🚀 go-pugleaf

From: matthews@eecs.ucdavis.edu (Thomas W. Matthews)
 
>In my application (not audio), I care only about the value of the input
>waveform at the sample times.  I will not be attempting to reconstruct the
>waveform BETWEEN the sample times.  Hence, the input signal need not be
>bandlimited to half the sampling frequency.  
 
You CAN do that. But what you get when you look at the freq. domain you'll see 
all aliases and you can't tell them apart from the freq. below fs/2. I suspect
the sampling rate you are using is much higher than 2 times the highest freq 
contained in the signal.
 
>However, it is the goal of audio devices to reconstruct the input waveform
>from the samples, including the part of the waveform between the samples.
 
Correct. You can recover the waveform between samples almost perfectly if you 
filter out enough HF garbage.
 
>I have heard it said that digital leaves out part of the sound
>(the part between the sample times). 
 
That's argument that analog "golden ears" make without knowing enough about 
sampling theory. They think the sampling process throws some data between 
samples away in the 20-20kHz bandwidth.
 
>these can be mitigated by
>increasing the sampling frequency (to decrease required filter complexity) or
>by doing more of ther filtering in the digital domain (certain types of
>filter responses are possible there). 
 
You cannot solve the problem of aliasing with digital filter. Once you've gone 
from analog to digital, the aliases are already there and you cannot tell 
which is which if you use too low a sampling rate.
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Kong Kritayakirana  (kong@leland.stanford.edu)
Remotely reading and posting to rec.audio from
a nice beach somewhere in south pacific.  Damn
the technology that helps  us to get in touch.
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