Please write out for me an algorithm (assumably based on Fast Fourier
Transform) which will derive all possible information about frequency
content, from blocks of digital audio samples such as are found in a
.WAV file or on an audio CD. The algorithm needs to be couched in
terms from which a computer program can be written in C, and it needs
to specify the values of any fixed factors and give valid ranges for
any variable factors. It must assume no mathematical ability on the
part of the programmer, other than use of the standard mathematical
operators in the C language and the standard libm functions. Any
concerns such as ill-conditioning, or the reliability or validity of
results should be described in detail. The algorithm should produce
the correct results when correctly implemented!

You might want to take a look at Numerical Recipes in C... It has lots
of these types of solutions.

     -welte-ga