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From: rammya.tv on 9 Feb 2010 23:08 hi.... i found w0 is not clear in my previous post, sorry i'm resending my queries again. i'm currently working on Analog Devices Dsp processor ADAU 1701.. Sigma studio is the software we are now on,in that i have some queries in the way they calculate filter coefficient.  main specification i needed to design a filter 1)Type[LPF,HPF, etc] 2)frequency 3)Q 4)Gain eg: I want a 2nd order LPF with Frequency= 1000 Q=0.5 Gain=1 The given below derivation is the one which i got from sigma studio help window. >w0 = 2*pi*f0/Fs >gainLinear = 10^(gain/20) >Lowpass >Transfer Function >H(s)=1/(s^2+(s/Q)+1) >Coefficients >alpha = sin(w0)/(2*Q) >a0 = 1 + alpha >a1 = 2*cos(w0) >a2 = 1  alpha >b0 = (1  cos(w0)) * gainLinear / 2 >b1 = 1  cos(w0) * gainLinear >b2 = (1  cos(w0)) * gainLinear / 2 After compiling we'll get coefficients a1,a2,b0,b1,b2 in hex format in CAPTURE WINDOW of software. i did manual calculation using the expression provided above and compare with that of coeffients of software, but it differs. Please help me to get detail calculation with the specification i given above. We know w0 = 2*pi*f0/Fs what is the value of pi(180 0r 3.14) w0 is the angular representation of requency i think it's 180 then in above value of alpha will be zero always. it means Q doesnt have any importantce in filter design please suggest a good book which describe in detail about filter design with regards rammya
From: Jerry Avins on 9 Feb 2010 23:36 rammya.tv wrote: > hi.... > i found w0 is not clear in my previous post, > sorry > i'm resending my queries again. > > i'm currently working on Analog Devices Dsp processor ADAU 1701.. > Sigma studio is the software we are now on,in that i have some queries in > the way they calculate filter coefficient. > >  main specification i needed to design a filter > 1)Type[LPF,HPF, etc] > 2)frequency > 3)Q > 4)Gain > > eg: > I want a 2nd order LPF with > Frequency= 1000 > Q=0.5 > Gain=1 > > The given below derivation is the one which i got from sigma studio help > window. > > >w0 = 2*pi*f0/Fs Are ypu sure this isn't w0 = 2*pi*f0/2*pi*Fs ? > >gainLinear = 10^(gain/20) > > >Lowpass > > >Transfer Function > >H(s)=1/(s^2+(s/Q)+1) > > >Coefficients > >alpha = sin(w0)/(2*Q) > >a0 = 1 + alpha > >a1 = 2*cos(w0) > >a2 = 1  alpha > >b0 = (1  cos(w0)) * gainLinear / 2 > >b1 = 1  cos(w0) * gainLinear > >b2 = (1  cos(w0)) * gainLinear / 2 > After compiling we'll get coefficients a1,a2,b0,b1,b2 in hex format in > CAPTURE WINDOW of software. > i did manual calculation using the expression provided above and compare > with that of coeffients of software, but it differs. > Please help me to get detail calculation with the specification i given > above. > > We know w0 = 2*pi*f0/Fs > what is the value of pi(180 0r 3.14) > w0 is the angular representation of requency i think it's 180 > then in above value of alpha will be zero always. > it means Q doesnt have any importantce in filter design > please suggest a good book which describe in detail about filter design Check with http://www.musicdsp.org/files/AudioEQCookbook.txt Jerry  Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������

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