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digital audio and ADC and DAC



From:           	"Copeland, Peter" <Peter.Copeland@xxxxx>
To:             	"'ARSCLIST@xxxxxxxxxxxxxxxx'" <ARSCLIST@xxxxxxxxxxxxxxxx>
Subject:        	RE: arsclist reel to reel player/recorder
Date sent:      	Fri, 15 Jun 2001 14:23:29 +0100
Send reply to:  	ARSCLIST@xxxxxxxxxxxxxxxx

Dear Everyone,

a couple of things have been nagging me about Peter Copeland's 
contribution dated 15 June.

The first is that it looks as if Peter and I were disagreeing over 
fundamental matters:

Point 2: Sorry George, you are just plain wrong here. The
> widest uncorrupted frequency range (let's pick a nominal 
frequency of  22.0499999kHz to show what I mean) will inevitably 
mean a very considerable smearing along the time axis. No filter, 
analogue or digital, can cut sounds between 22.0499999kHz and 
22.05kHz without generating what looks like "ringing" on (say) 
square waves. If "ringing" isn't synonymous with destruction of 
transient response, I don't know what is! .......... But,  for 
documenting the performance of A-to-D converters (or D-to-As),  
*including* their anti-aliasing filters, I cannot think of a better  test 
than an impulse test.

----- Peter, I don't think we disagree, I was simply taking your 
quotes from the "Hi-Fi News" tests: "The reviewer didn't go into the 
differences very deeply,  but being familiar with the information-
theory issues, I could see  that some were made to give the widest 
uncorrupted frequency ranges, others made to give the best 
transient responses, etc." The two statements, to me, amount to 
the same characteristics in an environment where we have 20 kHz 
as the upper limit, "uncorrupted" obviously referring both to 
amplitude and phase responses. I was not considering the 
possibility of recording and replaying even 22 kHz, because that is 
supposed to be dealt with by the sharp, but not brickwall, anti-
aliasing filter. I, too, like an impulse test, however I have not yet 
seen an argument for injecting it at the place you suggest. 
Obviously Richard Hess is right in referring to the oversampling 
techniques now in vogue. If you really want to scare the DAC DC-
wise you would use an impulse pair, one going one way and the 
other going the other. But again, what do you expect to achieve 
that is relevant to the audio performance?

The other thing that has been nagging me is the following statement:

> Well, I'm sorry this topic has now reached the stratosphere, but
> George has raised some interesting points. The rest of you may 
> go back to sleep, since (as I said last time) we are now at the 
> leading edge of technology.

----- I make my contributions in order for as many as possible of the 
readership to be aware of problems - large and small - that I find 
interesting. I don't see why we should not all be able to appraise 
"the leading edge of technology". 

Apart from certain experiments, I have most of my knowledge 
about digital audio technology from reading (I was definitely analog 
signal processing at university, and no ham - radio amateurs had a 
way with picofarads which made digital (noise and crosstalk) much 
simpler for them). I have been thinking which book to recommend 
that is accessible (to me at least - I could easily identify texts, 
which require much more than the mathematics I can muster, but 
for whose gratification?). 

One book that I can heartily recommend (it is compulsory reading 
in courses I teach) is --

John Watkinson: "RDAT", Focal Press, Oxford 1991, ISBN0-240-
51306-1. It may tell you more than you want to know about R-DAT, 
but it has a very good overview of basic digital techniques:

Chapter 2, "Conversion" (pp. 14-60) (this is what Peter and I 
discussed above) 

Chapter 3, "Digital audio processing (pp. 61-78 in particular)

Now, John Watkinson has also written a book simply called "The 
Art of Digital Audio" which has come in several editions. But that 
may be too detailled.

Kind regards to all,


George
Preservation Tactics


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