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| The digital filter http://forum.alglib.net/viewtopic.php?f=2&t=574 |
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| Author: | slava_32 [ Wed May 02, 2012 12:42 pm ] |
| Post subject: | The digital filter |
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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| Author: | mig00 [ Thu Jul 17, 2014 11:19 am ] |
| Post subject: | Re: The digital filter |
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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| Author: | Algernon [ Wed Apr 08, 2015 10:30 am ] |
| Post subject: | Re: The digital filter |
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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