Search Engine **www.edaboard.com**

## Sampling Signal Using Matlab |

Are you looking for?:

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

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: **761**

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: **1097**

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: **3176**

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: **1429**

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: **1835**

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: **6406**

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: **2433**

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: **4320**

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: **3209**

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: **4142**

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: **1667**

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: **926**

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: **780**

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: **1455**

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: **2666**

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: **1386**

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: **4922**

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: **2011**

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: **6693**

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: **2971**

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

Digital Signal Processing :: 16.09.2010 07:48 :: horizon12 :: Replies: **3** :: Views: **3363**

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: **1560**

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: **1974**

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: **842**

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: **1544**

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 some problem with the (...)

Digital Signal Processing :: 19.04.2012 07:37 :: gauravnanda18 :: Replies: **2** :: Views: **2546**

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: **761**

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: **322**

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: **706**

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: **1376**

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: **2389**

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: **1783**

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: **4385**

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: **1493**

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: **1012**

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: **3917**

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: **1400**

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: **1793**

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: **1327**

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: **1543**

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: **1210**

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: **1915**

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: **654**

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: **862**

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: **690**

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: **1271**

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: **898**

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: **852**

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: **1509**

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: **2717**

Last searching phrases:

talus | substrate model | tektronix tds | magic tee | pic book | digital image processing books | patch efficiency increase | artificial neural | ofdm frequency | ee214 notes

talus | substrate model | tektronix tds | magic tee | pic book | digital image processing books | patch efficiency increase | artificial neural | ofdm frequency | ee214 notes