Thank you very much.
I can understand why ‘zeroing the result of the FFT’ produces the result: I think that is the first time in signal-processing that the how-to seems so straight forward.
I have your code up and running with my data (collected at 50 Hz) and it works with range 0..5 and so on.
If I might take advantage of your expertise with a question:
The lower I set the second frequency (high pass?) the more the filtered signal deviates from the unfiltered signal at the signal start and end.
Am I correct in thinking that this is related to the fact that the signal I pass to the DFT is actually the combination of a unit impulse for the duration of the signal and the signal itself?
If I am correct why does the filtered signal start and end deviation from the unfiltered signal decrease as higher frequencies are introduced?
Many thanks once again.