## How to extract IIR coefficients w/Matlab?

Started by in comp.dsp15 years ago 8 replies

Hello. 1. Is there an easy way in Matlab to take a z-domain transfer function and extract coefficients for an IIR filter type...

Hello. 1. Is there an easy way in Matlab to take a z-domain transfer function and extract coefficients for an IIR filter type implementation? I first designed my filter in the s-domain and tweaked it to get my desired response. I then used Matlab's c2dm with the Tustin method to get a z-domain transfer function. I am aware of "fdatool" but my filter started out life in s-domain and it...

## Measuring Lag of a Black-Box Filter

Started by in comp.dsp15 years ago 8 replies

I'm curious how one goes about measuring the lag in the output of an arbitrary lowpass filter. That is, I want to measure the lag in terms of...

I'm curious how one goes about measuring the lag in the output of an arbitrary lowpass filter. That is, I want to measure the lag in terms of sample count or a function of cutoff wavelength. For example, in the case of a simple 1-pole IIR filter having a 3 dB cutoff wavelength of pi*n : a0 = 2/(n+1) b1 = 1 - a0 y[0] = a0*x[0] + b1*y[1] This will have a lag of (n-1)/2, a linear r...

## Wanted: critically-damped high-pass IIR filter

Started by 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 ...

## Rounding off problems in IIR filters

Started by in comp.dsp15 years ago 11 replies

Hi, I have some problem in implementing a second order high pass IIR filter.The problem is that when I tried finding the step response of the...

Hi, I have some problem in implementing a second order high pass IIR filter.The problem is that when I tried finding the step response of the filter, the response I got was varying from what was actually expected of it. When I traced back, I found the following problem. Earlier I had thought that the problem was only with respect to the scaling of the IIR filter coefficients. But I had n...

## what is audio shapping?

Started by in comp.dsp15 years ago 1 reply

Hi, What is audio shapping? and why we want to shape the audio signal before sending it to speaker? what are the main components of...

Hi, What is audio shapping? and why we want to shape the audio signal before sending it to speaker? what are the main components of audio shapping? What kind of IIR filter should be used? Thanks in advance for your informations. Gold

## Imp.inv/bilinear trans. vs. expm()

Started by in comp.dsp15 years ago 7 replies

Hi, What is the reason to that dicrete IIR filters are usually designed using impulse invariant transformation (= Euler integration) or...

Hi, What is the reason to that dicrete IIR filters are usually designed using impulse invariant transformation (= Euler integration) or bilinear transformation (= trapetzoidal integration) instead of the closed form solution that can be computed by the matrix exponential function? At least in DSP books IIR filter design seems to be taught like this: 1. Design a continuous-time (analo...

## How can I get Frequency Response from Impulse Freponse In Matlab?

Started by in comp.dsp15 years ago 3 replies

Hi! I want to get FRF of IIR Filter from Impulse Response. So, I got FRF using two method. One is FREQZ, another is FFT of impulse response....

Hi! I want to get FRF of IIR Filter from Impulse Response. So, I got FRF using two method. One is FREQZ, another is FFT of impulse response. But two results are different. Why? My source code are below : clear all close all clc fs=2048; % Sampling Frequency N=65536; % Buffer Size t=[0:N-1]'./fs; % Time f = [0:N/2]'*fs/N; % Frequency x = zeros(N,1); % Impulse Input...

## IIR filter frequency response

Started by in comp.dsp15 years ago 20 replies

Sorry if this seems a bit too lame, but I was wondering if someone might possibly tell me how to calculate the frequency response of an IIR...

Sorry if this seems a bit too lame, but I was wondering if someone might possibly tell me how to calculate the frequency response of an IIR filter having only it's coefficients. Thanks

## IIR filters revisited

Started by in comp.dsp15 years ago 14 replies

Hello mates, Another question about IIRs from me. I managed to fix my buggy code and I have further problems now. Is it possible (and it...

Hello mates, Another question about IIRs from me. I managed to fix my buggy code and I have further problems now. Is it possible (and it what way) to alter the IIR filter coefficients (the ones from the diff equation) to make it stable if the filter is unstable? And I need to know the stable filter's coefficients! What is the optimal way to check if an IIR filter is stable (having o...

## Deriving IIR coefficients for exponential

Started by in comp.dsp15 years ago 8 replies

All right, I've learned a lot from this group in the past couple of months. Starting from almost zero knowledge of DSP, I learned how to...

All right, I've learned a lot from this group in the past couple of months. Starting from almost zero knowledge of DSP, I learned how to derive digitial IIR filter coefficients from any arbitrary all-pole polynomial filter. I have a layman's understanding of s-plane and z-plane. I published the solution for generalized all-pole filter coefficients at http://unicorn.us.com/alex/allpolefi...

## How to achieve a lower pass band ripple?

Started by in comp.dsp15 years ago 4 replies

Hi there! I've got a problem with my 8th order IIR filter. In fact, it's a elliptic high pass designed with matlab. Cutoff frequency is...

Hi there! I've got a problem with my 8th order IIR filter. In fact, it's a elliptic high pass designed with matlab. Cutoff frequency is 350Hz, attenuation 40dB and sample rate 8000Hz. Pass band ripple was designed to 0.1dB, but after transforming it into biquad structure and programming it, the ripple is about 1.5dB. That is too much for me. Only 0.5dB are acceptable. Has anyone some suges...

## DSP, IIR filter C/C++ Code, Need Help

Started by in comp.dsp15 years ago 5 replies

Hi, I am running into a problem, and that is developing a C code for TMS320C6713 DSP processor. The application requires coding of a complex...

Hi, I am running into a problem, and that is developing a C code for TMS320C6713 DSP processor. The application requires coding of a complex valued IIR filter. The filter equation is: H(z) = 1/(1+ SUM(i=1:8) (Ai Z^-i)) Hoopefully you can understand the above equation, the SUM is actually summation sign (SIGMA) and (i= 1:8) is lower and upper limit of the sigma. I have precalculated ...

## IIR filter and SoS

Started by in comp.dsp15 years ago 4 replies

Hi all ! I'm playing a little with matlab's fdatool and I noticed this : when I design a filter, as a 30th order one, with second order...

Hi all ! I'm playing a little with matlab's fdatool and I noticed this : when I design a filter, as a 30th order one, with second order sections, the sections doesn't have the same coefficients. It is not as if I just designed a 2nd order filter and applied it 15 times. What is the difference ? Couldn't I just apply the second method described here ?? Thanks in advance ! Sam

## [newbie] how to build DSP prototype board

Started by in comp.dsp15 years ago 6 replies

Hi all, Suppose I've got well working algorithm (IIR filter) developed in Code Composer Studio along with DSP Starter Kit, say C5510. I've got...

Hi all, Suppose I've got well working algorithm (IIR filter) developed in Code Composer Studio along with DSP Starter Kit, say C5510. I've got a built COFF file (*.out) that loads into DSP through DSK and works there like a charm. Now what. I want to have that code to work on a standalone prototype board that I'll build. Please, tell me what are the steps to be taken in order to transfer m...

## Question about IIR filter design using Impulse Invariance method

Started by in comp.dsp15 years ago 8 replies

I am designing low pass IIR filter using Impulse Invariance method. I've got transfer function H(z) which depends on impulse sample period...

I am designing low pass IIR filter using Impulse Invariance method. I've got transfer function H(z) which depends on impulse sample period T. I was asked to choose an appropriate impulse sample period T for H(z) such that the input signal x(t) = 5 cos(2pi(8000)t ? pi/3) ? 4 cos(2pi(40000)t + pi/2) sampled at a rate of Fs = 100 kHz has only the 8000 Hz sinusoid in the passband. I do...

## what does phase distortion look like in image processing?

Started by in comp.dsp15 years ago 7 replies

I read the following in Matlab Help Document of Image Processing ToolBoxes, "FIR filters can be designed to have linear phase, which helps...

I read the following in Matlab Help Document of Image Processing ToolBoxes, "FIR filters can be designed to have linear phase, which helps prevent distortion. " I am wondering what does phase distortion look like in image processing? Also, "Another class of filter, the infinite impulse response (IIR) filter, is not as suitable for image processing applications. It lacks the inheren...

## Noise-feedback...

Started by in comp.dsp15 years ago 1 reply

Hi group! I'm an engineering student and I'm trying to implement an IIR filter. It's a 6th order Chebyshev LP filter. We perform L2 norm...

Hi group! I'm an engineering student and I'm trying to implement an IIR filter. It's a 6th order Chebyshev LP filter. We perform L2 norm scaling of the coefficients. To simulate the implementation environment we programmed the whole thing in Matlab, rounding the products in the appropriate places. Everything works fine, but as the final task in this assignment we are supposed to ...

## IIR filter paradox?

Started by in comp.dsp15 years ago 7 replies

Hello all, This is just a though experiment, and Im sure Im wrong, I just dont know why Im wrong :-). Consider a simple recursive filter,...

Hello all, This is just a though experiment, and Im sure Im wrong, I just dont know why Im wrong :-). Consider a simple recursive filter, defined as H(z) = 1 / (z - 0.5) On the z-plane, this translates to a pole at 0.5 DC. Meaning that the output of the filter is infinite when the input is a theoretically perfect DC value of 0.5. This makes a filter like that unstable, even when ...

## HELP : IIR in an FPGA

Started by in comp.dsp15 years ago 3 replies

Hello and thank you for reading my post ! I would like to design an IIR filter in a FPGA. It will be a 16-order filter, made with 8...

Hello and thank you for reading my post ! I would like to design an IIR filter in a FPGA. It will be a 16-order filter, made with 8 second-order sections. The signal to filter is 24 bit wide and I need those 24 bits of precision. My question is : how many bits do I need at the outputs of my multiplier tu be sure that I still have 24 bits of precision at the output ? Well, if we sup...

## Noise-equivalent bandwidth of digital filter

Started by in comp.dsp15 years ago 4 replies

Hello all, Havent been around in some time, but I have a question that has been needling me. I want to determine the noise-equivalent...

Hello all, Havent been around in some time, but I have a question that has been needling me. I want to determine the noise-equivalent bandwidth of an FIR or IIR filter. I started by attempting to convert the analog definition to a digital equivalent. Although my attempts dont appear to give me the correct answer. Hoping someone could lend a hand in this seemingly simple problem (not en...