IIR filter

# slider1 = Frequency
# slider2 = RIT
# slider3 = Tone
# slider4 = Volume

# set the samplerate to 48000 samples per second
samplerate = 48000

# down-convert to 1st IF:
# z is a sawtooch oscillator
# (ai,aq) form the mixer output
# (ini, inq) are the soundcard's inputs
z = mod1(z+slider1+slider2/40);
ai = ini * cos1(z) - inq*sin1(z)
aq = inq * cos1(z) + ini*sin1(z)

# filter IF:
# samplerate: 48000
# 8 pole low-pass Butterworth cutoff: 1800 Hz
# http://www-users.cs.york.ac.uk/~fisher/mkfilter
# I branch filter
ixv0 = ixv1
ixv1 = ixv2
ixv2 = ixv3
ixv3 = ixv4
ixv4 = ixv5
ixv5 = ixv6
ixv6 = ixv7
ixv7 = ixv8
ixv8 = ai / 4.758542880e+07
iyv0 = iyv1
iyv1 = iyv2
iyv2 = iyv3
iyv3 = iyv4
iyv4 = iyv5
iyv5 = iyv6
iyv6 = iyv7
iyv7 = iyv8
iyv8 =   (ixv0 + ixv8) + 8 * (ixv1 + ixv7) + 28 * (ixv2 + ixv6) + 56 * (ixv3 + ixv5) + 70 * ixv4 + ( -0.2977843725 * iyv0) + (  2.7418349049 * iyv1) + (-11.0754467360 * iyv2) + ( 25.6394142260 * iyv3) + (-37.2107924000 * iyv4) + ( 34.6753158910 * iyv5) + (-20.2650962730 * iyv6) + (  6.7925493799 * iyv7)
ai = iyv8


# Q branch filter
qxv0 = qxv1
qxv1 = qxv2
qxv2 = qxv3
qxv3 = qxv4
qxv4 = qxv5
qxv5 = qxv6
qxv6 = qxv7
qxv7 = qxv8
qxv8 = aq / 4.758542880e+07
qyv0 = qyv1
qyv1 = qyv2
qyv2 = qyv3
qyv3 = qyv4
qyv4 = qyv5
qyv5 = qyv6
qyv6 = qyv7
qyv7 = qyv8
qyv8 =   (qxv0 + qxv8) + 8 * (qxv1 + qxv7) + 28 * (qxv2 + qxv6) + 56 * (qxv3 + qxv5) + 70 * qxv4 + ( -0.2977843725 * qyv0) + (  2.7418349049 * qyv1) + (-11.0754467360 * qyv2) + ( 25.6394142260 * qyv3) + (-37.2107924000 * qyv4) + ( 34.6753158910 * qyv5) + (-20.2650962730 * qyv6) + (  6.7925493799 * qyv7)
aq = qyv8

# down-convert to audio using a second mixer
y = mod1(y+0.0275)
af = ai*cos1(y) - aq*sin1(y)
audio = audio - slider3*(audio - af)

# send audio to output of soundcard
# slider4 controls the volume
uitL = audio * slider4 * 500
uitR = audio * slider4 * 500