forum.alglib.net

Hi. I have a digital signal with a sampling rate of 8000 per second. I want filter the signal to leave it in the desired frequency band. I want to implement like this: convert the signal from a function alglib.fftr1d (), equate to zero all the values that lie outside the desired frequency range, and the inverse transformation by alglib.fftr1dinv (). The problem: where to use the sample rate per second, because depending on how fast to play the signal the frequency is changed. Thanks.

hi, can you help me to implement the filter before do fftr1dinv(). I dont know how to do it with complex numbers.

Solved my own problem while no one had yet replied. Here is my solution, for anyone who might find it useful:
For information about my particular linux distribution, see below (I do not have administrative rights on it).