinite number of samples (of periodic signal). Because it is not the case, interpolation is done with Sinc functions but with constantly changing signal that is close to 1/2 of sampling frequency it is very coarse. More samples would be better IMHO.

As for oversampling in A/D process - even if you sample at 192kHz your filters have to get 96dB attenuation at 96kHz to be 16-bit perfect. Such Bessel filter would have to have perhaps 16 or so poles. Attenuation of 20kHz/-3dB 8 pole Bessel filter is only 50dB at 96kHz. Fortunately signals at 20kHz have very low amplitude so that might be OK.

I like 16/44 and agree that a lot can be improved in other areas. Jitter, being source of noise, is one of them. We learned to remove jitter by better (dual) Phase Lock Loops or asynchronous rate converters (upsampling) but there is still some jitter from less than perfect A/D processing that cannot be removed (common for older recordings).

Kijanki - 20KHz reproduction with a 44KHz sampling rate is perfect for sine waves, not "coarse". A higher sampling rate doesn't improve accuracy within the frequency response of the lower rate, it just extends the frequency response. That doesn't mean I think digital recording and reproduction is perfect overall, it just means that in terms of capturing the frequency domain information at 20KHz, 44.1KHz sampling is completely sufficient to perfectly capture the sine waves. I think people confuse digital sampling with analog interpolation, and it isn't similar.

"capturing the frequency domain information at 20KHz, 44.1KHz sampling is completely sufficient to perfectly capture the sine waves"

Maybe sufficient for sinewaves but not for the music because it would call for brick wall filters that have very uneven group delays (non-linear phase if you prefer) and will cause wrong summing of harmonics. Such setup will be OK for single frequency reproduction but will be very unpleasant with music (dynamic signal).

Yes it is coarse because Nyquist-Shannon theorem requires infinite amount of terms (samples). Fixing it with sin(x)/x works poorly for short bursts around 1/2 of the sampling frequency. Sound of instruments producing continuous sound might be not affected (like flute) but anything with transients will sound wrong (piano, percussion instr. etc). Notice, that when people compare analog to 16/44 first thing they notice is different sound of the cymbals.

On the other hand, if you still think it is perfect system - enjoy.

"Maybe sufficient for sinewaves but not for the music because it would call for brick wall filters that have very uneven group delays (non-linear phase if you prefer) and will cause wrong summing of harmonics. Such setup will be OK for single frequency reproduction but will be very unpleasant with music (dynamic signal)."

I have no idea what you're talking about. The wrong summing of harmonics, and it'll be very unpleasant? I don't know about that, Kijanki. You talk like a technically competent person, but then you make these outlandish claims. So if these summed harmonics will be so screwed up, why is it that those of us with very good high frequency hearing and high quality speakers can't hear anything very unpleasant? And if they do sound so unpleasant, why when I listen to higher res stuff through a Benchmark DAC it doesn't sound noticeably better?

I think you're exaggerating the issues, and wrapping the arguments in technical-sounding reasons that really don't alter the music audibly.