Technical discussions related to Audio Signal Processing (digital effects, acoustics, noise reduction, musical signal processing, etc).
hi.... 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. ->Ï0 = 2*pi*f0/Fs ->gainLinear = 10^(gain/20) ->Lowpass ->Transfer Function ->H(s)=1/(s^2+(s/Q)+1) ->Coefficients ->alpha = sin(Ï0)/(2*Q) ->a0 = 1 + alpha ->a1 = -2*cos(Ï0) ->a2 = 1 - alpha ->b0 = (1 - cos(Ï0)) * gainLinear / 2 ->b1 = 1 - cos(Ï0) * gainLinear ->b2 = (1 - cos(Ï0)) * 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 Ï0 = 2*pi*f0/Fs -what is the value of pi(180 0r 3.14) -Ï0 is the angular representation of requency i think it will be 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______________________________