I came across the following problem in FIR design using window and convolution techniques . Project a FIR filter using a Hamming...

I came across the following problem in FIR design using window and convolution techniques . Project a FIR filter using a Hamming window to achieve the following specifications: Passband 0,3-3,4 kHz Stopband 0-0,2 e 4-8 kHz Stopband atenuation > 25 dB Sampling frequency 32 kHz Since the transition band is different

Hello folks, I would really appreciate your insight in terms of how to design digital filters to match both magnitude and phase...

Hello folks, I would really appreciate your insight in terms of how to design digital filters to match both magnitude and phase of a narrow band complex Analog gain(simple 1st order or 2nd or 3nd order gain). This is for impedance matching in voice band. I understand Matlab ca

Dear all, any one know how to set this>?[T,X,Y] = SIM ('model',TIMESPAN,OPTIONS,UT)..i read the help sim already. But i ...

Dear all, any one know how to set this>?[T,X,Y] = SIM ('model',TIMESPAN,OPTIONS,UT)..i read the help sim already. But i dont realli understand it. How can I set my output as cell array? My model is a filter design. The output is too long already. How can I

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

hello, im new to the topic of adaptive filters and its implementation in matlab. im doing a project on 'filtering and analysis of an ECG wave'. 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 200hz..the file is in .dat format. we have been able to impl...

Hello, I have got a theory on this link of filters http://ece-www.colorado.edu /~ecen2260/slides/FilterSlides.pdf It...

Hello, I have got a theory on this link of filters http://ece-www.colorado.edu /~ecen2260/slides/FilterSlides.pdf It gives in a formula a relationship between wo and wc. wo should be chosen in suc

Hello, I am simulating OFDM transmission over a Fading Channel in Matlab (using the 2009a version) at my university I am concatenating 100...

Hello, I am simulating OFDM transmission over a Fading Channel in Matlab (using the 2009a version) at my university I am concatenating 100 OFDM symbols, where one symbol has 80 samples (64 subcarriers and 16 cyclic prefix). I create a Rayleigh fading channel object and filter my serial stream of OFDM symbols with the fading channel. Now, the problem is that I want each OFDM symbol be ...

> Hi, > I am wondering is there any way to remove random spikes in the data.? > If someone can help me I can send out a sample of my data...

> Hi, > I am wondering is there any way to remove random spikes in the data.? > If someone can help me I can send out a sample of my data set. > Your help is highly regarded > Thanks > Thisara Thisara, There are a lot of ways you can remove random spikes in data. You can do some kind of running average (i.e., a lowpass filter), you can simple remove data points which exceed some thre

Its fun trying to implement LMS on Matlab, but if you are pressed for time, there should be couple of examples on the web , for...

Its fun trying to implement LMS on Matlab, but if you are pressed for time, there should be couple of examples on the web , for instance http://www.utdallas.edu/~golden/ MATHANN/ANNSOFTWARE.html or can try

Hi I'm trying to compare performance with and without minimum phase prefilter for 8PSK equaliser with a DDFSE. When we employ a prefilter,...

Hi I'm trying to compare performance with and without minimum phase prefilter for 8PSK equaliser with a DDFSE. When we employ a prefilter, we use min phase all pass prefilter for 2nd half and max phase all pass prefilter for 1st half. Hmin_ap , Hmax_ap would be these all-pass filters for filtering data, and Hmin and Hmax would be channel estimates to give to DFSE, correspondi...

Hi tu you all! I have a problem of distortion while filtering. I have a signal with bandwith of 10 KHz centered on a carrier frequency of 30...

Hi tu you all! I have a problem of distortion while filtering. I have a signal with bandwith of 10 KHz centered on a carrier frequency of 30 KHz without noise. I'm trying filtering that with a butterworth filter of signal bandwith of 40 KHz centered on 30 KHz using [Bf1,Af1]=butter(5,[wn1 wn2]); y=filtfilt(Bf1,Af1,x); But if i try to compare x and y they are different even if the filte...

Bonjour. I have an signal(rough.wav) which is impulse response of a system. The impulse response should ideally be like smooth.wav...

Bonjour. I have an signal(rough.wav) which is impulse response of a system. The impulse response should ideally be like smooth.wav .i.e. A smooth damped harmonic oscillation. Please tell me how to filter rough.wav to get smooth.wav , Both these files are in signals.zip in the files s

Hi, I am an Mechanical student and i am trying to generate a random rough surface (with specified ACF and Std Deviation) using 2D FIR filter...

Hi, I am an Mechanical student and i am trying to generate a random rough surface (with specified ACF and Std Deviation) using 2D FIR filter in Matlab. I come across a paper that helps me doing that. For that, first i need to generate a input sequence composed of independent random numbers {?(I,J)}. The paper says its PSD must be a constant. I used matlab to generate independent r...

Hi , I am working with 16-QAM transmitter/receiver . The problem is I encounter some extra symbols after downsampling ( INPUT - 1000 SYMBOLS ,...

Hi , I am working with 16-QAM transmitter/receiver . The problem is I encounter some extra symbols after downsampling ( INPUT - 1000 SYMBOLS , OUTPUT-1012 SYMBOLS). Let me say in detail. I send 4000 bits/s -> *QAM Coder* -> 1000symbols/s -> *upsampling* (factor=8 , length becomes 8000 samples) -> *Transmit SRRC* ( delay 6, filter length = 97 samples) -> *pulse shaped output* (length = 809

I'm to design a 2nd order, N=2, butterworth HPF starting from the normalized transfer function for a...

I'm to design a 2nd order, N=2, butterworth HPF starting from the normalized transfer function for a LPF: HN(s)=1/(s^2+1.414*s+1) Given: T=0.004 ?c=0.8 pi wc=192 pi To find the transfer function of the HPF, substitute s for wc/s: H(s)=HN(wc/s)=s^2/(s^2+853*s+363833) Here's my MATLAB code: T=.8/192; Fs=1/T; nums=[1 0 0]; dens=[1 853.03 363833.1]; figure [Hws,w]=freqs(nums

Hi to all: I have this program. My question is at the end of this program below. clf; h=[3 2 1 -2 1 0 -4 0 3]; x=[1 -2 3 -4 3 2...

Hi to all: I have this program. My question is at the end of this program below. clf; h=[3 2 1 -2 1 0 -4 0 3]; x=[1 -2 3 -4 3 2 1]; y=conv(h,x); n=0:14; subplot(2,1,1); stem(n,y); xlabel('Time index n');ylabel('Amplitude'); title('Output Obtained by Convolution');grid; x1=[x zeros(1,8)]; y1=filter(h,1,x1); subplot(2,1,2); stem(n,y1); xlabel('Time index n');ylabel('Amplitude'); ...

Dear all my name is payam, my research area is on control noise with adaptive filter but as you now the LMS block has an input link with the...

Dear all my name is payam, my research area is on control noise with adaptive filter but as you now the LMS block has an input link with the name of desired input, how can i define the desired input for this block if i want to control the noise in unknown models? with best wishes payam

Hi, I am working on the 16-QAM modulator & demodulator. I need some help in that. 1.I want to find the received energy per symbol Es. I...

Hi, I am working on the 16-QAM modulator & demodulator. I need some help in that. 1.I want to find the received energy per symbol Es. I read in a book that we can transmit one single symbol and calculate the received signal energy(sum(x.^2))to find the energy per symbol.But in 16 qam I have two different energy level 1 and 3 .so how do I do it? 2.I want to scale my carrier and filter ga...

Hello friends, I have a question regarding to the relationship between step response and group delay. I have designed a FIR filter with...

Hello friends, I have a question regarding to the relationship between step response and group delay. I have designed a FIR filter with fdatool with following characteristics. Taps = 64 Fs = 10000 Fpass = 100 Fstop = 200 In step response screen it shows me 6.4ms delay before I get the actual value. However the group delay shows 32 samples... when we multiply 32 samples with the sam...

I am trying to design a multiband Kaiser window with passband cutoff freqz of band1(3000-7000)Hz band2 (10000 - 13000)Hz and...

I am trying to design a multiband Kaiser window with passband cutoff freqz of band1(3000-7000)Hz band2 (10000 - 13000)Hz and Stopband Cutoff freqz of band1(2000-8000) and band2 (9000-14000) Pass band ripple <= .01 (all passbands) Stopband attenuation <= 40dB (all

HERE IS THE CODE WITH SAMPLE FILE  ANC COMPARING BETWEEN THE FILTERED AND  NOISY SIGNAL  YOU CAN REMOVE WHATEVER IS NOT NECESSARY...

HERE IS THE CODE WITH SAMPLE FILE  ANC COMPARING BETWEEN THE FILTERED AND  NOISY SIGNAL  YOU CAN REMOVE WHATEVER IS NOT NECESSARY   [Yo,FS,NBITS1]=WAVREAD('1o.wav'); % read original signal [Yn,FS,NBITS2]=WAVREAD('1n.wav'); % read corrupted signal n1 =length(Yo);%Obtain the length of the original signal n2 =length(Yn);%Obtain the length of the corrupted signal % Time Domain Plot f...