From: gilchr@ee.ualberta.ca (Andrew Gilchrist)
Newsgroups: rec.audio.high-end
Subject: Re: Why digitize: some bunk
Date: Wed, 11 Nov 1992 03:49:58 -0700
Organization: University Of Alberta, Edmonton Canada
NNTP-Posting-Host: 129.89.7.4
Originator: tjk@csd4.csd.uwm.edu

Paul Winser <paul@prl.phil.co.uk> writes:


>kharstin@mtholyoke.edu (Ken Harstine) writes:
>> If that something is a 20-20kHz audio
>> signal, it has been proven mathematically that all of the information
>> present can be represented by samples taken at a rate greater than 40kHz.
Absolutely correct.

>For steady-state sine waves, OK. But steady-state sine waves
>(mathematically) carry zero information. Music, on the other hand is by
>definition an information signal and consists of a series of transients
>or events in the time domain. Can you be sure that all non-repetative
>transient audio waveforms can be losslessly represented by 44KHz samples?
Yes, as long as the waveform is band-limited to 20kHz. And we assume
Nyquist (ideal) sampling.  As another poster points out, practical
sampling introduces other complications.

>> If you mean to say that using a band
>> pass filter to restrict the information to the 20-20kHz signal is
>> removeing information, then how is this any different than the mechanical
>> band pass filters of the phonograph process, or the electro-magnetic band
>> pass filters of the "analog" tape recording process.  This filtering
>> processes behave in exactly the same ways (re phase shift and amplitude
>> vs. frequency).
Basically, the question is, how much information is contained beyond
20kHz.  The number didn't arise by testing pure tones for human audibility...
that number is coincidental.  The highest fundamental tone created by
a musical instrument is about 8kHz, created by a pipe organ. Pianos go
to about 4kHz.
The harmonic spectrum of a pipe organ extends almost to 20 kHz.
Music recording equipment must be able to accomodate this.

>There *is* a difference between the practice of bandlimiting signals in
>analogue and common digital equipment. Consider the response to an
>impulse of very short duration. A band-limiting analogue filter will
>output an initial large excursion followed by a ringing of diminishing
>amplitude. Digital filters for audio are nearly always designed to take
>advantage of the possibilty for mathematically optimal filter design
>that the storage of samples can provide. The output of such filters is
>seen to be a large excursion with *symmetrical* pre- and post-ringing.
>The pre-ringing of the impulse response in the time domain is acausal,
>that is we appear to get an output before the input has arrived.
>Gross Speculation: Might it be possible that our perception mechanisms,
>optimised by evolution for processing naturally filtered sounds, is
>upset by this very un-natural behaviour? Have there been any attempts
>to design digital filters for audio that mimic the response of analogue
>filters to produce strictly causal outputs?

What you describe is the result of FIR digital filters with linear
phase response.  Linear phase response implies that the total phase
distorion from the filter adds up to a pure time delay.  i.e. there
is no phase distortion, since we don't have a digitally locked phase
discriminator hooked up between the digital source and our brains.

What you are suggesting then, is that the sound of no phase distortion
from a filter is somehow less preferable than the sound of an IIR system
(digital or otherwise) which gives phase distortion.
i.e. distorted output sounds better.
It has been shown that people do not necessarily prefer undistorted sound,
so you may have a point. However, this has nothing to do with accuracy
of musical reproduction.

>Paul Winser                  Usual Disclaimers

Andrew