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From: steve@amdimage.UUCP (Steve eidson)
Newsgroups: net.audio
Subject: Re: CD digital filters-Eidson errs.
Message-ID: <514@amdimage.UUCP>
Date: Wed, 28-Aug-85 11:58:56 EDT
Article-I.D.: amdimage.514
Posted: Wed Aug 28 11:58:56 1985
Date-Received: Sun, 1-Sep-85 04:46:20 EDT
References: <493@amdimage.UUCP> <4213@alice.UUCP>
Reply-To: steve@amdimage.UUCP (Steve eidson)
Organization: AMDIMAGE, Sunnyvale, CA
Lines: 66
Summary: 

In article <4213@alice.UUCP> jj@alice.UUCP writes:
>>are linear phase FIR filters. The response of a linear 
>>phase FIR filter has no ringing. The most common filter
>Oh wow!  What a jem.  "Linear Phase FIR filters have no ringing"  Well,
>sir, may I recommend any simple, first year text on
	First, I'd like to apologize for making this statement
sound like a theorem. Of course FIR filters exhibit ringing, but
the interpolation filter I was referring to in the article does not.

>>I have used for interpolation is the simple: 
>>	y(n) = 0.25*x(n) + 0.5*x(n-1) + 0.25*x(n-2). 	
>What is its frequency response?  How well does it anti-alias?
><I know, you tell ME!  You've picked the simplest filter of the
>bunch.>
	If you knew what the frequency response was, why did 
you make this comment. You and I both know it is a low-pass
filter and the frequency response is far too complex to
broadcast over the net.

>Well, the model for the ideal interpolation filter is 
>the sync function (sin x)/x, which has lots of zeros, more and more
	Somebody else should check their elementary communications
text, it's "sinc" function.

>>Zeros are alternately inserted into the sequence to get a 
>>factor of two increase in the output data rate, and the output 
>>is multiplied by 2 to produce a properly scaled output.
>You're half right.  Figure out which half yourself.
	You better figure it out, it is all correct.

>>poles near the unit circle, which can be translated into 1-2^n,
>>but they are not as simple as the one-bit coefficients in
>>the FIR interpolation filter. I suggest that an IIR digital
>sigh.  wrong. figure it out yourself.
	you're wrong. you figure it out.

>>filter may appear to have the same response as its analog 
>>counterpart in an infinite precision computer simulation, but
>>in a real implementation, an digital filter will exhibit sensitivity
>>to coefficient truncation, while an analog filter will be
>>sensitive to the component values selected. 
>True completely for IIR designs.
	True completely for all digital filters.

>Please go read a good communications text, and then come back and
>make your explainations again.  Rabiner and Gold, Rabiner and Shaeffer,
>and many other books will quickly disabuse you of some of your
>rather random comments.
>This is just another proof of why audio is such a rotten field to
>work in.  I agree completely with Dick Pierce(sp) in that.
>Arrogance, indeed, to suggest that "Linear phase filters have
>no ringing".  

	Sir, I have no quarrel with you, but the overall general
hostile tone of your reply to my article was totally uncalled for.
I posted the article hoping to earnestly enlighten some of those
who are not familiar with DSP. How can honestly call me arrogant
after your reply. I admit my mistake, so please tone down your
conversation.
 ----------
 "...but you've got no arms and no legs,
 what are you going to do, bleed all over me ..."

 Steve Eidson (408) 749-2303
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