(I am trying to create an IIR audio filter that adds reverb to an initial sample)
Say I designed an analog filter to model acoustic attenuation based on the following mathematical model:
$$ I = I_0 e^{pt}, $$
Where $p$ is some constant such that $-1 \leq p < 0$.
In the laplace domain, this is simply
$$ \mathfrak{L}\{I\} = \dfrac{1}{s-p} $$
Using the impulse invariant transform, I get the digital equivalent
$$ Z\{I\} = \dfrac{z}{z-e^{pT_s}}, $$
Which inherently has one pole.
How do I add more resolution to this filter? Implementing this as a digital filter makes my audio sample sound like garbage. Would adding arbitrary taps keeps it from modeling the original analog decay?
Using experimental attenuation data and frequency sampling for an FIR equivalent, I can easily obtain 50000 poles and create a very clear reverb-adding filter.
Both IIR and FIR methods were tested in MATLAB, with the FIR using MATLAB's built-in church impulse response.
(Sorry if the question is unclear, I'm a bit new at this.)