Wide bandwidth = necessary?


Hi folks, there is one paradigm that bothers me a bit: many experts and audiophiles are stating that Red Book technology is outdated because of it's bandwidth limited function. I've read the human ear is capable of perception of frequencies beyond the normal human hearing, up to 40kHz. But this is only with live music! When listening to recorded music there is a restricted bandwidth because many microphones can only pick up frequencies up to 20kHz. So why the need for more and more bandwidth with regard to digital sound reproduction technology? What is not present in the recording can't be heard either, even with very wide bandwidth music reproduction gear.
What is also laughable is that many vinyl adepts say that phono playback gear can reproduce tones as high as 40kHz and that is one of the reasons phono playback sounds more "natural" than digital playback. This is a bit of a contradictio in terminis because most LP's are very band limited (30Hz to 16kHz is quite common). Your comments please.

Chris
dazzdax
Post removed 
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.
Your post makes sense *if* you are about to do an A-->D conversion

That's exactly what I was talking about. Please note that I said:

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.

I wasn't talking about DAC's or playback equipment at all.

-- Al