speakers for 24/96 audio

That video is wrong. It is showing a stair step signal which is NOT what the output of a DAC would look like. The output will be smoothed by a filter in order to eliminate all that horrible spurious high frequency signal from the stair steps. The output filter will remove the stair step and restore the sine wave so that the signals are much more alike - even a little above Nyquist - absolutely No need to got 100Hz sampling to properly render a 10Hz sine wave.

warning not everything you see from Universities is accurate.

Also it is WRONG to compare signals in this way. We hear frequencies NOT the waveform as presented graphically! The closeness of the waveforms as presented graphically is NOT a proxy for how close alike they will sound!

"I think that our hearing ability ends up slightly above 16-bit perhaps 18-20bits but I'm more concerned with sampling rate because low sampling rate in addition to phase shifts in steep low pass filters increases quantization noise (or size of square steps to make it simpler)."

It's statements like that, Kijanki, that make me wonder if you know what you're talking about. The width of the data word has nothing to do with our hearing ability. How many bits per word determines how many loudness levels there are. It's the sampling rate in KHz that determines how high the frequency response goes. You know that, right?

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?