speakers for 24/96 audio

Shadorne - You have no clue. Filter will smooth-out the steps but will never remove few Hz modulation that was shown at 21Hz sampling rate. Al, please help me here or I'm going to kill myself.

Irvrobinson - I'm not talking about frequency range of our hearing but rather resolution of our hearing similar to number of shades of gray you can distinguish taking into consideration adverse conditions like ambient noise, system noise, THD, IMD etc.

I don't care anymore to defend myself from such attacks. You guys have no basic education in electronics and post nonsense just to keep arguing. Signing off.

Kijanki & Shadorne, you're both basically right but you're referring to different things.

Shadorne is alluding to the fact that a low pass reconstruction filter will smooth out the steps and restore an essentially perfect sine wave, if the original analog input was a sine wave at a frequency slightly less than the Nyquist rate (or lower). Of course, the filter itself may have significant side effects, but that is another subject.

Kijanki was alluding to the fact that if the analog input is a brief transient lasting for a limited number of samples and having spectral components approaching the Nyquist frequency, then the mathematics won't work out ideally no matter how ideal the reconstruction process is. Which is correct, although as I said earlier whether or not that may be audibly significant with worst case material (e.g., high frequency percussion) is probably a matter of conjecture. Admittedly, the video does not directly relate to Kijanki's point.

As far as the relation between low sample rates and quantization noise is concerned, while lower sample rates would obviously result in coarser steps in the sampled (unreconstructed) waveform, I think that Irv is basically correct to the extent that the reconstruction process can be accomplished ideally. However, given the possible effects on high frequency transients that we've been discussing, that may result from having a limited number of samples, and given the non-idealities of real-world filters, I suppose there could be some second-order relation between sample rate and quantization noise. It's been a long time since I took the relevant courses. :-)

Best regards,
-- Al

Kijanki, are you implying that 24bit data words have a "finer grain" than 16bit data words? That each bit represents a smaller incremental signal level?

"Kijanki, are you implying that 24bit data words have a "finer grain" than 16bit data words? That each bit represents a smaller incremental signal level? "

That's the basic reason to us more bits in each sample in digital signal processing of any kind, isn't it?

Most sounds last at least a hundredth of a second or longer. My point is that even for a 15 KHz sound you are likely to be hearing 15000/100 = 150 cycles. It is irrelevant that the amplitude of a few cycles may not be graphically represented perfectly. The problem is the context we are talking about is related to hearing rather than graphical presentation of a waveform.

Although Kijanki is right about the graphical accuracy my point is that,as regards to human hearing and music, this is not so relevant. In essence the engineers at Sony and Philips did a thorough job when they came up with rebook CD! Perhaps if redbook CD was not as good as it is then SACD would not have failed. The problem is that SACD and other higher resolution formats are very much into diminishing returns compared to a well produced CD.

I would add that the graphical representation of waveforms and the "digital staircase" form one of the biggest and most enduring audiophile myths that analog is inherently better than digital. In fact, most of the benefits of analog come from the added distortion that is pleasing to the ear - analog tape machines are wonderful devices for audio compression(removing dynamic range)!