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 Post subject: The digital filter
PostPosted: Wed May 02, 2012 12:42 pm 
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Joined: Mon Apr 09, 2012 9:53 pm
Posts: 1
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.


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 Post subject: Re: The digital filter
PostPosted: Thu Jul 17, 2014 11:19 am 
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Joined: Thu Jul 17, 2014 11:16 am
Posts: 3
hi, can you help me to implement the filter before do fftr1dinv(). I dont know how to do it with complex numbers.


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 Post subject: Re: The digital filter
PostPosted: Wed Apr 08, 2015 10:30 am 
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Joined: Wed Apr 08, 2015 10:11 am
Posts: 1
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).


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