# Type : LPF 24dB/Oct¶

notes
```The filter tends to be unsable for low frequencies in the way, that it seems to flutter,
but it never explode. At least here it doesn't.
```
code
 ``` 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28``` ```First calculate the prewarped digital frequency: K = tan(Pi * Frequency / Samplerate); K2 = K*K; // speed improvement Then calc the digital filter coefficients: A0 = ((((105*K + 105)*K + 45)*K + 10)*K + 1); A1 = -( ((420*K + 210)*K2 - 20)*K - 4)*t; A2 = -( (630*K2 - 90)*K2 + 6)*t; A3 = -( ((420*K - 210)*K2 + 20)*K - 4)*t; A4 = -((((105*K - 105)*K + 45)*K - 10)*K + 1)*t; B0 = 105*K2*K2; B1 = 420*K2*K2; B2 = 630*K2*K2; B3 = 420*K2*K2; B4 = 105*K2*K2; Per sample calculate: Output = B0*Input + State0; State0 = B1*Input + A1/A0*Output + State1; State1 = B2*Input + A2/A0*Output + State2; State2 = B3*Input + A3/A0*Output + State3; State3 = B4*Input + A4/A0*Output; For high speed substitude A1/A0 with A1' = A1/A0... ```

```It turns out, that the filter is only unstable if the coefficient/state precision isn't high enough. Using double instead of single precision already makes it a lot more stable.
```Just found out, that I forgot to remove the temporary variable 't'. It was used in my code for the speedup. You can simply ignore and delete it.
```Changing the frequency also seems to affect the volume quite a lot. That can't be right? Maybe you should re-post this (and remove the "t" this time)?  ;)
```Your filter does not work! Arise overload in the calculation of coefficients A and B in the cutoff frequency of approximately 10 khz. At low frequencies, the coefficient of the gain increases proportional to the frequency cutoff, which causes congestion. I also noticed that the filter on this basis, the package of ASIO/DSP - DDspBesselFilter.pas does not work, although Butterworth (DDspButterworthFilter.pas) and Chebyshev (DDspChebyshevFilter.pas) work perfectly!