Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers?
Particularly, I want to understand
- What mathematical features (e.g. monotonicity of coefficients) of filters are suited for real-time modulated filters?
- How to analyze filters for the required features?
But I want to also understand things such as "filter instability" and the ways they are implemented in real-time (does one apply them in buffers or what?)?
Dynamic recalculation: That the filter's cutoff and resonance can be recalculated on the fly. Or that the parameters can be adjusted while the filter is running. W/o the filter "blowing up" or something else.