Wide bandwidth = necessary?

Chris, not to put too fine a point on it, but the issue is not really the bandwidth itself, but the brickwall filters used to rapidly diminish response at half the sampling frequency and their very adverse effect on phase response. Of course, when you refer to redbook replay as "bandwidth limited" your are pretty much only referring to the ultrasonic range, not the infrasonic. Have you ever seen square waves taken of redbook CD players? They are not pretty. Of course, the audible effects of preserving phase have been debated since the sun was invented, and I won't get into that one here, except to say that many, many people believe that these timing relationships are key to preserving the fine detail in the musical information.

Likewise, I will avoid the analog/digital debate minefield, but I will say that the lazier HF rolloffs in LP replay do a much better job of mainting a recognizable square wave, though these are often overlaid with the ringing of the primary high frequency resonant frequency of the phono cartridge.

I've read the human ear is capable of perception of
frequencies beyond the normal human hearing, up to 40kHz.

Where have you read this? Was it in a reputable science journal or someone who
just bought a supertweeter?

The ear drum is filter - I don't think anything much above 20 Khz gets through.

I agree with Viridian. The main benefit of higher sample rates, which figures to be a very significant one, is that it allows for a gentler rolloff of the "anti-aliasing filter" that precedes the a/d converter in the recording chain.

The theoretical maximum frequency that a sampled data system can capture is 1/2 of the sample rate, or 44.1 x 1/2 = 22.05 kHz in the case of redbook cd. Anything above that frequency must be prevented by the filter from entering the a/d converter, or it would "fold down" to a much lower (and very audible) spurious frequency when reproduced.

So the anti-aliasing filter has to pass everything up to 20kHz, with flat frequency response, but attenuate everything at 22.05kHz and above to (hopefully) below what is the threshold of audibility at lower frequencies.

A filter with such an extremely sharp rolloff will inevitably introduce both phase distortions and frequency response ripple, within the 20 to 20kHz passband. 96 or 192kHz sampling would drastically reduce the sharpness of the rolloff, and minimize those effects correspondingly.

Regards,
-- Al

Al,
you got it all wrong, I'm afraid!

The main benefit of higher sample rates, which figures to be a very significant one, is that it allows for a gentler rolloff of the "anti-aliasing filter" that precedes the a/d converter in the recording chain.

The higher sampling rate is meant to benefit the reconstruction filter that follows the DAC.
Remember that the data going into the DAC is digital - a stream of 1s & 0s. What sort of anti-aliasing filtering are you going to do on a stream of 1s & 0s???

Your post makes sense *if* you are about to do an A-->D conversion. Then, yes, you need to band-limit the analog signal.

So the anti-aliasing filter has to pass everything up to 20kHz, with flat frequency response, but attenuate everything at 22.05kHz and above to (hopefully) below what is the threshold of audibility at lower frequencies.

it's the reconstruction filter that needs to pass everything DC-->22.05KHz & *not* the anti-aliasing filter.