Hi all! I'm trying to realize the work of Sadjadi named "A full plane block kalman filter for image restoration". In this paper, there is...

Hi all! I'm trying to realize the work of Sadjadi named "A full plane block kalman filter for image restoration". In this paper, there is a section to find a parameter named "p". Simply, I can explain my problem like this. There are some vector observations, x, that represents some pixel values on the image and their sizes are 1xN. It is wanted to find the parameter p(k) and its definit...

Hi guys.. Can anyone help me how to spin and de spin the data in matlab and then pass it into a matched filter and down sample...

Hi guys.. Can anyone help me how to spin and de spin the data in matlab and then pass it into a matched filter and down sample it... Actually I got to the point until de-spinning it but don't know how to do the down sampling part... Hoping 2 c any response... Div

Hi, I have a few problems, looking for there solutions: How to set filter type, length, order?and sampling rate using DWT command? I mean how...

Hi, I have a few problems, looking for there solutions: How to set filter type, length, order?and sampling rate using DWT command? I mean how all parametrs that we set in fdatool, can be adjusted in dwt comman. secondly, how wavelet (haar type) transform of a 1 sec audio clip can be computed in simulink. I mean which block is required. regards Morgan J.

Hi guys I really need help designing FM system using PLL as a demodulation. I have matlab code below and it is not working at the ...

Hi guys I really need help designing FM system using PLL as a demodulation. I have matlab code below and it is not working at the filtering end am I doing everything right? I basically how do I set the filter up to work? The way I set this up would be a PLL as a demodulation right? THanks guys [Y,Fs,NBITS,OPTS] = mp3read('jem.mp3'); snd=Y(1:30000

hi, i need help in sampling(matlab coding) a four channel AM modulated signal each at carrier frequencies 400Hz,1000Hz, 1600Hz and 2600Hz...

hi, i need help in sampling(matlab coding) a four channel AM modulated signal each at carrier frequencies 400Hz,1000Hz, 1600Hz and 2600Hz respectively... how will i sample my signal in such a way that 400Hz signal spectrum gets repeated at zero carrier frequency, so that i can apply a LOW PASS FILTER to extract my very first channel... kindly help me out... regards Nosheen

I implemented the "Recursive LSL Algorithm Using A Posteriori Estimation Errors" described in Haykin's "Adaptive Filter Theory" book. However, I'm...

I implemented the "Recursive LSL Algorithm Using A Posteriori Estimation Errors" described in Haykin's "Adaptive Filter Theory" book. However, I'm not getting the performance. I have seen published papers using the same algorithm and implemented them, but I still didn't get it right. Can someone point me to a verified implementation of the Recursive LSL or give me some pointers on where the most c...

>Hi .... We are developing a nonuniform frequency sampling FIR digital filter (lowpass) with linear phase using Matlab5.3 and DSP...

>Hi .... We are developing a nonuniform frequency sampling FIR digital filter (lowpass) with linear phase using Matlab5.3 and DSP toolbox (I dont remember the version of the DSP), however, We have arrived to a stuck point in the code becouse i am not very pro in Matlab. We have de

Hi Premkiran Mannava, The lms algorithm for adaptive filtering is not that complex to implement. just get hold of this book - ...

Hi Premkiran Mannava, The lms algorithm for adaptive filtering is not that complex to implement. just get hold of this book - Adaptive filtering by Simon Haykin hope this helps. logesh. > Premkiran Mannava- > >

Hi, here is the quesiton: I have a simple linear phase FIR LPF designed by windowing using hamming window, obtained its mathematical expression...

Hi, here is the quesiton: I have a simple linear phase FIR LPF designed by windowing using hamming window, obtained its mathematical expression (lets call it h[n]). Now, decimating the sequency of the coefficients by 2, i.e. making new filter to be h'[n] = h[2n] (an new n sequence runs from 0 to original n/2, so if original LPF length is 51 (order 50), the new will run from 0 to 25)) and che...

Hi, I have a few doubts regarding the calculation of bandwidth for various Analog/Digital modulation, please correct me if I have gone...

Hi, I have a few doubts regarding the calculation of bandwidth for various Analog/Digital modulation, please correct me if I have gone wrong Fm=Frequency of message signal. deltaF=Freq.deviation SR= Symbol Rate AM = 2 * Fm FM = 2 * deltaF + Fm 2FSK = 2 * deltaF + SR 4FSK = 2 * deltaF + SR BPSK = SR QPSK = SR Will the Root Cosine Filter at the transmitter side will effect t...

Hello, I am beginning to learn dsp and would very much appreciate some help regarding adaptive filtering. I have an input...

Hello, I am beginning to learn dsp and would very much appreciate some help regarding adaptive filtering. I have an input signal and an output signal. I would like to design a filter such that when convolved with the input signal, the output signal will be produce

I was wondering if any of you are familiar with filtering a plot. Here's my problem, I can plot the points, but I need to filter...

I was wondering if any of you are familiar with filtering a plot. Here's my problem, I can plot the points, but I need to filter the plot with the function y(n)=1/3[x(n)+x(n-1)+x(n+1). If anyone knows this, I will really appreciate it if you can help me.   Thanks

Hi all, I am using kalman filters for localisation of a robot. I am trying to implement the propagation and updation...

Hi all, I am using kalman filters for localisation of a robot. I am trying to implement the propagation and updation equations of a kalman filter shown below, using MATLAB. xhat(k+1) = phi * xhat(k) where phi is a state transition matrix.

dear all ? the 802163c-01_29r4.pdf document propose a code to model SUI channel . at end of code we have matrix that have 3 row (3 tap ) and...

dear all ? the 802163c-01_29r4.pdf document propose a code to model SUI channel . at end of code we have matrix that have 3 row (3 tap ) and N=10000 column . these are coefficient for three tap , but i don't know how could i use?it to filter my ofdem signal with? sample period of for example T.( why tap delay not use where should i enter it )? please help me ,really i need it. N = 10000; % n...

please check the following programs, the 1st is used to call on a .wav file, change it into a matrix, fft it, filter it, and then rewrite it, but...

please check the following programs, the 1st is used to call on a .wav file, change it into a matrix, fft it, filter it, and then rewrite it, but the problem is the new file is just a file, there is no sound?? the 2nd is the same but without filtering and the same problem arises, I will really appreciate any help !! :) thanks

Hello everyone. Could I have some discussion, if you are interested, please? In communication books such as Proakis, there are formulas for...

Hello everyone. Could I have some discussion, if you are interested, please? In communication books such as Proakis, there are formulas for MLSD or Viterbi equalization. On the other hand, if we have a matched filter at the receiver, we'll have a correlated noise sequence. Without whitening the correlated noise, can we still use a Viterbi equalizer? In other words, can the Viterbi equal...

I have a signal measured from a mechanical system. What I want to do is shift the signal in phase, but leave the magnitudes the same. I guess...

I have a signal measured from a mechanical system. What I want to do is shift the signal in phase, but leave the magnitudes the same. I guess this would basically just be an allpass filter (???). The amount of phase shift is dependant on frequency, but not linearly. Each frequency component is shifted by a different amount. What is the most appropriate way of doing this? I've tried searching...

Please help me, I have been struck here for a long time. Here is my problem I have a real signal in the time domain.I need to remove...

Please help me, I have been struck here for a long time. Here is my problem I have a real signal in the time domain.I need to remove a band of frequency from it to get a new real signal. So first I FFT this input real signal and multiply the FFT ouput by a filter function I chose from matlab. I hope I can get a new power spectrum with that

Hi, I'm recording motor cortex NIRS signals from the human brain where the heart rate(~1.5Hz), respiration(~2Hz) and Mayer wave(~0.1Hz) cloud a...

Hi, I'm recording motor cortex NIRS signals from the human brain where the heart rate(~1.5Hz), respiration(~2Hz) and Mayer wave(~0.1Hz) cloud a lot of the motor signal that is desired. The filtering technique has to be very phase neutral (like FIR not IIR filtering) unlike the filter.m where a dramatic affect occurs in phase shifting. Filtfilt.m is the usual one that is used that is effec...

I have Matlab 2008a installed on a Server housing 2 Intel Xeon Quad core processor (i.e. a total of 8 cores) and 64 GB RAM. When I perform...

I have Matlab 2008a installed on a Server housing 2 Intel Xeon Quad core processor (i.e. a total of 8 cores) and 64 GB RAM. When I perform operations like matrix multiplication, matrix inverse, Schur product of two matrices I can see all the 8 cores being utilized in the system monitor tool. However the fft(A) routine (where A is an N by N real matrix and N=8192, a power of 2) runs only on a singl...