Obtaining band-pass/stop coeffiecients from low/high-pass
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From: robert bristow-johnson on 20 Jul 2010 21:47 On Jul 20, 6:42 pm, "AndrewDSPdev" <aritchie(a)n_o_s_p_a_m.vernier.com> wrote: > Thanks for all the help. I have managed to get a 2-pole band-pass filter > working. It seems to work fine, but for some of the inputted pass-bands, > the frequency response has a gain that is not equal to one at the center > frequency. What kind of operation can I perform on my filtering > coefficients to normalize this gain? divide the numerator coefficients by whatever gain you want to set to one. r b-j
From: Rick Lyons on 23 Jul 2010 05:37 On Tue, 20 Jul 2010 13:11:09 -0700 (PDT), Clay <clay(a)claysturner.com> wrote: >On Jul 20, 3:49�pm, "AndrewDSPdev" <aritchie(a)n_o_s_p_a_m.vernier.com> >wrote: >> I'm currently working on a general digital filtering app, and already have >> functionality to run a low/high-pass filter. The type of filter that I'm >> implementing is 2-pole IIR Chebysheb. Essentially, the app uses a borrowed >> algorithm to calculate the filter coefficients (using the z-transform) and >> then I use those coefficients to filter it as is standard. >> � � �I'd like to also add an implementation of a band-pass and band-stop >> filter. However, I'm newish at DSP and don't want to deal with the >> z-transform or the Bilinear Transformation. Is there any simple way to >> tweak my coefficient algorithm, or transform the low-pass/high-pass >> coefficients into ones for a band-pass/stop filter? Any simple online >> sources that could help with this? >> >> Thanks for any help. >> >> Andrew > >The book "Discrete-Time Signal Processing" by Oppenheim and Schafer >has filter transformation formulae for converting lowpass into band >pass, band stop and highpass filters. These are in other DSP books as >well. > >IHTH, >Clay Hi Clay, Yep, "filter transformations" are covered starting on page 430 of O&Ss' 1st Edition (1989). As far as I know that subject was omitted from their 1999 2nd Edition. See Ya', [-Rick-]
From: Clay on 23 Jul 2010 11:41
On Jul 23, 5:37 am, Rick Lyons <R.Lyons@_BOGUS_ieee.org> wrote: > On Tue, 20 Jul 2010 13:11:09 -0700 (PDT), Clay <c...(a)claysturner.com> > wrote: > > > > > > >On Jul 20, 3:49 pm, "AndrewDSPdev" <aritchie(a)n_o_s_p_a_m.vernier.com> > >wrote: > >> I'm currently working on a general digital filtering app, and already have > >> functionality to run a low/high-pass filter. The type of filter that I'm > >> implementing is 2-pole IIR Chebysheb. Essentially, the app uses a borrowed > >> algorithm to calculate the filter coefficients (using the z-transform) and > >> then I use those coefficients to filter it as is standard. > >> I'd like to also add an implementation of a band-pass and band-stop > >> filter. However, I'm newish at DSP and don't want to deal with the > >> z-transform or the Bilinear Transformation. Is there any simple way to > >> tweak my coefficient algorithm, or transform the low-pass/high-pass > >> coefficients into ones for a band-pass/stop filter? Any simple online > >> sources that could help with this? > > >> Thanks for any help. > > >> Andrew > > >The book "Discrete-Time Signal Processing" by Oppenheim and Schafer > >has filter transformation formulae for converting lowpass into band > >pass, band stop and highpass filters. These are in other DSP books as > >well. > > >IHTH, > >Clay > > Hi Clay, > Yep, "filter transformations" are covered starting on > page 430 of O&Ss' 1st Edition (1989). As far as I know > that subject was omitted from their 1999 2nd Edition. > > See Ya', > [-Rick-]- Hide quoted text - > > - Show quoted text - It's kinda sad when a newer version of a book turns out to not be a superset of the earlier one. The book "Digital Signal Processing" by Peled and Liu covers filter transformations in great detail. They show the steps and provide worked out examples so they come very close to being a "cookbook." But you have to do the algebra. Clay |