Hello friends. I have an apparently unsolvable problem: I have a given IIR filter (they give me the poles and the zeros). The filter was originally sampled at a given (very high) frequency. I also have a signal, which is sampled at a much lower rate. When I filter my signal, I have to resample it at the same (high and unuseful) frequency of the filter, in order to have a correct filtering. I'd like instead to filter my signal at its original (low) rate. Is it possible to lower the frequency of an IIR (just repositioning the zeros and the poles)? If that's possible, I could filter my signal at the original rate, saving much computational time! Please, let me know if there's any solution ;) Bye Stefano 

how to resample a z transformation???
Started by ●November 28, 2002
Reply by ●November 29, 200220021129
Stefano I hate to ask this, but if you are able to consider changing the poles and zeros, then why can you not just design a new filter? Why all the concern? Just match the freq and phase response of the old one, but at the desired cutoff. Jeff Brower DSP sw/hw engineer Signalogic stefanosorrentino wrote: > > Hello friends. > I have an apparently unsolvable problem: > I have a given IIR filter (they give me the poles and the zeros). > The filter was originally sampled at a given (very high) frequency. > I also have a signal, which is sampled at a much lower rate. > When I filter my signal, I have to resample it at the same (high and > unuseful) frequency of the filter, in order to have a correct > filtering. > I'd like instead to filter my signal at its original (low) rate. > Is it possible to lower the frequency of an IIR (just repositioning > the zeros and the poles)? If that's possible, I could filter my > signal at the original rate, saving much computational time! > Please, let me know if there's any solution ;) > Bye > Stefano 

Reply by ●November 29, 200220021129
Stefano: you may also wish to look into decomposing your filter into polyphase components, if you haven't already. chatonda  Jeff Brower <> wrote: > Stefano > > I hate to ask this, but if you are able to consider changing the poles and zeros, > then why can you not just design a new filter? Why all the concern? Just match the > freq and phase response of the old one, but at the desired cutoff. > > Jeff Brower > DSP sw/hw engineer > Signalogic > stefanosorrentino wrote: > > > > Hello friends. > > I'd like instead to filter my signal at its original (low) rate. > > Is it possible to lower the frequency of an IIR (just repositioning > > the zeros and the poles)? If that's possible, I could filter my > > signal at the original rate, saving much computational time! > > Please, let me know if there's any solution ;) > > Bye > > Stefano > ===== __________________________________________________ 
Reply by ●November 30, 200220021130
My filter is a digitalization of an analog filter. I obtained it using a bilinear transformation. Using the bilinear, I need to work with low frequencies in order to have a correct response. This is the reason why I cannot just create the filter again with a lower sampling freq. Thanks anyway Stefano  Original Message  > I hate to ask this, but if you are able to consider changing the poles and zeros, > then why can you not just design a new filter? Why all the concern? Just match the > freq and phase response of the old one, but at the desired cutoff. > > Jeff Brower > DSP sw/hw engineer > Signalogic 
Reply by ●November 30, 200220021130
Stefano So obtain *another* digital filter from *another* analog filter. It's just filter design; it shouldn't be difficult. Create as many as you need during processing  the only cost is to store the filter coefficients and make your code be able to switch filters as/when needed. Jeff Brower DSP sw/hw engineer Signalogic Stefano Sorrentino wrote: > > My filter is a digitalization of an analog filter. I obtained it using a > bilinear transformation. > Using the bilinear, I need to work with low frequencies in order to have a > correct response. > This is the reason why I cannot just create the filter again with a lower > sampling freq. > Thanks anyway > Stefano > >  Original Message  > > I hate to ask this, but if you are able to consider changing the poles and > zeros, > > then why can you not just design a new filter? Why all the concern? Just > match the > > freq and phase response of the old one, but at the desired cutoff. > > > > Jeff Brower > > DSP sw/hw engineer > > Signalogic 
Reply by ●December 2, 200220021202
If you have the analog filter design, preferably laplacedomain numerator and denominator coefficients, then you can usually do better than a bilinear transform. As you mentioned, the bilinear transform is only accurate at low frequencies. However, the MATLAB control systems toolbox, if you have it, does a pretty good job of converting continuoustime filters to discretetime filters. See the function c2d (Control Systems toolbox), which has various transformation methods, and impinvar (Signal Processing Toolbox). You can try these different filter transformation methods to make your filter work at the lower sampling rate (assuming, of course, that there isn't an aliasing problem with implementing the filter at the lower sample rate). Naturally, you must check each of the discretetime filter designs. Your other option, still using the bilinear transform, is to make a "prewarped" analog filter (based on your desired filter) whose specifications have been moved slightly such that the bilinear transform will give the correct digital filter. Sincerely, Glen Ragan "Stefano Sorrentino" <> on 11/30/2002 06:43:14 AM To: "Jeff Brower" <> cc: (bcc: Glen Ragan/US/IO INC) Subject: Re: [matlab] how to resample a z transformation??? My filter is a digitalization of an analog filter. I obtained it using a bilinear transformation. Using the bilinear, I need to work with low frequencies in order to have a correct response. This is the reason why I cannot just create the filter again with a lower sampling freq. Thanks anyway Stefano  Original Message  > I hate to ask this, but if you are able to consider changing the poles and zeros, > then why can you not just design a new filter? Why all the concern? Just match the > freq and phase response of the old one, but at the desired cutoff. > > Jeff Brower > DSP sw/hw engineer > Signalogic _____________________________________ Note: If you do a simple "reply" with your email client, only the author of this message will receive your answer. You need to do a "reply all" if you want your answer to be distributed to the entire group. _____________________________________ About this discussion group: To Join: To Post: To Leave: Archives: http://www.yahoogroups.com/group/matlab More DSPRelated Groups: http://www.dsprelated.com/groups.php3 ">http://docs.yahoo.com/info/terms/ 