Kaiser Filter | Algorithm Design | Forum

Hi all,

Not sure if this is the right section of the forum to post this, feel free to move it if it's not.

I'm attempting to create an FIR filter based on the Kaiser algorithm. The Kaiser algorithm, in a few words, takes 3 parameters, the attenuation frequency, the pass frequency, and the attenuation amount in dB. It then generates a list of coefficients (impulses) with which to build a filter. 

I have implemented the Kaiser filter successfully in a standard C++ program. The program takes a sound file, filters it based on hardcoded (or prompted) parameters, and returns an output file. What I want to do now is find a way to implement this process in RackAFX. 

Something I am wondering though, is that since the coefficient list is based on parameters which will (in plugin form) be changed in real time, will this be too computationally intensive or slow to recalculate the entire list every time a UI control changes? 

This is the first plugin I have attempted to create that isn't wrote out step by step in the FX book, so i'm also a bit confused on where I should be placing my functions and variables in the framework. 

Is there a way to translate the C++ program I have now into something that will work in RackAFX? I can upload that program if it's helpful.

Any help would be majorly appreciated. If i'm totally going about this the wrong way, please let me know! And if there's any other information I need to provide, I would be happy to. 

Thank you for reading if you've made it this far!

Aidan
Student at The Evergreen State College, Olympia, Washington