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> how can a single sample represent the sounds
It can't. Sound = Changing (usually) air preasure, perceived by ear. Loudspeakers cause changing air preasure by membrane movement caused by electric current changes caused by amplifier ... .
If you make a curve of this electric current change, cut this curve in equal "time-slices" and measure the current for one of those slices you get a sample value.
Every sound consists of and can be transformed mathematically into a combination of sinus waves. A sinus wave is defined by frequency, amplitude and phase (delay). To get those 3 variables for 1 sinus function, you need 3 sample values. So 1 sample value contains no audible information.
One instrument (or a single tone played by 1 instrument) contains several sinus waves that are added, a lot of these are added again and result in the signal (it's the same e.g. in analog vinyl recordings, just not cut into slices with discrete sample values).
> how does increasing the sampling rate let you hear more high range instruments like a high hat?
Nyquist Limit: The maximum frequency that can be represented by digital signals is 1/2 of the sampling rate. The higher the sampling rate the higher the frequencies that can be included in the signal.
You have to write a paper? Some keywords for google: "Nyquist Limit", "A/D conversion" (or "analog digital conversion"), "fourier transformation", "cosinus transformation", "sampling rate", "psychoaccoustics" ... (I guess you've already searched for things like "sound", "digital audio" before you asked this here.)
This message has been edited since posting. Last time this message was edited on 10. March 2003 @ 06:55
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