🚀 go-pugleaf

In article <2757@phred.UUCP> jefft@phred.UUCP (Jeff Taylor) writes:
>
>Linear Interpolation introduces disortion (changes the spectrum).
>[which can be componsated for in a digital filter].  Adding zeros does not.
>(plus the calculations are so simple).  

This explains how digital mixing works.  My first musings on digital
mixing involved using addition to mimic analog op-amp mixers.  This
method has the same drawbacks as analog mixing, because you must now be
concerned with signal overload due to wave peaks coinciding.  In the
analog world the voltage sum reaches the supply limits, and in the
digital realm the values overflow the arithmetic unit.

But the first digital sampler circuitry I examined utilized time
multiplexing of the signals.  My first reaction was that this was a cheap
solution full of distortion from unwanted high frequencies.  But
reviewing your post, I see that time multiplexing N signals is equivalent
to oversampling each channel by N times and then adding them (where
adding zero is a no-op).  The only difference is that each channel is
shifted in time.  This method of digital mixing of multiple sample
channels obviously requires that you have a constant sampling rate for
all channels, and that you follow the combined output with an appropriate
digital filter.

I hope I wasn't too far off on that, since I don't practice DSP math.

>What follows is a hand waving (no math) justification on why this is logical
>(although it defies common sense). (posted about 3 years ago to rec.audio)

Thanks for a very informative (and surprisingly understandable) post.
Your explaination might defy common sense, but I still can't help but
examine the results of zero-filling intuitively.  I.e. following the
zero-filled, over-sampled data with a digital filtering algorithm tends
to smooth out the data so that ideally you have a wave which looks very
similar to a linearly interpolated wave (to the eye), but has less
distortion from errors in approximation.

Brian Willoughby
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