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45 Threads found on edaboard.com: Low Pass Filter Code In Matlab
hi, how to write matlab code for RC 1st order low pass filter?? i saw many matlab codes from internet , like butterworth, chebychev... but don't have the one for RC low pass filter?? pls advice how to do it? i was (...)
how to design -a LPF(fir) by window method sampling freq=100hz and cut off=20khz..without using FFT builtin function. plz can u give me a code(program) to implement this using matlab softwear.please.
required matlab code for low pass filtering of a digital sequence for e.g i want how to found the value of this every elemnt in it after passing through low pass filter i requesting matlab code for it
hi. I have written a code which emulates a qpsk tx n rx system. it works perfectly and i'm getting the BER curve exactly. As my next task i have been asked to design an fir low pass filter through which i am to pass the qpsk signal. i'm not getting much assistance from the guide and i'm pretty much on (...)
Hello, I have a signal vector and I wanted to low pass filter it. My knowledge of discrete filters is very limited so any help would be greatly appreciated. My signal is basically a sampled instance of a sum of sinusoids of 100Hz and 1KHz. What I want to do is basically filter this sampled (...)
Use the following code for creating a low pass filter : Determine the cut off frequency and change Fc appropriately Fc = 500 ; %cut off frequency w = Fc/(2*fs); % Normalized frequency %design a low pass fitler with above mentioned specifications (...)
phase is defined as -arctan(w/wc), where w is frequency and wc is cut-off frequency. Whatever we have instead of w/wc, phase should not have the value lower than -90 deg. Would be true only for a first order low-pass.
I need a matlab code to promt the user to enter a sinusoid signal ex: x= 3*sin(2*pi*50*t) + 3*sin(2*pi*100*t) + 3*sin(2*pi*200*t) + 3*sin(2*pi*400*t) and if any other variables are needed from the user. then display the signal in time domain and frequency domain. and then apply a low pass filter, to (...)
I have one problem. I have a data vector. for example : data=rand(1,n) ; n=100; I want to obtain discrete fourier transform of it by DCT command. and then,by a low-pass filter,remove unwanted high frequencies of larger than f1. f1 is determined. and finally , again convert filtered data from discrete frequency domain to (...)
Hi How to use matlab write a low_pass filter .. I can use simulink for simulation , but I want to know how to write a matlab code.. simulink file can novert to matlab code or not ?? a low pass (...)
I just designed an FIR filter in matlab, and saw the impulse response. I obtained a low pass filter. Up to now there were no problems. I read the cut-off frequency(0.144cycles/sample) and had 2 null frequencies at 0.325cycles/sample and 0.5cycles/sample. Then I inputted a discrete sine wave block(with (...)
hello, can anyone provide me with the code or implementation ideas for low pass filter of particular cut-off frequency and notch filter without using inbuilt functions on matlab ... please thanks in advance ...
Hello, could you guys please give me an explanation why if I low pass filter an upsampled (zero stuffed) ramp signal I get some oscillation which increases in amplitude with time? The time frame that I am looking is very big in comparison to the response length of the filter, in other words, I am NOT seeing just the (...)
here the low pass filter code with NUM and DEN designed by matlab (sptool) that fsample=1KHZ;Fc=40Hz;ButterWorth IIR filter #define pi acos(-1) double Numerator={0.00018321602337,0.000732864093479,0.001099296140218,0.000732864093479,0.00018321602337}; double (...)
hello everybody,now i need to wirte the matlab code to realize timing recovery and baud recovery for qpsk signal.I have read the Garnder algorithm.But I can not apply it to code matlab program.anyone can draw code flow chart for me or explain the algorithm in detail.Thank you very much.
What exactly is wrong? As far as I can tell, you don't have a "canonical" two pole low-pass filter because your transfer function has a pole at zero, another pole and a zero.
Hi guys , i need some advice/tips/guidance. Pls share. So , i am supose to design a low pass filter with a certain cutoff frequency and i'm suppose to filter those high frequency component of the sound in time domain(convolution) and freq domain(multiplication). So , i am not sure how to do it..i took a snippet from (...)
I'm pretty sure I'm screwing this up pretty badly due to my limited understanding of matlab but I am trying my best. A passive low pass filter was assembled and measured the output via an o-scope. At each frequency from the signal generator we wrote down the input/ouput Vpp. Using that data I am trying to (...)
I have one problem. I have a data vector. for example : data=rand(1,n) ; n=100; I want to obtain discrete fourier transform of it by DCT command. and then,by a low-pass filter,remove unwanted high frequencies of larger than f1. f1 is determined. and finally , again convert filtered data from discrete frequency domain to (...)
Hi all, I have a waveform as shown below, I have clearly mentioned in this waveform the imformation which is neccessary for me. Now I want to remove all other noise and wish to see only these sharp pulses without noise.The data is available with me and I want to use matlab for this purpose, I need to know how can do this. thanks and regards,
Hi There is a book named "Microwave Devices & Circuits" by Bhal & Bhartia. You can find it very usefull. I have done implementation of low pass filter (Butterworth & Chebyscheff). Refer the code attached.
Any idea how to code in matlab to restore the following images which have been degraded by an addictive
Hi..... you know your design better.. so asking a code :?: i suggest some steps.. follow that.. its easy to use matlab... in command prompt type >> fdatool i.e filter design automation tool... just fill out your required bandpass or low pass (...)
my Rf signal is 102.3 Mhz and it is fed to mixer and I need IF as 5Mhz so i set the local ocillator to 97.3Mhz to obtain the 5Mhz that is my IF. I have to pass this IF through low pass filter and then decimation and obtain the same IF frequency as 5Mhz. But my sampling frequency is 409.8Mhz for filter (...)
Hi, I have a matlab code here it acquires FM modulated signal from sound card can perform FM demodulation however, i don't why it set vcok (VCO K constant) at 0.176 anyone can explain to me how and why it is set to 0.176? Fc = 2144; %select VCO carrier frequency vcok = 0.176; %select vco constant Fs = 40000; %select sound card sampl
Baseband signal can be generated in number of ways . Refer a standard communication text book to represent band pass signal interms of baseband signal. One simple way to generate a arbitrary baseband signal is to generate a PRBS pattern and pass it through a low pass filter with appropriate cut off (...)
hi All, I am doing my thesis on adaptive a first step towards the project...I need to write a matlab code for the low pass FIR filter.. Which means that I have to write a code for the following equation y(n)=∑_(i=0)^Lb_i x(n-i) where x is the input to the (...)
I think, you got integer overflow inside filter. Check that signal's amplitude and frequency are in valid range. Decrease amplitude. Try to analyze filter output as integer numbers (not dB representation)
We would need to see what you mean. Can you post an image of the spikes? A simple low pass filter may fix it, but it depends on the underlying signal. Also, the problem may be in your circuitry so the solution would be to fix the problem rather than try to filter the signal afterwards. A bit more detail would therefore (...)
Hi, I have written a test program in matlab for lowpass fir, when i m plotting its magnitude response after fft, it seems to be a high pass filter. Why its doing so?? My code is as follow: N=50; %filter order Length=N+1; %length n0=N/2; %delay (...)
Hi, I have written a test program in matlab for lowpass fir, when i m plotting its magnitude response after fft, it seems to be a high pass filter. Why its doing so?? My code is as follow: N=50; %filter order Length=N+1; %length n0=N/2; %delay (...)
Hey guys , help me to check this code.. i cant figure out how to make it run .. T.T Its just to filter out some high frequency in the song file i cut .. function filter_a() = wavread('cut2'); N = length(song); %length of song is 99093 k=; f1=k*fs/N; cf = 2000 ; n = ; t = n/cf; N = length(t); f = n*cf/N; hf = [o
I have ISE but not System Generator, is a digital filter implementable on a 30K gate FPGA? What is the minimum number of gates to generate a decent filter for low frequencies such as a band pass filter less than 100Hz. - Jayson
For my fir filter,I need a low taps fir for implementation.I had tried windows and remez method,but the taps is too high. I heard that optimization fir filter design method is good,anyone can provide some program or papers about it to help finish the design.
Hi! I have trouble to understand how a digital PLL for discrete valus work. Say that we sample a 1 MHz sinus wave with an ADC 12bit/15 Msps. Then we hava a bunch of values to do signal procesing on. What I do not understand how each sample (dscrete value) should be processed in a digital PLL. The first sample will be multiplied with
Thanks for your help first. Then, I jsut used C for a reference model. Here I try to design a LPF. I mixed 1.5M and 10M as input. The LPF pass band is set to 1.5M. I can see the freq. of the output is less than that of the input, but not equal to 1.5M low-freq input. Is this result right? How to verify the detail performance requirement?
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 (...)
A script with a strip down version of a DMT Adsl system. Description: The following is a simulation of the DMT system, similar to implementation as done in ADSL systems. The bandwidth has been divided into 32 sub-channels of equal width. Each sub-channel has been allocated bits in the range of 1-15 bits/symbol/Hz. In a finite length DMT sy
Hi All, I have this one small audio file which is corrupted by an unknown interference signal which needs to be filtered out. I loaded the file in matlab and found that its sampling frequency is 44.1kHz...I then plotted its FFT and found two peaks at + and - 21.99 kHz...So, I figured that I need to design a filter to remove the (...)
Dear All, I am trying to implement an FIR band pass filter using fir1() function but I am confused in normalizing the frequencies.Whether I had to divide my frequencies by fs or fs/2 for normalization to be used as Wn. Which one will give me the correct results?I had divided my band freq range by fs/2
the first test is to do time domain tests. An impulse like 0x40, 0,0,0,0,0 ... this will give scaled versions of the coefficients. This is a good test signal for systems that perform rate-changes as well. This is mainly to show the filter was correctly translated to HDL. Variations on this can be done. Ideally, if the coefficients are correct,
Hi, I am working on EEG signal. At first i applied the Butterworth low pass filter to extract 0-64 Hz frequency. Then i applied DWT to extract BETA (16-32Hz) and ALPHA(8-16Hz) wave . So, according to theory , 2nd and 3rd level coefficient of DWT should provide the beta and alpha wave. But when i performed the fft of D3 wave i did not get (...)
x1 = load('ecg3.dat'); x2=x1; fs = 1000; % Sampling rate N = length (x2); % Silength t = /fs; % tiidx figure(1) subplot(2,1,1) plot(t,x1) xlabel('second');ylabel('Volts');title('Input ECG Signal') % Cancellation DC drift and normalization x1 = x1 - mean (x1 ); % cancel DC conponents x1 = x1/ max( abs(x1 )); % no
Hi qgimol I don't have working code but would like to propose a procedure. You can start fom white noise. One with uniform PDF can be generated quite simple using the congruent algorithma or just use classical PRBS with shift register and xor beed backs. Then you need pink like filter in order to shape this noise. You can design one
Hi Manikanta some explanation about Group delay %Group delay: % % Imagine an AM signal with a sinusoidal modulation. The carrier is high-frequency and the modulation is low-frequency. Compare the AM signal with the unmodulated carrier: they are in phase (the zero-crossings are coincident). % Now, let pass the AM signal by a linear network