🚀 go-pugleaf

>(Also on Studio Session, he pointed out how he got 6 voices -
>and could have had 8 - when the standard Sound Driver is only
>4.  The standard Sound Driver uses a 22 kHz sampling rate,
>but the -3 dB point of the electronics is 8 kHz, so most of
>it is wasted; Steve said he used an 11 kHz rate instead.)

	An interesting note since people have been discussing sound a bit:

A good rule to follow is to sample at double the maximum frequency you will
you will ever output.  In actuallity, you want to go a little faster.  Any 
frequencies above MaxF/2 create what is called 'aliasing'.  This is why CD's
sample at such a high frequency.

Aliasing is considered a worse problem then 'bits'.  If you had the choice
between 8 bits at 44Khz and 16 bits at 10Khz, the 8 bit sound would be
audibly better (*much* better).

The standard method to remove aliasing when sampling at low frequencies is
to have an anti-aliasing filter.  The implementation of such a filter 
digitially usually requires a 2:1 or 4:1 oversample (which defeats the purpose
in many applications).  Thus, such filters are usually analog and placed 
after the DAC.  With such a filter installed, you can easily output frequencies
at MaxF/2 (though it is still about 3Db down, I believe), and frequencies
above that don't cause as much distortion as they do otherwise.

Sound programs can design around the aliasing problem by carefully calculating
the waveforms of the instruments.  Voice and digitized (anything) almost
require the filter if you are not sampling at a high enough rate.

Antialiasing filters are low-pass filters, BTW.

				-Matt