From: Paul Winser <paul@prl.phil.co.uk>
Newsgroups: rec.audio.high-end
Subject: Re: Why digitize: some bunk
Date: Fri, 13 Nov 92 10:08:03 GMT
Organization: University of Wisconsin - Milwaukee
          id <sg.06493-0@ben.uknet.ac.uk>; Fri, 13 Nov 1992 10:48:01 +0000
          id <18112-0@eros.uknet.ac.uk>; Fri, 13 Nov 1992 10:47:22 +0000
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Originator: tjk@csd4.csd.uwm.edu



Previously, I wrote:

>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.

jj replies:

: This is entirely incorrect.  Symmetric filters are used so that
: there is NO group delay, i.e. the delay through the filter
: is uniform at all frequencies.  This directly implies that the
: filter show some form of symmetry.  (One can choose various
: kinds, but that doesn't much matter in this discussion.)  This symmetry
: is in place of the "near-minimum-phase" characteristics of the
: analog filter. This doesn't relate directly to the storage of
: samples, nor to the efficiency of calculation except that a particular
: kind of symmetry makes the calculation more efficient.  You have confused
: outright the reason for symmetric filtering.

You misunderstood me. Optimisation with respect to group delay is the
primary objective of the filter design (after giving a brick wall-ish
frequency response of course), and symmetrical time reponse is a side
effect of this. My crime was to be too vague in my terms. I dont think
this is the ideal forum for detailed discussions of signal processing
theory. 

My original post was intended to stimulate the debate beyond the
concrete technical information that we already have. I'm fully aware
of the implications of Fourier and Nyquist, I'm just not so certain
that they tell the whole story when it comes to the perception of music.

: The "ringing" has tails of (again, I'm designing the filter as we speak,
: these are real numbers) about 24 samples at 44.1 kHz.  This compares to 
: the response of the fastest cochlear filter of about 22 samples (the
: cochlear filter is not linear phase, but I'm calculating the first
: time-peak of the envelope, thus being conservative), so there exists
: some very minimal possibility that this effect could be MARGINALLY
: audible, given no other considerations.  Considering the effects of
: the ear canal and mechanical system (bones, eardrum), I think that it's
: very unlikely that this makes a difference.  This is as opposed to the
: issue of bandwidth, which can be argued to make a significant difference
: in a few cases with a few listeners.

Very unlikely, maybe. But progress in audio partly depends on the
investigation of unlikely things. Many of my friends whose opinions I
respect think that even the best CD systems are inferior to a good
vinyl playback system. As an engineer, this disturbs me, because the
distortion mechanisms of CD are so much smaller than that of vinyl. As
digital audio is here to stay, we should at least try to find out what
the problems are, if they really exist at all. Who would catagorically
state that sampling music at 44.1KHz (never mind about quantiastion)
has definitely *no* effect whatsoever on the subjective perception of
that music by a human being? Not me.

The parallel discussions on jitter in digital audio are also relevant
to this point. Here we have a time-domain phenomena introducing
distorions to the signal that should, by our conventional wisdom, be
too small to make an audible difference. And yet a large body of
opinion holds that they do. I dont think we know the whole story yet,
and I would rather approach the subject with an open mind and not just
say "Nyquist says ... and so digital audio must be inherently perfect".

As deanaj@elec.canterbury.ac.nz (A. J. Dean) writes:

: When we go
: limiting a frequency response (however it is done) we start making
: assumptions about how and what we hear. Our ears are very good frequency
: detectors, using extremely linear (finally, I can use that word when
: referring to the ear) mechanical transforms before sensing occurs for most
: of what we hear. It is by no means a fourier transform, but I speculate it
: mostly obeys the (linear) sum of sinusoids approach we currently use, so
: linear filtering will be, for the most part relevant and predictable. Also
: the ear is not purely a frequency sensing device, it also has good time
: domain properties, possibly allowing it to detect time domain events that
: escape our frequency based model. In fact quite likely come to think of it.
: And the thing is, when our assumptions break down, all this mathematical
: stuff becomes even further from the truth, especially when they break down
: in ways we don't realise or understand. Then to top it all off, the
: information (roughly 74 Mbits/s per ear) goes into the most sophisticated
: neural processor we know of, to be disected in strange and confusing ways. A
: lot of that info is lost, and I presume a fair bit is added as well.

Paul "eggs to babies" Winser