Second Order, Two Pole, Two Zero filter using the following formula:
y(n) = 2.0 * (A0 * x(n) + A1 * x(n - 1) + A2 * x(n - 2) - B1 * y(n - 1) - B2 * y(n - 2))
where y(n) is Output, x(n) is Input, x(n-1) is a delayed copy of the input, and y(n-1) is a
delayed copy of the output. This filter is a recursive IIR or Infinite Impulse Response filter.
It can be unstable depending on the values of the coefficients. This filter is basically the same
as the FilterBiquad with different ports. A thorough description of the digital filter theory
needed to fully describe this filter is beyond the scope of this document. Calculating
coefficients is non-intuitive; the interested user is referred to one of the standard texts on
filter theory (e.g., Moore, "Elements of Computer Music", section 2.4). Special thanks to Robert
Bristow-Johnson for contributing his filter equations to the music-dsp list. They were used for
calculating the coefficients for the lowPass, highPass, and other parametric filter calculations.