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## Sampling Signal Using Matlab |

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sampling signal matlab , signal energy sampling , sampling sinusoidal signal , signal sampling simulink

103 Threads found on edaboard.com: **Sampling Signal Using Matlab**

kirjmaru, Do you try Gibbs **sampling**, yet ?

Digital Signal Processing :: 26.05.2013 10:43 :: phongphanp :: Replies: **6** :: Views: **701**

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

RF, Microwave, Antennas and Optics :: 08.02.2005 19:07 :: brmadhukar :: Replies: **3** :: Views: **1072**

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 (...)

Digital Signal Processing :: 19.06.2005 12:08 :: heab :: Replies: **6** :: Views: **3144**

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 (...)

Digital Signal Processing :: 20.12.2006 05:08 :: helios :: Replies: **1** :: Views: **1407**

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 (...)

Digital Signal Processing :: 24.02.2007 13:05 :: tarakapraveen :: Replies: **3** :: Views: **1799**

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** (...)

Digital Signal Processing :: 24.03.2008 02:09 :: triquent :: Replies: **4** :: Views: **6332**

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.

Digital Signal Processing :: 23.03.2010 08:50 :: aka07 :: Replies: **4** :: Views: **2398**

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

Digital Signal Processing :: 11.02.2005 12:44 :: brmadhukar :: Replies: **3** :: Views: **4260**

Hi,
you need to set the step variable in the settings as the **sampling** time interval.
hope this helps
brmadhukar

Electronic Elementary Questions :: 14.02.2005 11:37 :: brmadhukar :: Replies: **5** :: Views: **3126**

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

Analog Circuit Design :: 17.04.2006 23:14 :: tlihu :: Replies: **7** :: Views: **4097**

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.

Software Problems, Hints and Reviews :: 27.08.2005 05:55 :: chihwt2003 :: Replies: **4** :: Views: **1619**

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 **signal**s in **matlab**. Most of the well known **signal**s 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 (...)

Digital Signal Processing :: 25.01.2006 04:47 :: emrek :: Replies: **3** :: Views: **906**

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

Digital Signal Processing :: 28.08.2007 15:01 :: nnm :: Replies: **2** :: Views: **770**

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 (...)

Electronic Elementary Questions :: 14.11.2007 12:44 :: hbaocr :: Replies: **4** :: Views: **1425**

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 (...)

Digital Signal Processing :: 19.12.2008 00:27 :: MynameNayface :: Replies: **3** :: Views: **2632**

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 (...)

PC Programming and Interfacing :: 17.06.2009 06:49 :: engr_najam :: Replies: **1** :: Views: **1369**

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 **signal**s 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 (...)

Hobby Circuits and Small Projects Problems :: 12.07.2009 01:45 :: engr_najam :: Replies: **10** :: Views: **4860**

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.

Digital communication :: 24.07.2009 12:34 :: Mohamed El-Shimy :: Replies: **5** :: Views: **1966**

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 (...)

Digital communication :: 24.08.2009 16:10 :: Aya2002 :: Replies: **13** :: Views: **6639**

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.

Digital Signal Processing :: 18.07.2010 18:22 :: JoannesPaulus :: Replies: **5** :: Views: **2919**

Your PSD is correct and by default it plots it to the half of the used **sampling** frequency. To only observe the region from 0 Hz to 30 Hz you need to re-set the axis of the figure.
To do this first check what is the number of the figure your PSD plot is drawn to. This can be seen in the top part of the plot-window (for me its Figure 2). You probably

Digital Signal Processing :: 17.09.2010 03:34 :: veryADD :: Replies: **3** :: Views: **3272**

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 conf**using**.
Here's my **matlab** code and also attached:
e = zeros(1,1000); % Initializing the error **signal**
wc = 2*pi*95/800; % Omega for the (...)

Electronic Elementary Questions :: 31.10.2010 01:13 :: sixers0130 :: Replies: **0** :: Views: **1533**

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...

Digital Signal Processing :: 22.01.2011 20:32 :: neera :: Replies: **2** :: Views: **1930**

here's the easiest method.
periodogram(x, ,'onesided',512,fs);
fs is the **sampling** rate.

Electronic Elementary Questions :: 19.05.2011 13:46 :: ninju :: Replies: **2** :: Views: **824**

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

Digital Signal Processing :: 14.09.2011 05:46 :: Ziko87 :: Replies: **2** :: Views: **1510**

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

Digital Signal Processing :: 07.12.2012 10:10 :: ajex :: Replies: **2** :: Views: **2478**

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 )

Digital Signal Processing :: 03.05.2012 05:36 :: victor.yu :: Replies: **2** :: Views: **695**

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

Microcontrollers :: 25.10.2012 11:44 :: kanni1303 :: Replies: **2** :: Views: **267**

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** (...)

Digital Signal Processing :: 25.11.2012 07:19 :: electronicstudent :: Replies: **3** :: Views: **662**

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

Digital Signal Processing :: 28.07.2013 13:54 :: akji890 :: Replies: **3** :: Views: **1315**

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

Digital Signal Processing :: 18.01.2004 12:12 :: eda-bond :: Replies: **4** :: Views: **2361**

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

Analog IC Design and Layout :: 06.10.2004 17:54 :: tucura :: Replies: **3** :: Views: **1757**

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 (...)

Digital Signal Processing :: 15.09.2004 08:18 :: Circuit_seller :: Replies: **15** :: Views: **4335**

it is simple.
use simulink's sinwave block to generate a sinwave based **sampling** mode, and do fft to it

Analog Circuit Design :: 27.12.2004 21:55 :: sadfish :: Replies: **13** :: Views: **1460**

any idea on how to develop the software?
can we use **matlab** to **sampling** the voice **signal**?
somebody plz help me

Hobby Circuits and Small Projects Problems :: 02.08.2006 03:45 :: razwell :: Replies: **4** :: Views: **990**

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.

Analog Circuit Design :: 18.08.2005 01:45 :: mr_chip :: Replies: **8** :: Views: **3850**

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

Analog Circuit Design :: 23.08.2005 22:10 :: sunking :: Replies: **8** :: Views: **1379**

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

Digital Signal Processing :: 06.12.2005 13:54 :: chvti :: Replies: **0** :: Views: **1761**

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.

Microcontrollers :: 19.06.2006 00:24 :: waseem :: Replies: **2** :: Views: **1302**

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 (...)

Digital Signal Processing :: 19.12.2006 02:59 :: wajahat :: Replies: **1** :: Views: **1514**

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 (...)

Digital Signal Processing :: 21.04.2007 15:58 :: vjfaisal :: Replies: **2** :: Views: **1194**

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

Digital Signal Processing :: 21.08.2007 00:35 :: naresh850 :: Replies: **7** :: Views: **1894**

hi
if anyone knows some about the down**sampling** , please explain it to me....
Q: when we down sample our data, our original data could be lost or not???
regards

Digital communication :: 20.09.2007 16:44 :: vjfaisal :: Replies: **3** :: Views: **638**

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 (...)

Digital Signal Processing :: 18.11.2007 23:31 :: ~farah_r~ :: Replies: **0** :: Views: **843**

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!!!

Digital Signal Processing :: 15.03.2008 11:27 :: hweontey :: Replies: **0** :: Views: **676**

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 (...)

Microcontrollers :: 23.03.2008 17:21 :: piskot :: Replies: **1** :: Views: **1238**

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?

Analog Circuit Design :: 10.09.2008 21:54 :: jeremy_zhu :: Replies: **2** :: Views: **878**

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 (...)

Analog Circuit Design :: 18.09.2008 11:24 :: lhlbluesky :: Replies: **4** :: Views: **838**

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 (...)

Digital Signal Processing :: 05.01.2009 09:27 :: lionelgreenstreet :: Replies: **0** :: Views: **1488**

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 (...)

Digital Signal Processing :: 17.01.2009 17:13 :: eng_shady00 :: Replies: **9** :: Views: **2675**

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