IIR Filter Design

Started by jerseygirl in comp.dsp13 years ago 3 replies

I was given a continous signal x(t) = 1.272*cos(2*pi*50*t)−0.424*cos(2*p*150*t)+0.255*cos(2*pi*250*t) and I generated discrete signal...

I was given a continous signal x(t) = 1.272*cos(2*pi*50*t)−0.424*cos(2*p*150*t)+0.255*cos(2*pi*250*t) and I generated discrete signal {x(n)}, 0


IIR filter reference designs

Started by Rune Allnor in comp.dsp15 years ago 23 replies

Hi all. I'm struggling to implement my own codes for IIR filter design by analog prototypes. The Butterworth filter works, the Chebyshev type...

Hi all. I'm struggling to implement my own codes for IIR filter design by analog prototypes. The Butterworth filter works, the Chebyshev type 1 doesn't. The books I have available are Proakis & Manolakis' 3rd edition, Oppenheim & Schafer 1975+1999, Leland B Jackson 1989. Most show, relatively crudely, the main steps in filter design. Only P&M show any examples of Cheb 1 filters in suff...


Wanted: critically-damped high-pass IIR filter

Started by axlq in comp.dsp15 years ago 92 replies

A critically-damped 2-pole Butterworth lowpass filter is described in http://people.umass.edu/exsci735/Robertson&Dowling.pdf A method to...

A critically-damped 2-pole Butterworth lowpass filter is described in http://people.umass.edu/exsci735/Robertson&Dowling.pdf A method to convert this filter to a highpass filter is described in http://www.health.uottawa.ca/biomech/lab/docs/ncb2_sm.pdf Those two papers are short and simple enough for a non-DSP guy like me to understand. One thing the second paper doesn't say is that one ...


Importance of phase in Audio

Started by Anonymous in comp.dsp13 years ago 12 replies

Hello all, i am designing a low pass filter and high pass filter to pass my audio samples. in matlab i checked out both butterworth...

Hello all, i am designing a low pass filter and high pass filter to pass my audio samples. in matlab i checked out both butterworth and chebyshev(cheby2) 3rd order filter. I have observed there is a lot of differance in the phase plot of both the filters. Now my query went to what actuall effect of phase in audio? please anybody help me in solving this. Looking for the repl...


Re: Questions regarding Octave

Started by bharat pathak in comp.dsp12 years ago 44 replies

Eric, Sending you the working code back again, %------------------------------ modified code start % 5/1/96 Eric Jacobsen % % Design a...

Eric, Sending you the working code back again, %------------------------------ modified code start % 5/1/96 Eric Jacobsen % % Design a butterworth filter and determine the equivalent FIR taps. % order=4; % Order of filter. cutoff=0.25; % Cutoff relative to 1.0 = 1/2 sample rate. [B,A]=butter(order,cutoff); %---------------- removed "end;" line (not required) -- Regards B...


IIR BandPass nominator

Started by Blaazen in comp.dsp9 years ago 7 replies

Hello everybody, I have problem with nominator of H(z) of bandpass filter. I develop musical open-source software (non-commercional) and I...

Hello everybody, I have problem with nominator of H(z) of bandpass filter. I develop musical open-source software (non-commercional) and I wrote my filter-design algorithms based on polynomials (without complex aritmetic). I have all working well (Bessel, Butterworth, Chebyshev, inverse Chebyshev and Cauer's elliptic filters Low, High Band pass and band Reject). Now I decided do rewrite my al...


IIR notch filter

Started by itsh11 in comp.dsp12 years ago 18 replies

I am trying to design a second order digital IIR band stop (notch) filter with the following specs: 3dB cut off frequencies: 55Hz and 65Hz I...

I am trying to design a second order digital IIR band stop (notch) filter with the following specs: 3dB cut off frequencies: 55Hz and 65Hz I want the notch at 60Hz with atleast 90dB attenuation at the 60Hz. Sampling frequency: 200hz I tried various filter configurations like a Butterworth or Chebyshev but could not get the attenuation higher than 70 dB.I want a second order filter to accom...


High-pass filter without phase shift

Started by futureignobel in comp.dsp8 years ago 3 replies

Hi, I'm trying to design a high-pass filter to filter out DC component (bias) from a noisy accelerometer measurement. I've tried to go with a...

Hi, I'm trying to design a high-pass filter to filter out DC component (bias) from a noisy accelerometer measurement. I've tried to go with a second order Butterworth filter with coefficients from http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html. The filtering works just fine, but the filter has a non-linear phase shift, so I don't know if that's the right choice. I'd also like to go w...


IIR Filter Co-efficients Formula

Started by right05 in comp.dsp12 years ago 17 replies

Hello All, I am in the process of creating a front end for a butterworth low pass filter of order 4.This will allow the user to enter the cut...

Hello All, I am in the process of creating a front end for a butterworth low pass filter of order 4.This will allow the user to enter the cut off frequency and the software will calculate the co-efficients for the user. I used the information(formula for co-efficients) from this website http://www.apicsllc.com/apics/Sr_3/Sr_3.htm The formula is correct for the band stop filter but it do...


LPF butterworth filter question

Started by Kai7918 in comp.dsp10 years ago 10 replies

Hey guys, I have created a butter worth filter using SPTOOL in matlab. the specifications are ? Sampling frequency: 1000Hz ? Passband...

Hey guys, I have created a butter worth filter using SPTOOL in matlab. the specifications are ? Sampling frequency: 1000Hz ? Passband ripple: 3dB ? Stop band edge frequency: 250Hz ? Stopband attenuation: 40dB Pas band edge frequency: 225Hz Minimum order design Single section The filter is stable. Here is a screen shot of the Magnitude response: h


what is the roll-off factor and cutoff-slope of Butterworth filter?

Started by lucy in comp.dsp15 years ago 2 replies

Hi all, I am doing this in digital domain... I have used the Butterwork lowpass filter grabbed from Internet: % X and Y matrices with...

Hi all, I am doing this in digital domain... I have used the Butterwork lowpass filter grabbed from Internet: % X and Y matrices with ranges normalised to +/- 0.5 x = (ones(rows,1) * [1:cols] - (fix(cols/2)+1))/cols; y = ([1:rows]' * ones(1,cols) - (fix(rows/2)+1))/rows; radius = sqrt(x.^2 + y.^2); f = 1 ./ (1.0 + (radius ./ cutoff).^(2*n)); Then "f" is ...


Recipes for non-ripple 2-pole filters available

Started by axlq in comp.dsp15 years ago 5 replies

For the newbies here (like me): I have compiled recipes (step by step instructions) for coding three kinds of 2-pole digital filters, both...

For the newbies here (like me): I have compiled recipes (step by step instructions) for coding three kinds of 2-pole digital filters, both low-pass and high-pass, complete with correction factors to ensure that the 3 dB cutoff frequency stays where you put it when you cascade filters together. http://unicorn.us.com/alex/2polefilters.html The recipes cover Butterworth, Critical...


Problem measuring frequency response of filters

Started by Malcolm Haylock in comp.dsp16 years ago 15 replies

Hi everyone, I'm fairly new to DSP and am having trouble measuring the frequency response of various (mainly Butterworth) filters that I wish...

Hi everyone, I'm fairly new to DSP and am having trouble measuring the frequency response of various (mainly Butterworth) filters that I wish to use in a program. To test the response I'm running the filters on white noise using a 1024 point buffer (which is what the program will use) then taking the DFT of the output. I'm doing this 1000 times and averaging the result. The result...


Problem With Implementation of costas loop In MATLAB

Started by Neophyte in comp.dsp14 years ago 2 replies

I am trying to implement a costas loop in matlab.The code which i have written is based on blockwise most of it using the available functions...

I am trying to implement a costas loop in matlab.The code which i have written is based on blockwise most of it using the available functions of the matlab...the signal frequency 600bits/sec and the carrier frequency is 3khz.In trackin the error the the lowpass filter used is butterworth filter...after removing the carrier frequency component....the left value should be just a dc componet ...


How to decide good 2D filter size for image filtering?

Started by lucy in comp.dsp15 years ago 1 reply

Hi all, I am trying to design a 2D low pass filter that filters some image. I have decide the cutoff frequency to be 0.2, in terms of...

Hi all, I am trying to design a 2D low pass filter that filters some image. I have decide the cutoff frequency to be 0.2, in terms of normalized frequency in [0, 0.5]. I decided to use Butterworth low pass filter and 7th order. I generate the frequency response using the "lowpassfilter" function as attached below, then by using "fwind1" I got an 2D impulse response of the filter wit...


would you like to try my IIR low pass digital filter design program?

Started by Nasser M. Abbasi in comp.dsp9 years ago

Hello DSP experts; I took an introduction to digital filters course at cal state univ and learned something about design of IIR digital...

Hello DSP experts; I took an introduction to digital filters course at cal state univ and learned something about design of IIR digital filters (was a hard course!). I used the textbook we had and the instructor notes to write this small Mathematica demonstration to design a low pass IIR digital filter using Butterworth method. I used Mathematica since I wanted to display H(s) and H(...


Linkwitz-Riley transient response

Started by jeff227 in comp.dsp13 years ago 13 replies

I coded a very nice crossover using 4th order Linkwitz-Riley filters whereby their sine wave outputs do indeed sum to a flat...

I coded a very nice crossover using 4th order Linkwitz-Riley filters whereby their sine wave outputs do indeed sum to a flat frequency response. However, the square wave response of the summation is terrible. I am getting over 4dB of overshoot and ringing. I understand Butterworth IIR filters have non-linear phase shift - with respect to their inputs. But in the 4th order Linkwitz-Riley al...


Why there are amplitude distortion for a single frequency signal passing through filter?

Started by Anonymous in comp.dsp7 years ago 11 replies

I designed a bandpass filter under sptool of Matlab with the following specs: 1) Fs=25MHz, 2) fstop1=20MHz, 3) fpasss1=25MHz, 4)...

I designed a bandpass filter under sptool of Matlab with the following specs: 1) Fs=25MHz, 2) fstop1=20MHz, 3) fpasss1=25MHz, 4) fpass2=30MHz, 5) fstop2=40MHz. 6) IIR Butterworth filter. It gives me an IIR filter of 32-order. After that, I generated a sequence of single frequency signal of 3MHz and let this signal being filtered by the designed IIR filter. When I observ


What is the standard output of a low pass digital filter design program?

Started by Nasser M. Abbasi in comp.dsp10 years ago 2 replies

Hello, I wrote a small program for class to design a low pass digital filter. it is an IIR filter designed using Butterworth. We are given...

Hello, I wrote a small program for class to design a low pass digital filter. it is an IIR filter designed using Butterworth. We are given the specifications, and I have generated H(s) and H(z) (using impulse invariance, and using bilinear methods). What is considered as a "standard" output to print as a result of such a design run? I am now printing the pole locations for Butterwor...


How I can design real time dsp system

Started by pwaiaung in comp.dsp13 years ago 5 replies

Hello everyone, I want to design a real time DSP system.Band of interest extends from o to 4kHz.I will use 12 bit ADC and 3rd order butterworth...

Hello everyone, I want to design a real time DSP system.Band of interest extends from o to 4kHz.I will use 12 bit ADC and 3rd order butterworth lowpass filter. What I want to know is how I can estimate the minimum stopband attenuation for the anti-aliasing filter, minimum sampling frequency and The level of the aliasing error relative to signal level in the passband for the estimated Amin an...