speakers for 24/96 audio

I dunno, Kijanki, I randomly looked at two good power amps in JA's testing, a Moon and a Pass, and they were both had measured s/n ratios of about -84db at 1 watt, which is actually excellent performance. You keep forgetting that most power amps have about 30db of gain. Measuring s/n ratio at full power is sort of cheating for marketing's sake. A speaker with 95db/2.83v/1m efficiency, like my old Legacy Focus, will let you hear the hiss from such an amp rather readily. So I still contend that for listening the amps are the limiting factor, not well implemented 16/44.

As for your comments about Nyquist, it would seem your real thesis is that digital reproduction isn't very good, even with a DAC1. I still wonder, why does it sound so good if you're correct? I'm missing something.

Irvrobinson - I assume that you buy properly sized amp for the speakers and the room. My amp is rated 150W at 6ohm and I am pretty sure I am getting peaks even larger than that (headroom). It corresponds to largest digital number coming from CD - meaning covers full dynamic range. If you listen at 1W then I agree that you have no chance to experience full dynamic range, not only because of the noise floor of the amp but more likely because of the ambient noise and threshold of our hearing.

To test if power amp is limiting factor is very simple - Just turn on power amp, set volume to zero and listen. Can you hear anything? I cannot - dead silent. If I cannot hear anything in very quiet room in my listening position why even bring numbers into discussion?

As for Nyquist - digital reproduction is decent from 16/44 media and, according to reviews, pretty good with SACD. I seriously doubt that they would release 24/192 master tapes to public. What is released right know as high resolution is often the same as 16/44 (I read article about it). SACD is a different story because it cannot be copied (pit width modulation) but it does not work with the server and selection is very limited. I settled at 16/44 for all the reasons I mentioned before but understand its limitations. I adjusted my gear accordingly with very forgiving Hyperion speakers.

. 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