Radium said:
I thought sampling rate only applied to digital audio (PCM, MPEG, WMA,
etc.)
FM synths are digital. There's no theoretical reason why they could not be
analog, though, other than that it would not be practical. After all, you
can simply use the output of one VCO to modulate the pitch of another and
that's called FM in the analog world (sometimes also called cross modulation
for some reason).
FM is a lot more practical digitally because it is easy to generate a number
of sine waves and implement things like different operator configurations,
patch memory etc...
Even after converting to analog, the FM signal would have to be
Ummm, I don't get that. Aren't you talking something about FM radio
technology here? In FM synthesis, though I'm no guru, it's about sine waves
modulating each others frequencies so that extra harmonics are generated.
The result coming out is "ordinary" audio just as sampled sounds are.
You are right, the digital representation is lowpass filtered in the DAC.
Basically they remove all the components above the Nyquist that would cause
aliasing in a given sampling rate (anti-alias filter.
Aliasing is when you try to say record components that are higher than the
Nyquist. In that case, in stead of getting the frequency n Hz above the
Nyquist, you get it's alias n Hz below.
Would the "sampling rate" of the FM driver determine the frequency
response of the chip-based FM synth?
I don't know anything about the sampling rates of FM chips. I guess the rate
at which data is actually processed in the hardware determines the frequency
response (the highest frequency you can represent in the synthesis phase).
Even if some software drivers did resampling adfter the synthesis in
hardware, it would not generate extra precision, frequencies or quality to
the sound.
It's the same with digital images. If you have a 320x200 bitmap, scaling
that to 640x480 does not bring in any new detail.
