speakers for 24/96 audio

Mizuno - in order to avoid aliasing there should be no signal at 1/2 of the sampling frequency. In order to achieve it data has to be filtered out at 1/2 of sampling frequency in A/D processing.

Notice, that we are talking about preserving frequency information only (no aliases). Amplitude wise 16/44.1 will be very limited. Lets assume that you can hear 15kHz. Make picture of one full cycle of sinewave on a paper and try to place 3 points on it (reconstruct with 3 points only). You see the problem. Second problem is that filtering out info above 22.05kHz requires steep filters. Steep filters time shift different frequencies by different amount (uneven group delays) making inaccurate summing of harmonics. This will also screw-up step response (transients). Steep filters are not used in SACD recording making step response better. Of course master tapes are recorded in higher rate and re-sampled down but 96kHz playback will be still better than 44.1kHz (more points). 192kHz contains even more points but playback at 192kHz is not necessarily better than at 96kHz where THD of the most D/A ICs is the lowest (unless DAC uses extra info - downsampling). Resolution wise 24bit is better but most of converters are limited to about 20 bits anyway. Traditional converters are limited by tolerance of components to about 18 bits while Delta-Sigma are limited by timing errors to about 20 bits. One possible exception is Ring-DAC used by DCS (and previously licensed to ARCAM) that gets extra resolution by switching identical components of divider ladder in order to obtain more accurate average value. Some of the resolution will get buried in system noise that comes either from jitter (noise in time domain)or power amp's S/N.

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.