1. Do not open more than one forum discussion for the same issue. I have seen you have open 3 for this filter issue.
2. What do you expect to see after filtering the signal? You want the square signal? You should now that a square signal has an infinite frequency response. See this external link:
So if you make that filter you will have a sinewave (delta in frequency).
3. If you want to do it anyway introduce a higher order for the filter. Higher order high accuracy but more calculation time. See example IIR Filter Design (hep -> Find examples) and you can try how it works.
4. Do not introduce a waveform, as you can see the input for this funcion is an array of DBL. For waveforms (in LV 8.0) i recommend you to use the filter function. SIgnal processing -> Wfm Condition -> Filter
5. Visualize the FFT after the filter and not before and you will see how the filter works.
Regards,
Jaime Cabrera
NI Applications Engineering
Regards,
Jaime Cabrera
NI Applications Engineering Spain