## DSP sample rate question

Started by in comp.dsp18 years ago 6 replies

Hello, I have a quick question. Say I am working on a DSP low pass filter that will be utilizing a 10Hz cutoff frequency. This is a two pole...

Hello, I have a quick question. Say I am working on a DSP low pass filter that will be utilizing a 10Hz cutoff frequency. This is a two pole butterworth based recursive (IIR) filter. I -think- our highest frequency component will most likely be (rate of change in digital input to the IIR) around 8kHz. My manager who is primarily experienced in analog filter thinks that we'll only need to w...

## Simulating higher order IIR filters using cascaded low order sections

Started by in comp.dsp10 years ago 4 replies

I'm trying to make a tool that implements various types of IIR filters up to a certain maximum filter order (8 in my case). I've been learning...

I'm trying to make a tool that implements various types of IIR filters up to a certain maximum filter order (8 in my case). I've been learning about the bilinear transform and how to use it to turn an analog transfer function into a something that can be implemented as a digital filter. I already have working implementations for 1-5th order butterworth low pass filters, but I've not

## Recipes for non-ripple 2-pole filters available

Started by in comp.dsp18 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 With Implementation of costas loop In MATLAB

Started by in comp.dsp17 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 ...

## IIR Filter Co-efficients Formula

Started by in comp.dsp15 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...

## Fixed point implementation of 4'th order IIR filters

Started by in comp.dsp19 years ago 7 replies

Hi Does anyone have some guidelines on how to implement a 4'th order low-pass Butterworth IIR filter in fixed point. My cut-off frequency is...

Hi Does anyone have some guidelines on how to implement a 4'th order low-pass Butterworth IIR filter in fixed point. My cut-off frequency is relatively close to the DC frequency so high precision is needed for the coefficients. What about realization structure and so on! I have implemented the bit-flipping algorithm in http://www.cmsa.wmin.ac.uk/~artur/pdf/Paper16.pdf for quantization of...

## Low Pass Reconstruction Filter

Started by in comp.dsp16 years ago 7 replies

I'm trying to build a Low Pass Reconstruction Filter for use after a DAC . The thing is it has to be a Zi = 300 Ohm input and the fc has to be...

I'm trying to build a Low Pass Reconstruction Filter for use after a DAC . The thing is it has to be a Zi = 300 Ohm input and the fc has to be 37 MHz and 50 dB attuenuation @ 118 Mhz and the passband ripple has to be < 1dB. Already tried the Butterworth but the coils were to large in fuction of the self resonant frequency of it . The Chebyshev filter has a too large ripple in th

## IIR filter order

Started by in comp.dsp12 years ago 7 replies

I'm finally about ready to release an IIR filter design program called - you guessed it - ScopeIIR (TM). It currently supports the basic...

I'm finally about ready to release an IIR filter design program called - you guessed it - ScopeIIR (TM). It currently supports the basic design types of Butterworth, Chebyshev, and Elliptic. I'm trying to finalize the default and maximum IIR filter order that the program will support. To help me do that, please tell me about your experiences with IIR filter order, assuming the above t...

## IIR notch filter

Started by in comp.dsp14 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...

## LPF butterworth filter question

Started by in comp.dsp12 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 in comp.dsp18 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 ...

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

Started by in comp.dsp18 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 ...

## Controllable cut-off freq for DF2 SOS

Started by in comp.dsp11 years ago 11 replies

Hi everybody I'd like to have a 2nd order low-pass filter with real-time controllable cut-off frequency (fc). I am working on a 16-bit...

Hi everybody I'd like to have a 2nd order low-pass filter with real-time controllable cut-off frequency (fc). I am working on a 16-bit fixed-point processor and therefore I've chosen a DF2 SOS structure to implement a Butterworth LPF. In order to simplify things, I've made up a table containing g, a1 and a2 of the filter for different cut-off frequencies = fn*(0.006250, 0.006250, 0.012500, 0.02...

## I have a sampled signal that was HP-filtered before sampling. Is there a way to revert the HP-filtering?

Started by in comp.dsp6 years ago 12 replies

I have a sampled signal that was HP-filtered before sampling. Is there a way to revert the signal back to its pre-HP-filtered state? I know...

I have a sampled signal that was HP-filtered before sampling. Is there a way to revert the signal back to its pre-HP-filtered state? I know it is impossible to restore the DC-component of the signal. But we can assume it is 0V. I MATLAB script to show how the signal x looks HP-filtered: % % Butterworth Highpass filter designed using FDESIGN.HIGHPASS. % All frequency values ar...

## seeking scipy examples of swept-coefficient (time varying) filters

Started by in comp.dsp2 years ago 5 replies

I've been pleased with ease of use of the scipy.signal package for implementing bandpass filters: the `signal.butterworth()`,...

I've been pleased with ease of use of the scipy.signal package for implementing bandpass filters: the `signal.butterworth()`, `signal.sosfilt_zi()` and `signal.sosfilt()` functions do just what I expect for a filter with constant center frequency and bandwidth. But now I'd like to make the frequency and bandwidth settings be time varying functions themselves, and I'm not sure how to

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

Started by in comp.dsp18 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...

Started by in comp.dsp16 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...

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

Started by in comp.dsp12 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 in comp.dsp16 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...

## Higher order Bessel filter

Started by in comp.dsp15 years ago 10 replies

Can anybody give me an idea on how to implement nth order bessel filter with cascaded biquad sections? I am able to implement higher order...

Can anybody give me an idea on how to implement nth order bessel filter with cascaded biquad sections? I am able to implement higher order Butterworth with cascaded biquads by following equations given in the link http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt I found that to design filter coefficients for the biquads, Q has to be changed in the equation Alpha = alpha = sin(w0...