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Simple low-pass filter from scratch in Core

Discussion in 'Building With Reaktor' started by Aleksandr Smirnov, Feb 1, 2009.

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  1. Aleksandr Smirnov

    Aleksandr Smirnov NI Product Owner

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    1,539
    Hello!

    This is another newbie question, but I'd like to learn Reaktor step by step making my own things in core (filters, envelopes, etc.). I'm already familiar with OSC and LFO (thanks herw and this thread http://www.native-instruments.com/forum/showthread.php?t=77977), but I'd like to go deeper by making my own filter now.

    I've found this very useful book (or tutorial) about filters:

    http://www.dsprelated.com/dspbooks/filters/Simplest_Lowpass_Filter.html

    I'd like to start by building simple lowpass filter with say Hz knob and any audio input (saw osc or whatever). Is it a good point to start with?

    I've read this article very carefully and tried to make it in Core, but nothing came out. Actually I have some questions I haven't figure out yet.

    Please, take a look at this article and formula (attached picture):

    http://www.dsprelated.com/dspbooks/filters/Definition_Simplest_Low_Pass.html

    In article it is said that T is the sampling interval. Is it the same as SR.C in seconds. What values exactly should I use for n and for T? What is the main tip to start with?

    Any help will be much appreciated!
     

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  2. Aleksandr Smirnov

    Aleksandr Smirnov NI Product Owner

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    BTW, I've looked inside Vadim's tutorial about filter design, but I still can't get the main principle of building it (also looked through the uploaded files in UL). I understand what to do theoretically, but practically to make it in Core is hard for me.
     
  3. _nico

    _nico NI Product Owner

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    172
    Although I have put my DSP studies on old for the time being, it is still of great interest of mine.
    From what I understand.
    If you read further down the article it says that for DSP purposes T is usually set to 1.
    n is the sample number.
    so your equation is y(n) = x(n) + (n-1)x

    x(n) is the filter input amplitude at time (or sample) n, and y(n) is the output amplitude at time n.

    After that I'm lost. Good luck!
     

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  4. Aleksandr Smirnov

    Aleksandr Smirnov NI Product Owner

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    1,539
    Thanks for reply. Yes, I also did that before (and it seems to look like simple highpass filter tho), but how can I adjust frequency to this formula? Say I want only some knob in Hz from 20 Hz to 20 KHz and I want to cut those frequencies with this button.

    I surely understand I need somehow to divide sample into frequencies.
     
  5. Vadim @ NI

    Vadim @ NI NI Team NI Team

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    280
    Check Wikipedia for the "Lowpass filter". On the first look seems to be a decent explanation.
     
  6. Aleksandr Smirnov

    Aleksandr Smirnov NI Product Owner

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    1,539
    Thank you, I'll take a look. ;)
     
  7. Aleksandr Smirnov

    Aleksandr Smirnov NI Product Owner

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    1,539
    I still didn't figure out how to make it, seems like I'm missing something.
     
  8. Vadim @ NI

    Vadim @ NI NI Team NI Team

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    280
    The filter in Wiki happens to be exactly the one described in the Core manual. What exactly don't you understand?
     
  9. Aleksandr Smirnov

    Aleksandr Smirnov NI Product Owner

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    1,539
    I fotgot it was in Core manual, sorry. But anyway I was trying to make it myself from scratch without using manuals, only tips and maths. I will look at Core manual once more later this evening. Thanks.
     

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