Content-wise, you also need to consider what the meaning is of
the terminology "anti aliased"; technically, it means you prevent
the double interpretation of wave partials, which is usually associated
with AD or DA conversion, as in that high frequency components that are
put on a AD converter will be "aliased" because they will mirror around
the Nyquist rate and become indistinguishable from a similar frequency in the
input signal.

A more common thing to think about in softwaere waveform synthesis, apart
from this principle but then in the "virtual" sampling of a waveform,, is
to consider the error an actual DAC (digital to analog converter, like your
sound card has), as compared with a "perfect reconstructor", which would take
your properly bandwidth limited signal (to prevent aliasing) and (given a very
long latency) turn it into a perfect output signal from you sound card.

The DAC in you soundcard will not do this job perfectly, even if you're
perfectly anti-aliasing or bandwidth limiting your digital signal. That's 
of the sampling reconstruction theorem's need for a very long filter, while
and actual DAC has a very short reconstruction filter.

One of the effects of this limitation is probably the most important to consider
for musical instruments producing sound which will be amplified into the higher 
domains: mid frequency blare. Especially in highly resonant spaces, like those 
un-damped parallel reflective walls, certain sound wave patterns tend to 
amplify through
reverberation causing a lot of clutter in the sensitive range of the human 
hearing, the
middle frequencies (lets say 1000 through 4000 Hz). This "blare" becomes louder 
of various digital processing and DAC reconstruction ensemble effects, and 
should be controlled.

So especially for serious "live" music reproduction, measures ought to be in 
place to
control the amount (and kind) of blare your software isntrument produces, 
probably as
higher priority than the exact type and amount of "anti-aliasing" you provide.

Theo V.
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