# Peak/Notch filter¶

**Author or source:**ed.bew@raebybot**Type:**peak/notch**Created:**2002-12-16 19:01:28

```
// Peak/Notch filter
// I don't know anymore where this came from, just found it on
// my hard drive :-)
// Seems to be a peak/notch filter with adjustable slope
// steepness, though slope gets rather wide the lower the
// frequency is.
// "cut" and "steep" range is from 0..1
// Try to feed it with white noise, then the peak output does
// rather well eliminate all other frequencies except the given
// frequency in higher frequency ranges.
```

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 | ```
var f,r:single;
outp,outp1,outp2:single; // init these with 0!
const p4=1.0e-24; // Pentium 4 denormal problem elimination
function PeakNotch(inp,cut,steep:single;ftype:integer):single;
begin
r:=steep*0.99609375;
f:=cos(pi*cut);
a0:=(1-r)*sqrt(r*(r-4*(f*f)+2)+1);
b1:=2*f*r;
b2:=-(r*r);
outp:=a0*inp+b1*outp1+b2*outp2+p4;
outp2:=outp1;
outp1:=outp;
if ftype=0 then
result:=outp //peak
else
result:=inp-outp; //notch
end;
``` |

## Comments¶

**Date**: 2010-03-02 03:19:21**By**: slo77y (at) yahoo.de

```
this code sounds bitcrushed like hell translated to c++, any suggestions ?
float pi = 3.141592654;
float r = dQFactor*0.99609375;
float f = cos(pi*iFreq);
float a0 = (1-r) * sqrt ( r * ( r-4 * ( f * f ) + 2 ) + 1 );
float b1 = 2 * f * r;
float b2 = - ( r * r );
float outp = 0.0, outp1 = 0.0, outp2 = 0.0;
for (i = 0; i < iSamples; i++)
{
float inp = fInput[i];
outp = a0 * inp + b1 * outp1 + b2 * outp2 + p4;
outp2 = outp1;
outp1 = outp;
fOutput[i] = (inp-outp); //notch
}
```

**Date**: 2012-05-05 08:22:08**By**: ten.xoc@53namhsima

```
After about 3 hours wondering why I was getting back the original un-altered audio, I finally got this version of a keeper filter, which I used with absurdly good success on a power grid comb filter. When the power grid filter was fed with audio from a lamp cord with one 1 Megohm resistor on each prong, all sorts of cool sounds become audio when the output is amplified 40 dB. For wall cord audio, use 60.0 for the cutoff.
---the function is below---
double keeper_1(double input, double cutoff,double rate,double *magnitude)
{
const double steepness=1.0;
const double p4=1.0e-24;
static unsigned char first=1;
static double nfreq=0.1;
static double old_cutoff=0.0;
static double the_magnitude=0;
static double average=0.0;
static int average_count=0;
static double a=0.0;
static double r=0.0;
static double coeff=0.0;
static double delay[3]={0.0,0.0,0.0};
static double delay1[3]={0.0,0.0,0.0};
static double delay2[3]={0.0,0.0,0.0};
static double delay3[3]={0.0,0.0,0.0};
static double b[3]={0.0,0.0,0.0};
if(first==1 || cutoff!=old_cutoff )
{
r=steepness * 0.99609375;
nfreq=(cutoff/(double)rate) * 2.0 ;
coeff= cos( M_PI * nfreq);
a=(1.0 - r) * sqrt(r * (r - 4 * (coeff * coeff) + 2) +1);
b[1]=2 * coeff * r;
b[2]=-(r * r);
first=0;
}
delay3[0] = a * input + b[1] * delay3[1] + b[2] * delay3[2] + p4;
delay3[2]=delay3[1];
delay3[1]=delay3[0];
delay2[0] = a * delay3[0] + b[1] * delay2[1] + b[2] * delay2[2] + p4;
delay2[2]=delay2[1];
delay2[1]=delay2[0];
delay1[0] = a * delay2[0] + b[1] * delay1[1] + b[2] * delay1[2] + p4;
delay1[2]=delay1[1];
delay1[1]=delay1[0];
delay[0] = a * delay1[0] + b[1] * delay[1] + b[2] * delay[2] + p4;
delay[2]=delay[1];
delay[1]=delay[0];
average+=delay[0];
average_count++;
if(average_count>dft_size-1)
{
double aver=average/(double)dft_size;
the_magnitude=sqrt(aver * aver); /* we're only interested in the root mean square */
average=0.0;
average_count=0;
}
magnitude[0]=the_magnitude;
old_cutoff=cutoff;
return delay[0];
}
```