I'm writing an audio analysis program and need to do some FFT to frames of the data. I've got some code (rather verbose so I'll leave out the details) that successfully performs the shuffling of the data according to the Danielson-Lanczos Lemma through bit reversal of indices. For example the sequence of data 0,1,2,3,4,5,6,7 becomes 0,2,4,6,1,3,5,7 (though I'm working with a larger window than 8 samples!)
The problem is that I'm a little confused on what to do with the resultant vector. I'm just completely lost. I've read something about doing the transform on pairs of data points and then combining them. So would I apply the DFT to 0 and 2, then add the resulting vector with that of the DFT of 4 and 6 (using the above example data points) and add that to (DFT(1,3) + DFT(5,7))?
If it matters, I'm dealing with just the right channel of a wav file, so the signal has no imaginary part.
(I'm developing in Unity with C# and the only libraries j could find are .Net 4.0, and Unity only supports 3.5 right now. Plus I think it's interesting, if not confusing!)
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