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Sample Frequency of Low Pass filter VI is ambiguous

I am having trouble understanding how the Sampling Frequency of the Low Pass Filter actually impacts the function buried within. The information passed in is an array with no relavance to time. How does the function actually sample this array of data at a frequency?

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The best thing I can come up with is that the IIR Filter in the low-level VI is building the waveform from the values in the array by assigning them a dt = 1/Sampling Freq. Does that sound right?

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I doubt that they actually build a waveform. Here is my guess: By defining a sample frequency they can get the sampling interval dt as you indicated.  By defining a sampling frequency the cutoff frequencies can be scaled to frequency units (Hz), making the VI easier to use.



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