Many applications that produce sound, such as software synthesizers, are able to apply a filter that varies with time, such as applying a low pass filter that varies with an LFO.
I currently have a means of calculating coefficients for an IIR filter given poles and zeros, but do not know how to achieve this varying without redoing the calculations from scratch each time it needs to change. Is there a good approach to this that does not rely on using only a first order filter?