speakers for 24/96 audio

. Closer you get to Nyquist frequency the more samples you need to properly reconstruct original waveform - not possible to do for short high frequency sounds.

Not so. The waveform is perfectly reconstructed. The mathematics are quite rigorous. The main issue with digital is

1. Anti alias filtering (higher frequencies must be eliminated prior to ADC or they can fold in)
2. Jitter

Both of the above add spurious non musical signals. Both can be managed

On the S/N discussion, this is usually around 100 dB on good gear. I am certain this is achievable because my speakers can hit about 112 dB SPL at the listening position (12 feet back) as measured with a SPL meter whilst I cannot hear any sound (when no music is playing) from the tweeter unless my ear is within about 6 inches. This translates to roughly 100dB(taking into account the difference in distance which is around 12 dB and assuming the threshold for hearing hiss is around 20 dB in the room with inherent ambient noise around)

I think the ambient room noise and the speakers peak clean SPL are the limiting factors in a typical setup.

I think tape hiss or vinyl noise is limiting you to about 60 or 70 dB dynamic range on analog recordings.

I think high quality digital recordings can probably achieve around 90 dB dynamic range - limitations being the ambient noise during the recording process.

This is why CD is more than good enough for playback. This is why there are a few rebook CD recordings that are world class.

Of course, in a studio the signals are manipulated - this creates the need for even greater dynamic range (24 bit or 144 dB) - not that they will necessarily have better S/N but they may want to boost some sounds by 20 dB or so and may apply digital filters (the accuracy of said filters improves significantly if you have more bits)

07-01-11: Shadorne
Of course, in a studio the signals are manipulated - this creates the need for even greater dynamic range (24 bit or 144 dB) - not that they will necessarily have better S/N but they may want to boost some sounds by 20 dB or so and may apply digital filters (the accuracy of said filters improves significantly if you have more bits)

Excellent point!

06-29-11: Kijanki
... 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).

06-30-11: Kijanki
Closer you get to Nyquist frequency the more samples you need to properly reconstruct original waveform - not possible to do for short high frequency sounds.

07-01-11: Shadorne
Not so. The waveform is perfectly reconstructed. The mathematics are quite rigorous. The main issue with digital is

1. Anti alias filtering (higher frequencies must be eliminated prior to ADC or they can fold in)
2. Jitter

Both of the above add spurious non musical signals. Both can be managed.

In theory Kijanki is correct. An infinitely long series of samples is required for the mathematics to work out perfectly. The consequences of that will be most significant for spectral components that are transient and that approach the Nyquist frequency (i.e., half the sample rate).

The extent to which that may be audibly significant on most recordings is probably conjectural. The Wilson Audio cd I referenced, among many others, leads me to believe that in general it is not a major factor as a practical matter.

Shadorne is of course correct, IMO, in emphasizing the significance of anti-alias filtering and jitter.

Best regards,
-- Al

Bottom line: I am not losing any sleep over hi rez digital. There are too many ifs to really matter at this point for me and the benefits are marginal compared to the extra cost and overhead associated with even larger data files.

Al, I found video to show what happens when sampling just above Nyquist frequency. It might be possible to fix the output with sinc or other reconstruction functions but only if signal lasts a lot of cycles. If signal is short and disappears reconstruction will have huge error.

http://www.youtube.com/watch?v=Fy9dJgGCWZI