Filters
Notch 
Notch Filter 
Band Pass 
Band Pass Filter 
High Pass 
High Pass Filter 
Low Pass 
Low Pass Filter 
High Shelf 
High Shelf Filter 
Low Shelf 
Low Shelf Filter 
FFT 
FFT Filter 
Peak EQ 
Peak EQ Filter 
About the audio filters (this section is for the advanced users):
All filter transfer functions were derived from analog prototypes (that
are shown below for each filter type) and had been digitized using the
Bilinear Transform (Waveview). BLT frequency warping has been taken into account
for both significant frequency relocation and for bandwidth readjustment.
First, given a biquad transfer function defined as:
b0 + b1*z^1 + b2*z^2
H(z) =
a0 + a1*z^1 + a2*z^2
This shows 6 coefficients instead of 5 so, depending on your architecture,
you will likely normalize a0 to be 1 and perhaps also b0 to 1 (and collect
that into an overall gain coefficient). Then your transfer function would
look like:
(b0/a0) + (b1/a0)*z^1 + (b2/a0)*z^2
H(z) =
1 + (a1/a0)*z^1 + (a2/a0)*z^2
or
1 + (b1/b0)*z^1 + (b2/b0)*z^2
H(z) = (b0/a0) *
1 + (a1/a0)*z^1 + (a2/a0)*z^2
The most straight forward implementation would be the Direct I form (second equation):
y[n] = (b0/a0)*x[n] + (b1/a0)*x[n1] + (b2/a0)*x[n2]  (a1/a0)*y[n1]  (a2/a0)*y[n2]
This is probably both the best and the easiest method to implement in the 56K.
Now, given:
sampleRate (the sampling frequency)
frequency ("wherever it's happenin', man." "center" frequency
or "corner" (3 dB) frequency, or shelf midpoint frequency,
depending on which filter type)
dBgain (used only for peaking and shelving filters)
bandwidth in octaves (between 3 dB frequencies for BPF and notch
or between midpoint (dBgain/2) gain frequencies for peaking EQ)
_or_ Q (the Electrical Engineering kind of definition)
_or_ S, a "shelf slope" parameter (for shelving EQ only). when S = 1,
the shelf slope is as steep as it can be and remain monotonically
increasing or decreasing gain with frequency.
First compute a few intermediate variables:
A = sqrt[ 10^(dBgain/20) ]
= 10^(dBgain/40) (for peaking and shelving EQ filters only)
omega = 2*PI*frequency/sampleRate
sin = sin(omega)
cos = cos(omega)
alpha = sin/(2*Q) (if Q is specified)
= sin*sinh[ ln(2)/2 * bandwidth * omega/sin ] (if bandwidth is specified)
beta = sqrt[ (A^2 + 1)/S  (A1)^2 ] (for shelving EQ filters only)
Then compute the coefs for whichever filter type you want:
The analog prototypes are shown for normalized frequency.
The bilinear transform substitutes:
1 1  z^1
s < *
tan(omega/2) 1 + z^1
LPF: H(s) = 1 / (s2 + s/Q + 1)
b0 = (1  cos)/2
b1 = 1  cos
b2 = (1  cos)/2
a0 = 1 + alpha
a1 = 2*cos
a2 = 1  alpha
HPF: H(s) = s2 / (s2 + s/Q + 1)
b0 = (1 + cos)/2
b1 = (1 + cos)
b2 = (1 + cos)/2
a0 = 1 + alpha
a1 = 2*cos
a2 = 1  alpha
BPF (constant skirt gain): H(s) = s / (s2 + s/Q + 1)
b0 = Q*alpha
b1 = 0
b2 = Q*alpha
a0 = 1 + alpha
a1 = 2*cos
a2 = 1  alpha
BPF (constant peak gain): H(s) = (s/Q) / (s2 + s/Q + 1)
b0 = alpha
b1 = 0
b2 = alpha
a0 = 1 + alpha
a1 = 2*cos
a2 = 1  alpha
Notch: H(s) = (s2 + 1) / (s2 + s/Q + 1)
b0 = 1
b1 = 2*cos
b2 = 1
a0 = 1 + alpha
a1 = 2*cos
a2 = 1  alpha
PeakingEQ: H(s) = (s2 + s*(A/Q) + 1) / (s2 + s/(A*Q) + 1)
b0 = 1 + alpha*A
b1 = 2*cos
b2 = 1  alpha*A
a0 = 1 + alpha/A
a1 = 2*cos
a2 = 1  alpha/A
LowShelf: H(s) = A * (s2 + beta*s + A) / (A*s2 + beta*s + 1)
b0 = A*[ (A+1)  (A1)*cos + beta*sin ]
b1 = 2*A*[ (A1)  (A+1)*cos ]
b2 = A*[ (A+1)  (A1)*cos  beta*sin ]
a0 = (A+1) + (A1)*cos + beta*sin
a1 = 2*[ (A1) + (A+1)*cos ]
a2 = (A+1) + (A1)*cos  beta*sin
HighShelf: H(s) = A * (A*s2 + beta*s + 1) / (s2 + beta*s + A)
b0 = A*[ (A+1) + (A1)*cos + beta*sin ]
b1 = 2*A*[ (A1) + (A+1)*cos ]
b2 = A*[ (A+1) + (A1)*cos  beta*sin ]
a0 = (A+1)  (A1)*cos + beta*sin
a1 = 2*[ (A1)  (A+1)*cos ]
a2 = (A+1)  (A1)*cos  beta*sin
Music Editing Master
