Hello, 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

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

Hello, I'm a beginner in Matlab. I would like to program a 6th order bessel Low pass filter using matlab. My specifications...

Hello, I'm a beginner in Matlab. I would like to program a 6th order bessel Low pass filter using matlab. My specifications are as follow : Cutott frequency 8.2MHz and my sampling rate is 52.32 MHz (for DVD purposes). My Code is the following clear all

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 ...

Hi everybody, First I like to appologize because my question is some how stupid, since I am mechanical engineer student and new in signal...

Hi everybody, First I like to appologize because my question is some how stupid, since I am mechanical engineer student and new in signal processing, I need to generate a random signal with a desired PSD, I found that this problem was asked before by Tony and Nandan gave him the answer. What Nandan said was to generate a white signal (uniform PSD) and then pass it through a filter whose f

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

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

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...

Dear all,   First What I wanna do is generate a S(x) 16 M-ary symbol use PPM. Then push S(x) through a filter which is...

Dear all,   First What I wanna do is generate a S(x) 16 M-ary symbol use PPM. Then push S(x) through a filter which is the impulse response. and then add gaussin noise after transmitted. at the receiver side, through the filter again and get the received symbol to fin

PLEASE HELP ME IN WRITING THE CODE FOR THE linearized GMSK pulse.. I'M NOT ABLE TO FIND THE DEFINITE INTEGRAL OF A FUNCTION ,, WHOSE UPPER...

PLEASE HELP ME IN WRITING THE CODE FOR THE linearized GMSK pulse.. I'M NOT ABLE TO FIND THE DEFINITE INTEGRAL OF A FUNCTION ,, WHOSE UPPER LIMIT IS A VARIABLE... I' HAVE TO INTEGRATE A GAUSSIAN PULSE OVER THE INTERVAL 0 TO 't' ACTUALLY I'M DESIGNING A FILTER FOR 8PSK MODULATION AND I HAVE TO DESIGN THE FILTER according to ETSI STANDARDS :: GSM 05.04 VERSION 8.1.2 rELEASE 1999 ,, ETSI ...

Hi, Can somebody tell me how to estimate the computational complexity of a code written in Matlab. For example, I wrote a...

Hi, Can somebody tell me how to estimate the computational complexity of a code written in Matlab. For example, I wrote a kalman filter implementation in Matlab and now I want to find out the computational complexity of implementing a Kalman filter. For a kalman f

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

Thanks so much, it's finally working. cheers, Lukas Dear Dwisuu, > > I think I know what's is going wrong. You are using the...

Thanks so much, it's finally working. cheers, Lukas Dear Dwisuu, > > I think I know what's is going wrong. You are using the coefficients in the > s-domain in a command that you should feed with coefficients in the > z-domain. You are trying to make an analog filter using the command > "besself" and matlab is returning coefficients in the s-domain, when you use > the matlab command "filtfi

hello, im new to the topic of adaptive filters and its implementation in matlab. im doing a project on 'filtering an ECG foetus'. we are...

hello, im new to the topic of adaptive filters and its implementation in matlab. im doing a project on 'filtering an ECG foetus'. we are carrying out all our analysis on an ECG wave with only a powerline interference (60hz). the signal available to us is an ECG wave with a noise component of 60hz, sampled at 360hz.. can you help me implement an adaptive filter using LMS algorithm for this s...

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

Hello, My name is Federico Baguear, Im from Buenos Aires and I study Biomedical Engineer. Im working in a real time system identification,...

Hello, My name is Federico Baguear, Im from Buenos Aires and I study Biomedical Engineer. Im working in a real time system identification, using a IIR LMS adaptive filter. I couldn`t find much references about how to estimate the μ coefficients for a particular signal. I try with a paper written by Wu, "Fast converging adaptive IIR LMS filter in direct form using the suboptimal ste

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...

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...

Dear all, I am creating a speech recognition system using simulink. I wish to compare my output signal from my filter in...

Dear all, I am creating a speech recognition system using simulink. I wish to compare my output signal from my filter in neural network. What type of input can a neural network recognise? Is it just a string of vector? Can it be a waveform/signal? If really ne

Salam Im Mansoor Mustafa , i'm graduate student. i want to plot the Raised Cosine Filter with following parameters: Bit Stream :...

Salam Im Mansoor Mustafa , i'm graduate student. i want to plot the Raised Cosine Filter with following parameters: Bit Stream : 1, 0, 1 Raised Cosine Roll of Factor : 1/2 Symbol Rate : 50kbps Truncation Time : + - 6 symbols Can any one provide me the code ? or help me to plot this ? Thanx Regards! Engr. Mansoor

Dear good people, In a nutshell: modulated filter banks[lapped transforms] are discrete time-frequency analysis tools where the sinusoidal...

Dear good people, In a nutshell: modulated filter banks[lapped transforms] are discrete time-frequency analysis tools where the sinusoidal waveforms are localized with a prototype filter[window function]. Based on the sheer count of published papers, one would also expect to find some pieces of useful code from the web. If such collections exist, they seem to be hidden from the search engines. ...