I've seen the various FFT questions on here but I'm confused on part of the implementation. Instead of performing the FFT in real time, I want to do it offline. Lets say I have the raw data in float[] audio. The sampling rate is 44100 and so audio[0] to audio[44099] will contain 1 seconds worth of audio. If my FFT function handles the windowing (e.g. Hanning), do I simply put the entire audio buffer into the function in one go? Or, do I have to cut the audio into chunks of 4096 (my window size) and then input that into the FFT which will then perform the windowing function on top?
You may need to copy your input data to a separate buffer and get it in the correct format, e.g. if your FFT is in-place, or if it requires interleaved complex data (real/imaginary). However if your FFT routine can take a purely real input and is not in-place (i.e. non-destructive) then you may just be able to pass a pointer to the original sample data, along with an appropriate size parameter.
Typically for 1s of audio, e.g. speech or music, you would pick an FFT size which corresponds to a reasonably stationary chunk of audio, e.g. 10 ms or 20 ms. So at 44.1 kHz your FFT size might be say 512 or 1024. You would then generate successive spectra by advancing through your buffer and doing a new FFT at each starting point. Note that it's common practice to overlap these successive buffers, typically by 50%. So if N = 1024 your first FFT would be for samples 0..1023, your second would be for samples 512..1535, then 1024..2047, etc.