## Forums Search for: Filter

## separating passband zeros and stopband zeros

inHi, I am working on the filter design, i am designing the FIR filter, i want to know exactly which zeros corresponds to...

Hi, I am working on the filter design, i am designing the FIR filter, i want to know exactly which zeros corresponds to passband and which zeros to stopband. how to get them i can get the zeros all by using the command tf2zp

## need help bandpass filter in simulink

Hi,I have a couple of questions see if anyone can help: There are about the simpowersystems of simulink. 1. I couldn't find the...

Hi,I have a couple of questions see if anyone can help: There are about the simpowersystems of simulink. 1. I couldn't find the block reference for "discrete 2nd-order variable-tuned fliter(BANDPASS FILTER)". in this block, there are two input-ports, F and in? what

## Noise Cancellation using MATLAB

inHello, I'm Natasha and I'm doing my project on ' active noise cancellation in headsets'. I am new to MATLAB and have written a code for noise...

Hello, I'm Natasha and I'm doing my project on ' active noise cancellation in headsets'. I am new to MATLAB and have written a code for noise cancellation of an audio signal using a simple lms filter. However, the program shows errors,particularly in the lms filter designing area; [d,r] = wavread('df3_n0H.wav.wav'); r =8500 size(d) ans =42500 M = 32; mu=0.004;...

## LP filter..

hi colleagues... i have generated a random signal in matlab with normal distribution (my code is written below). my signal is high...

hi colleagues... i have generated a random signal in matlab with normal distribution (my code is written below). my signal is high frequency and i want to filter it to have lower frequencies at about 1/300

## Digital Filter Package and ML Version 5.2

Does anybody know if there is an updated version of the Digital Filter Package that will run smoothly in MatLab 5.2.x.x.? The app...

Does anybody know if there is an updated version of the Digital Filter Package that will run smoothly in MatLab 5.2.x.x.? The app was originally written for Version 4.0, and must be modified to run in 5.2. I'm pretty sure the oroginal author has not updated it, and never plans to..bu

## Non-linear programming; GNU licence/freeware alternatives for MATLAB fmincon?

Greetings DSP enthusiastics and other earthlings, My problem is related with the design of perfect reconstruction modulated filter banks. It...

Greetings DSP enthusiastics and other earthlings, My problem is related with the design of perfect reconstruction modulated filter banks. It is one component of the self-made lapped transform/filter bank toolbox for MATLAB. I've been utilizing optimization toolbox fmincon to solve minimization problems such as min h^T Q h h subject to c (h) = 0, n where h is a vector ...

## Re: determining filter coefficients

Dear Sana: Taka look at "butter" and "buttord" functions. They will calculate the coefficientes for a "butterworth" filter....

Dear Sana: Taka look at "butter" and "buttord" functions. They will calculate the coefficientes for a "butterworth" filter. There are similar functions for "chebyshev" and "elliptic" filters. OK? Regards, P.David

## Inverse Filter

Hi there, I am measuring the output of a mechanical manipulator in response to band limited gaussian white noise. This then allows me to calculate...

Hi there, I am measuring the output of a mechanical manipulator in response to band limited gaussian white noise. This then allows me to calculate the transfer function using - cross-power spectral density of the stimulus and the response divided by the auto-PSD of the stimulus. I have been able to achieve this in matlab. Now, I want to nullify the effect of the manipulator using a inverse filter ...

## Very long filters and responses

Hello, I am kind of new to DSP, however part of my bachelor's thesis is about crosstalk cancellation as it pertains to binaural audio. The...

Hello, I am kind of new to DSP, however part of my bachelor's thesis is about crosstalk cancellation as it pertains to binaural audio. The problem I'm having is with designing filters. Part of my assignment is to approximate the inverse of a filter. This filter is 40000 samples long, since it is a combination of several room impulse responses sampled at 44khz. My question is this: How do y...

## remezord & weight vector

inHello, When using remezord to estimate the length of a filter given the desired specifications the remezord function also...

Hello, When using remezord to estimate the length of a filter given the desired specifications the remezord function also returns a weight vector w to be passed to the remez function to design the actual filter. My understanding was that these weights tell the rem

## How does Matlab implement the [y z] = filter(b,a,X,zi) function?

Hi... I am trying to convert a Matlab script to C++ involving the filter.m function. It is said in the Matlab doc that this function is...

Hi... I am trying to convert a Matlab script to C++ involving the filter.m function. It is said in the Matlab doc that this function is implemented using the Direct Form II transposed structure, but when I implement this directly as it is, it gives results which are not as close as usual with the Matlab output, that is several magnitudes of order difference compared to machine precision...

## Re: Colored Noise in Matlab

Can anybody elaborate on this.if h(t) is the impulse response of the filter I have to send white Gaussian noise to it,in continuous domain .In...

Can anybody elaborate on this.if h(t) is the impulse response of the filter I have to send white Gaussian noise to it,in continuous domain .In matlab simulation I have to generate a vector of Gaussian random variables using randn and convolve it with the discrete filter coefficents and use each element of the output vector as one time instant value.Is it correct Thank You Rakesh > > >

## Problem with rcosflt(Raised cosine filter) in Matlab 6.5 version

Hi, In a LAB experiment of digital communication system where rcosflt is used as channel, the error is produced due to...

Hi, In a LAB experiment of digital communication system where rcosflt is used as channel, the error is produced due to extended length of output of the filter. When I ran the program in Matlab 5.1, its giving perfect result but when running in 6.5, its giving error

## Finding impulse response from IIR Filter

inHi all, I made an IIR filter with its coefficients (a0, a1, a2, b0, b1, b2). I created it in z-domain at MATLAB as the usual standard. Now I...

Hi all, I made an IIR filter with its coefficients (a0, a1, a2, b0, b1, b2). I created it in z-domain at MATLAB as the usual standard. Now I want to find its impulse response (h[n]) which is in time domain. Please help me how to do it in MATLAB... I've search syntax to convert it, but I couldn't find one.. Thank you very much for your help...

## block processing, filter behaviour

Hello I have a question about filter behaviour and block processing. I have a signal x[k]=s[k]+n[k] where s is clean speech and n is ...

Hello I have a question about filter behaviour and block processing. I have a signal x[k]=s[k]+n[k] where s is clean speech and n is colored, gaussian noise. In the algorithm I am working on I do block processing. Each block xb[j]=(x[1+L*j],x[2+L*j],.....,x[L+L*j]) where xb[j] is the j'th block-input is sent thru 2 filters

## Re: Discrete Fourier Transform of a signal containing only dirac impulses

Mirco- > It's perhaps a silly question, but I don't see what I'm doing wrong. I'm > trying to calculate the FT of a signal containing only...

Mirco- > It's perhaps a silly question, but I don't see what I'm doing wrong. I'm > trying to calculate the FT of a signal containing only dirac impulses. To > avoid aliasing I first low pass the signal with a Butterworth filter Why will there be aliasing in a signal with pulses? If you look at a magnitude plot, you should get something like a dense comb filter. One pulse would give you

## how to chnage a bandpass filter to a low pass filter

hello and good day ! A pass band receiver system will process modulated signal , i.e. there iz RF stage , then IF stage and Baseband...

hello and good day ! A pass band receiver system will process modulated signal , i.e. there iz RF stage , then IF stage and Baseband (BB) stage. I want to model LPF instead of BPF in RF and IF stage i.e. model a LPF from a BPF, which gives the same output as passband system gives. I tackle this problem that i take hilbert transform to retain only positive frequencies and...

## Inverse filter

Hi All I'm doing on Robust Speaker identification/verification for telephony applications for my final year project.I pass through my clean...

Hi All I'm doing on Robust Speaker identification/verification for telephony applications for my final year project.I pass through my clean speech signal to low-pass filter with cut-off frequency 3.4kHz. I want to use inverse filtering about 4~5kHz. I think the step will be 1. transform the telephony speech into frequency domain 2. i need to get back the original S(z)=c(z)/H(z) where S(z) i...

## MPEG-4 LD problem

Hello all, I am trying to implement some unsymmetrical windows in Mpeg-4 LD. I acheived a delay reduction in filter bank but when tested with...

Hello all, I am trying to implement some unsymmetrical windows in Mpeg-4 LD. I acheived a delay reduction in filter bank but when tested with some standard test signals.I found little bit distorted in test signals.I would like to check PR in MDCT filter bank(alone).Any suggestions/revelant papers or books to find the problem.Please let me know. Thanking you, Deep

## decimation filter for high order decimation rate

Hi ALL I working on application that i need to sample 1.9Khz signal by 1Mhz , its mean 512 OSR - AFTER SIGMA DELTA MODULATION ( FOR SEPARATE...

Hi ALL I working on application that i need to sample 1.9Khz signal by 1Mhz , its mean 512 OSR - AFTER SIGMA DELTA MODULATION ( FOR SEPARATE the quantization noise from the signal) i need to decimate the signal by decimation rate of 512. i decided to use CIC filter to decimate by 128 and then cascade more 2 FIR - one as compensator and the other for the phase linearization. the DC gain in t...