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From: ogata@leviathan.cs.umd.edu (Jefferson Ogata)
Newsgroups: comp.music
Subject: Re: Oversampling: Re: DSP for Audio Marketing
Message-ID: <32864@mimsy.umd.edu>
Date: 12 Apr 91 22:05:36 GMT
References: <1991Apr5.172338.11258@vaxa.strath.ac.uk> <31771@usc> <1991Apr12.060657.19964@cynic.wimsey.bc.ca>
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Reply-To: ogata@leviathan.cs.umd.edu (Jefferson Ogata)
Organization: U of Maryland, Dept. of Computer Science, Coll. Pk., MD 20742
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In article <1991Apr12.060657.19964@cynic.wimsey.bc.ca> curt@cynic.wimsey.bc.ca (Curt Sampson) writes:
|> 
|> This is not completely correct.  When the sample is converted from
|> 44.1 to, say, 176.4 KHz, no interpolation or other changing of the
|> data is done.  The data is simply duplicated.  If you started with the
|> following sequence:
|> 
|> 	3 7 11 15
|> 
|> it would be turned into this when 4x oversampled:
|> 
|> 	3 3 3 3 7 7 7 7 11 11 11 11 15 15 15 15
|> 
|> Then a digital filter is applied, which is essentially a way of
|> "smoothing" the samples.  The "edges" you see (because it is
|> essentially a sqare wave before filtering) are what cause the high
|> frequences to be present.

A digital filter will do the same thing as interpolation in this
context. It may not be linear interpolation but it amounts to the
same effect as just straight interpolation with an appropriate
algorithm.

You still aren't saying anything about the analog Nyquist filter that
has to come after the DAC. Effectively oversampling allows this
analog filter to have lower attenuation because the high frequency
components of the digital steps have reduced amplitude from the
effect of interpolation (or filtering).

|> cjs
|> -- 
|>                         | "It is actually a feature of UUCP that the map of
|> curt@cynic.uucp         | all systems in the network is not known anywhere."
|> curt@cynic.wimsey.bc.ca |    --Berkeley Mail Reference Manual (Kurt Schoens)

--
Jefferson Ogata        	        ogata@cs.umd.edu
University Of Maryland          Department of Computer Science