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.
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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...

Comments

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!