Second Order, Two Pole, Two Zero filter using the
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