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103 Threads found on Sampling Signal Using Matlab
kirjmaru, Do you try Gibbs sampling, yet ?
hi, you may have to evaluate the function h(t) at your sampling intervals and then use it. Thus the length of the filter will vary. This is assuming you are using sampled data. If you want analog response, you should use symbolic toolbox.(or something like that) brm
I have some problem about Multirate signal Processing.(by using matlab) 1.) Design a standard sample rate converter to convert the signal x at sampling rate 8 kHz to y at sampling rate 22 kHz. Put the interpolator first and decimator later. Combine the anti-aliasing filter with the post (...)
Be clear that in matlab you can do only FFT which is a efficient form of DTFT. now lets take a analog signal x(n)= A sin (2*pi*f*t); ( in a PC nothing is analog)... so lets make it Discrete ... and the sampling freq fs = ( 1/Ts) lets and fs > 2*fm .. fm = max freq in our input signal lets take Ts=0.1 (...)
Hi friends, I am trying to display time domain signal in Frequency domain using matlab.I took frequency of signal as 100hz and sampling frequency as 1000sampels/sec I have tried using 'fft' function but not getting. Anyone of you give me the solution for displaying frequency (...)
I am trying to calculate the cross-spectral density for a voltage signal using matlab. I am using the fft function to do the Fourier transform. The fft function is Y=fft(X,n). But I donot know how to pick the points n. My voltage signal Y is 2seconds long. dt is 0.0001s. so sampling (...)
Its easy. u should know the sampling rate of the signal. Then you can create a frequency vector like F=linspace(0,1,number of data points/2)*Nyqusit_limit . So find the index of the point in your FT with maximum value( which is your highest frequency) and look the value of F with the same index. So at last u will get the frequency value.
Hi, FFT(h(t)) is different from H(w) in your matlab implementation. This is because, H(w) is based on the continuous time and when you compute h(t) you will be using discrete values. What will be your sampling rate? BRMadhukar
Hi, you need to set the step variable in the settings as the sampling time interval. hope this helps brmadhukar
Try this FFT in spice or spectre: N/(FFT point) = Fin/(sampling freq) N prime number 7,9,13,17,19,23,29,31............. FFT point : power of 2 , 1024,2048,4096,32768,65536 , larger will give a lower FFT noise floor Fin : your input signal Let say, I have 1024 sampled data wit Fs = 10 MHz. How can I can FFT them us
Thanks. However, what I want is to output the signal after sampling out a data waveform using simulink, not importing wave file or any other audio file to play.
Hi, For learning matlab there are a lot of tutorials in internet. You can also find help from . I think first of all you must learn building signals in matlab. Most of the well known signals can be created by built in functions of matlab. For example '>t=0:0.01:1; y=sin(2*pi*10*t); plot(t,y)' will give (...)
i want to analyze frequency which is recorded in matlab using wavrecord command , what is the highest frequency (for select best sampling rate next time recording) Reduced noise Understand behavior of frequency plz anybody knows those thing post it for everybody's knowledge
I have a analog signal through the lowpass Filter(fc=40hz) then I sample it which fsample =40hz, and I recieve 200 pieces of that signal into my computer by sampling which f(sample) =40hz.And my problem is how I can analyse spectrum of those 200 pieces discrete signal in matlab. PLZ help me some code (...)
Hello, I have an audio signal which contains a total of 39922 samples with a sampling frequency of 8000 Hz.For a specific reason that relates to my project requirement, i truncate the audio signal to just 100 samples. I later increase the sampling rate to 112 kHz. Pass the signal through an (...)
i think it can be done by using ur sound card. there is demo already in matlab. the input should be given through microphone. sampling rate can be set between 8kHz and 44 kHz. giving signal instead of microphone i think you should first know the max and min voltage ranges of microphone signal, then remove (...)
u can record ur ECG signal with matlab sound card. but for that u will need some extra amplifing circuitary for the input signal, also the ECG signals are of low frequencies but by using sound card as ur data acquisition card, the sampling frequency must between 8KHz 44 KHz whic is very (...)
Hello, I think to reduce the BER, you have to begin sampling your received signal at its maximum value. This will give the maximum SNR.
My Friend, If you have matlab installed on your computer you can search with its help, for example see the following: fmmod Frequency modulation Syntax y = fmmod(x,Fc,Fs,freqdev) y = fmmod(x,Fc,Fs,freqdev,ini_phase) Description y = fmmod(x,Fc,Fs,freqdev) modulates the message signal x using frequency modulation. The (...)
Of course it works! If you say that your sampling frequency is 256Hz, your max frequency is 128Hz (as shown). You only need to know what is your actual sampling rate for your captured data and substitute it in Fs... I believe the picture you are showing is a zoomed version of the spectrum.
this is my code for using in matlab pwelch(y1,window,,,1000) %Welch's method which y1 is my EEG signal. my sampling frequency is 1000Hz for the EEG signal, I want to get a frequency range of 0-30Hz.but I'm getting f
hey guys this is my first time to post question here... I need to design a PLL by using matlab, but what my professor lectures in class is very confusing. Here's my matlab code and also attached: e = zeros(1,1000); % Initializing the error signal wc = 2*pi*95/800; % Omega for the (...)
hi all.. i am doing my final year project.. i am new in matlab.. now, i need to do the framing and windowing using matlab.. i have 4098 samples and i want to frame it in 256 points.. and my sampling rate is 173.61. i hope all of u can help me.. plz...
here's the easiest method. periodogram(x, ,'onesided',512,fs); fs is the sampling rate.
Hi, I must to design a Butterworth filter using matlab. Parameter: - butterworth Highpass filter - cut-off frequency 0.2-05 Hz - attenuation 70\80dB In a Workspace I have two vectors (a time vector and a data vector) Example: time=; data=; signal=plot(time, data); Data vector conta
what I have to do is to calculate the noise in a signal and see how it depends on the frequency spectrum. I am trying to calculate PSD of a signal but everytime, I get an error saying "vectors must be the same lengths". I am not able to find a solution. Here is command which I'm using. please let me know if there is
hi sunny, i mean to classify the shape of graph, for example figure 1 and to belong to group 1. figure 3 belong other group. 2nd question is how to make both (figure 1 and 2) in same length? by using sampling ? thank )
Hi Friends, Please help me how to sample a signal in frequency domain using PIC16F877A. as like we sample in matlab giving a input signal and sample. I need how to frequency sample a signal using PIC 16F877A. Thnx in Advance
Hi all, I have never done a lot with matlab but have an assignment to do the requires me to use stem functions to investigate natural sampling of a single base band frequency. I have got the sampling frequency done using the stem functions but am not sure how to go about combing the two. The info signal (...)
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, Can any body suggest from where i can get materials to model phase noise , frequency offset, adc sampling offset for a testing performance of a spread sprectrum receiver
A decimation filter uses a Cascaded-Integrator-Comb section followed by a FIR section. The CIC section decimates down to 4 times the output sampling frequency and has a response of the form (sin x over x)^n, n being higher than the order of the analog section. The FIR can be any linear phase design, and is used for antialias pourposes. If th
Hi If you want to delay signal in multiple of sampling period, you can easily use a buffer to delay samples. Also you can design a fractional band linear phase all-pass filter using matlab. If you want a linear phase in whole the band you can combine interploation and a buffer to delay signal (General (...)
it is simple. use simulink's sinwave block to generate a sinwave based sampling mode, and do fft to it
any idea on how to develop the software? can we use matlab to sampling the voice signal? somebody plz help me
I think you're sampling the digital data at the output of the pipeline of the ADC, so settling is not a problem. Just make sure that the digital values are exactly vdd & vss (or 1 and 0) - use rounding if necessary.
75dB SNR with sampling at 200Mhz in 0.35u is impossible. Or, if it is possible, you should be a very good designer to do it. But 0.18u or 0.13u process, the 3.3v process is 0.35u too. by the way, the max freq is 1.5MHz,while OSR is 64, so the sampling freq is about 100MHz
I have a set of random data that samples the noise voltage at 20us. Right now I am plotting PSD in dB. How do I plot it with respect with dBV? The following is my code: Fs=50000; datasize=size(RANDOM); numsample=datasize(1); numsample=numsample; FFTX=fft(RANDOM(:,2),numsample); X=FFTX(1:floor(numsample/2)).*conj(FFTX(1:floor(numsample/2
Hi! You want to sample the 5v ac signal and send it to the pc or the 5v DC signal. It is fairly easy to write the code and i will certainly help you but first clarify me. More over what is the sampling rate that you need i.e how many samples per secound. Regards.
Hi everyone , I am new to filter designing and I am using the filter designing toolbox of matlab (FDATool) to design a notch and a comb filter. I want to remove 50Hz frequency from my signal. My sampling frequency is 8000. I can?t understand few of the fields in FDATool GUI window. I have attached the images of window with (...)
hi , please help me in designing the simple filter : The signal use to bandpass is in text form (EEG) data, is also uploded, and it needs without using the FDA Tool. the given specification is given as: Use the tool matlab simple coding Q: sampling (hz): 200 cutoff1: 12 hz cutoff2: 18 hz (...)
for FIR, First u have to calculate coefficient h{n} = sin( 2*pi*n*f/FS)/(n*pi) where f = fut off frequency FS = sampling frequency n = 0 to M-1; M = no coefficient; lenth of impulse response or filter kernel length If M is more then ur roll off of freq response is sharper. M = 3.3*FS/(Fstop -Fpass) Fstop = stop band freq of transition b
hi if anyone knows some about the downsampling , please explain it to me.... Q: when we down sample our data, our original data could be lost or not??? regards
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 (...)
How to get the sampled sine waveform by using "Simulink" in matlab before the sampled signal enters into the digital FIR filter??? Any settings I need to set?? If the sampling frequency is 1kHz and cutoff frequency is 50Hz??? 10s!!!
hi everyone my teacher wants me to change the frequency of an audio file(in .wav) using microcontroller. is it possible? the wav file will be read from matlab using wavread. the data from matlab , consist of the wave vector, and sampling freq will be used. if it is possible how can i do it. can anyone (...)
add a ramp signal as the source signal. waht's the relation between the freq of ramp and the freq. of sampling? i means what freq of ramp is suitable for testing the static performance . and what about the freq of sin wave for testing the dynamic performance, what's the coherent freq?
i designed a pipelined adc,and i simulated it at the frequency of nyquist(about half of sampling frequency);but when i deal with the data(txt in dec) in matlab using fuction cal_snr.m(from maxim), the noise power(Pn) is negetive,why?and snr is complex.i'm really confused.i find when i change the parameter "span",the result is different;but (...)
I want to realize a simple analog low-pass first order filter in matl ab, but i waant also obtain time domain output if an input signal is present. So: 1)input time domain signal is realized with an array of values (i'm not using symbolic) 2)fft of input signal: i consider only frequencies equal and under fs/2 (with (...)
I designed a circuit to detect the ecg signal. I applied it to pc through the mic. input, then i applied it to matlab using the wavrecord function. I can select my sampling rate and the number of samples to record. Now,first i want to plot the signal against time. Can this be acheived by just the plot (...)