[music-dsp] Bandpass IIR Gain Factor
developer at netcologne.de
Wed Apr 1 17:08:36 EDT 2009
after spending some hours on recalculating the gain factor with your
hint, watching the formulas growing I suddenly knew, this can't be the
right way - can't be, that a filter formula has a frequency dependent
gain! Went back to bilinear transformation theory and found the mistake.
There's been one erroneous line in my code:
m_fA1 = 4 * fDQ * fDiv;// from my first posting
m_fA1 = fDQ *fFd * fDiv;
now erverything works fine.
Vesa Norilo schrieb:
> Thomas Rehaag wrote:
> > Hi Vesa,
> > so the equiation is:
> > H(Z) = (a1 + a3 z^-2) / ( 1 + b2 z^-1 + b3 z^-2 )
> > = (a1 - a1 z^-2) / ( 1 + b2 z^-1 + b3 z^-2 )
> > = (a1 z^2 - a1) / ( z^2 + b2 z^1 + b3)
> > and I substitute Z = e^(i fRel Pi)
> > ?
> Yes, that's correct.
> The magnitude response of the filter is the absolute value of H(Z), and
> to get the point of Z-plane that corresponds to a real world frequency
> f/samplerate, you query H(e^i f pi). Absolute value is the magnitude and
> the argument is the phase response.
> A helpful identity with the absolute value of complex numbers is:
> |Z| = Sqrt(Z * Conjugate(Z))
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