Search Engine **www.edaboard.com**

## Matlab Sampling Frequency |

Are you looking for?:

frequency components sampling , bit rate sampling frequency , sampling matlab , sampling theorem matlab

frequency components sampling , bit rate sampling frequency , sampling matlab , sampling theorem matlab

275 Threads found on edaboard.com: **Matlab Sampling Frequency**

Hi all,
I have a problem..I have a discrete sine with 20000 sample and with variable **frequency**.
I need to plot this freq sweep, I expect a **frequency** ramp.
How I can do this with **matlab**?
Please someone help me?
Thanks

Software Problems, Hints and Reviews :: 28.07.2009 10:09 :: silverit :: Replies: **2** :: Views: **6571**

Q1 :
sir i want to ask you that if i am using an external ADC which have 400ksps t0 sample the **frequency** of 100khz.
if i applied input of 100khz to ADC which converts into bits and these bits can serialy transmit to **matlab** through pic16fxxx, so it is possible for pic to work as fast to transmit all the bits before taking to next sample duration ti

Microcontrollers :: 09.05.2012 09:38 :: waqassalam :: Replies: **13** :: Views: **831**

can somebody tell me what should i do for **sampling** discrete time signals in **matlab**
i think u ask for the CODE...
in that case it is done like...
fs = 1000 (**sampling** **frequency**)
f = 10 (signal **frequency**)
A=2(amplitude)
n=0:100;
wave = A*sin(2*pi*f*n/fs)
this would create 100 sample points wave (...)

Mathematics and Physics :: 24.04.2004 15:03 :: cedance :: Replies: **4** :: Views: **6638**

i need a funtion for **matlab** to get the fourier cn coefficients
is there any link about this
is it fourier series coefficients or fourier transform coefficients needed for u? the fourier transform can be obtained with a simple code as follows.. all are inbuilt commands...
t = 0:1e-3:10; fs = (some value)-- **sampling** fr

Electronic Elementary Questions :: 13.11.2004 10:15 :: cedance :: Replies: **4** :: Views: **3651**

I think you'll get a fine resolution (1/M)*fs. Then the new index should be (f0/fs)*M.
Please verify it with a simple **matlab** code.

Digital Signal Processing :: 12.04.2005 11:19 :: joshuashi :: Replies: **1** :: Views: **3961**

resolution 10bit, how to generate this binary or decimal digital sine wave using **matlab** command? I know litter about **matlab**, thanks.

Analog IC Design and Layout :: 25.07.2005 13:00 :: showtime :: Replies: **4** :: Views: **18543**

Hi
I tested the codes in **matlab** 7 and it worked well.
here is the graph.
cheers

Digital Signal Processing :: 16.10.2005 06:25 :: mro83 :: Replies: **3** :: Views: **800**

?i want to analyse a low **frequency** ECG signal (<100Hz and taken directly from human body not filtered by analogue electronics and thus having noise and other **frequency** components but digitized by ADC) in **matlab** can i use serial port or parrell port directly? and how?
?if not then how can i do it ?

Digital Signal Processing :: 30.10.2005 06:13 :: adeel yasin :: Replies: **4** :: Views: **2312**

Hi guys I need to design a filter for a Phase lock loop, this is for FM system, if the carrier **frequency** is 20000 and **sampling** **frequency** is 44100. I beleive I need a high pass filter for PLL right? and if I do how do I set the parameters for the filter? THanks.

Digital Signal Processing :: 04.12.2005 12:36 :: locchamp :: Replies: **0** :: Views: **808**

Hi all,
I am a **matlab** newbie :)
How to get **frequency** spectrum of a vector data? (something like )
I heard FFT is the time-**frequency** fransform. And I tried fft() function in **matlab**.
But it return a complex number. Is it stand for both **frequency** and phase?
Any suggestions will be appreciated!
Best (...)

Electronic Elementary Questions :: 24.12.2005 01:38 :: davyzhu :: Replies: **38** :: Views: **159925**

I have understood.
you can use the hspice **matlab** tools.

Analog Circuit Design :: 04.01.2006 06:55 :: kidman561 :: Replies: **8** :: Views: **2513**

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 (plot) you a sinus wit

Digital Signal Processing :: 25.01.2006 04:47 :: emrek :: Replies: **3** :: Views: **919**

Hi all,
I'm still rather new to **matlab** and have certain questions to ask. Is it possible to generate a signal at a specific **frequency**? and is it possible to control the power spectral density of the signal?

Digital communication :: 12.03.2006 20:26 :: fabbie :: Replies: **3** :: Views: **2689**

how to write codes in **matlab** in order to make it read a wave file.
when i click on the read file button, the software i built must be able to extract the data in the file such the **sampling** **frequency** and the number of samples.
the software could only analyze a singel channel PCM audio with **sampling** (...)

Digital Signal Processing :: 23.03.2006 10:54 :: puther :: Replies: **1** :: Views: **1157**

Dear all,
This is a part of sigma-delta modulator code in **matlab**.
tr = 1024*4; % points for transient
pt = 1024*32; % FFT points
tt = tr+pt; % total points
osr = 64; % osr=64
fc = 20000; % signal bandwidth = 4k
fs = 2*fc*osr; % **sampling** **frequency**

Analog Circuit Design :: 08.04.2006 09:55 :: shaq :: Replies: **2** :: Views: **1134**

hi everybidy,
i have simulated a code in **matlab** for the demodulation of ofdm signal (dab). In the synchronisation process, i found the clock offset of 13.035 khz in the spectrum after downconversion. i used a nco to remove the **frequency** offset. Now when i differentially decode the symbols, i get 5th and 6th symbols corrupted. I think this can be h

Digital Signal Processing :: 12.05.2006 03:45 :: shameem :: Replies: **0** :: Views: **1387**

hi all
please can any one let clear my doubt "can the **sampling** **frequency** of a digitized signal be varied i.e can we vary **sampling** frq after **sampling** by adc if so please let me know how we do it "are there any methods for doing so???
yes u can do so,u can change the **sampling** rate ucsing (...)

Digital Signal Processing :: 13.07.2006 05:24 :: Aryanfar :: Replies: **3** :: Views: **582**

I am new to **matlab** and I was trying to code a **sampling** and aliasing example. But I have a few questions:
clc;
clear all;
%A is the amplitude
A=1;
%f is the **frequency**(Hz)
f=100;
T=1/f;
%fs is the **sampling** **frequency**
fs=1000;
%phi is the arbitraty phase,
phi = 2*pi*0.75;
tstart=0; (...)

Digital Signal Processing :: 19.10.2006 07:53 :: leony :: Replies: **1** :: Views: **796**

can u send me ur **matlab** code.
my email chaitubek@yahoo.co.in
thank u,.. plz.. its urgent

Digital Signal Processing :: 30.11.2006 05:12 :: chaitubek :: Replies: **3** :: Views: **1593**

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: **1534**

AoA to ALL!
I want to read a sound file in **matlab**, and set the **sampling** **frequency** to 8K. and represent each sample in 8-bits. play the result and also plot the **frequency** spectrum. i expect the spectrum should be from 0 to 4khz. plz tell me how to do all these in **matlab**.
simple & early reply is required.

Digital Signal Processing :: 25.01.2007 05:40 :: techspeaks :: Replies: **2** :: Views: **2233**

hi, I have to design some filters for processing speech in **matlab**, I know the basic commands but how do I adjust the cutoff frequencies in HZ , how to convert them into the filter parameters

Digital Signal Processing :: 19.02.2007 02:03 :: mehboob_iiui :: Replies: **7** :: Views: **1119**

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: **1819**

Hi,
I am trying to design a digital filter using the following parameters with use of **matlab**
1) **sampling** **frequency** 2)Order 3)Filter cutoff **frequency**
Is it possible design a filter these three parameters.
Please tell me how to design a filter with **matlab**.since i have two frequencies at the input (...)

Digital Signal Processing :: 24.02.2007 13:16 :: PraveeGoud :: Replies: **5** :: Views: **1133**

I do not seem to understand how **sampling** affects the **frequency** spectrum. Help me find the **frequency** spectrum before and after **sampling**. How to do it in **matlab**?
I am interested in frequencies from 0~integer***sampling** **frequency**. I want to see how different (...)

Digital Signal Processing :: 17.06.2007 09:08 :: iggyboy :: Replies: **1** :: Views: **1597**

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 **matlab** or tell me which macro in (...)

Electronic Elementary Questions :: 14.11.2007 12:44 :: hbaocr :: Replies: **4** :: Views: **1444**

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** **frequency** is 1/0.0001s=10KHz. I donot know what value I (...)

Digital Signal Processing :: 24.03.2008 02:09 :: triquent :: Replies: **4** :: Views: **6387**

Hi all, i have a problem using the pulse shaping commands in **matlab** i.e. rcosine( ),rcosfir( ),rcosflt( ).
I can't understand the arguments of these commands.
Problem:
We can transmit 'Rs=2W' symbols/sec (Rs is the symbol rate,W is the available bandwidth) with zero ISI if the available bandwidth is W Hz.
But this requires nyquis

Digital Signal Processing :: 25.03.2008 01:31 :: wajahat :: Replies: **0** :: Views: **2177**

anyone help me generate an IMPUSLE impulse train with **matlab**???
use the Difference Equation to generate a 80-120Hz impulse train with a 8k Hz **sampling** **frequency**: y=D+y p.s. any value for k is fine.
Thanks!

Digital Signal Processing :: 25.03.2008 09:44 :: obs6 :: Replies: **0** :: Views: **1185**

helllo,
i am receiving data from a 2-D digital accelerometer..i am provided with a software that reads the accelerometer signal form parallel port and samples it at the chosen **sampling** rate .....and writes the tab separated text file .....file contains three columns: time,x,y....
after taking this file,i want to apply low pass filter on thia da

Digital Signal Processing :: 12.04.2008 03:05 :: roop123 :: Replies: **3** :: Views: **900**

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 (...)

Analog Circuit Design :: 18.09.2008 11:24 :: lhlbluesky :: Replies: **4** :: Views: **850**

Thank you for your code but I an need of a **matlab** code which actually implements an FFT algorithm, rather than a built in library function. Please send me one if you have it.
Thank you
:)

Digital Signal Processing :: 23.09.2008 09:39 :: mahaju :: Replies: **6** :: Views: **50173**

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 fs=**sampling** **frequency**)
3)real

Digital Signal Processing :: 05.01.2009 09:27 :: lionelgreenstreet :: Replies: **0** :: Views: **1501**

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 command, or i need to create a time vector.

Digital Signal Processing :: 17.01.2009 17:13 :: eng_shady00 :: Replies: **9** :: Views: **2705**

hi every one,
i would like to share some of the uses of fft for spectrum analysis.
Hope it will be useful for those who are novice to **matlab** programming.
DFT Notes:
DFT produces a discrete **frequency** domain representation.Also, DFT is only defined
in the region between 0 and Fs.
(as we know, One period extends from f = 0 to Fs, where Fs is

Digital Signal Processing :: 03.03.2009 03:48 :: rramya :: Replies: **12** :: Views: **17938**

Aliasing Example
The program below demonstrates this problem graphically. We simulate **sampling**
two sinusoids, 2 cos(2Π100t +Π/3) and 2 cos(2Π600t +Π/3), at 500 samples/second.
They are given the same magnitude and phase for a reason: the program shows that the sampled versions are identical.
%
% Ex

Digital Signal Processing :: 17.03.2009 18:12 :: Aya2002 :: Replies: **4** :: Views: **2508**

Hi iam wrinting a simple **matlab** code that generates a sinusoidal signal which is then sampled ploted. when I take the **sampling** **frequency** fs=5000 and f0=4500 hz. The number of samples n=100,.
I should get an aliaisnig error right because fs<2*f0.... The problem is that i get all the signal undistorted!!! This is the **matlab** (...)

Digital Signal Processing :: 24.03.2009 11:46 :: Ibaghdadi :: Replies: **3** :: Views: **903**

Hi,all
I am in a project of Sigma-Delta ADC. But I get in trouble when I design the digital filter in the **matlab**. The filter consists of a CIC, a CICCOMP and two halfband. When CIC is followed by the CICCOMP and then the twe halfbands, the CICCOMP can compensate the droop caused by the CIC in the passband well. But when I exchange the order betwee

Digital Signal Processing :: 12.05.2009 11:29 :: wqy1985 :: Replies: **6** :: Views: **1370**

Hi everyone
I am a serious newbie in this modulation techniques :( I have been reading up and still do not understand.
if i need to design a QASK modulation scheme for **matlab** of the following specifications, how do i do this?
Bandwidth: 200 kHz
Centre **frequency**: 1.420 MHz
Signal Power available: 1 mW
Noise Power Spectral density: 1.0 10

Digital communication :: 16.05.2009 23:40 :: yoshi4u :: Replies: **0** :: Views: **1117**

My IF is 5Mhz.
I need to reduce my **sampling** **frequency** 409.8Mhz to 20.48 Mhz by using FIR lowpass filter, so i am decimating it in 2 stages , first time i am using cic filter and decimation factor 5 then i will get the new **sampling** **frequency** 81.9 which is given to half band filter by decimating factor 4 then we (...)

Digital Signal Processing :: 21.06.2009 05:33 :: guest_1044 :: Replies: **0** :: Views: **1708**

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 so i have to reduce to 20.49Mhz. (...)

Digital Signal Processing :: 22.06.2009 07:59 :: guest_1044 :: Replies: **1** :: Views: **949**

I want to read the signal frequencies that can be seen in the spectrogram using **matlab** function(spectrogram).
But when i give different **sampling** frequencies i am getting the display of my signal at different **frequency** points.This is explained below.
I am using the below code to create a sine wave with freq = 100hz. and display its (...)

Electronic Elementary Questions :: 13.08.2009 01:22 :: dil639 :: Replies: **0** :: Views: **541**

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

Digital communication :: 06.09.2009 09:47 :: yiyi87 :: Replies: **0** :: Views: **3444**

I designed a filter hilbert with **matlab** (function fdesign.hilbert and design(,'firls'). I tested it with a cosine but the results are very stranges: for my **sampling** **frequency** 100 Mhz if I try with a cosine of an interger **frequency** (1,2,3 .. 49) Mhz the result is ok, hilbert filter returns the 90 degree shift signal of my (...)

Digital Signal Processing :: 15.09.2009 04:37 :: telosa :: Replies: **1** :: Views: **2930**

Help please!!!!
i have sinwave with 0.1 v amplitude and 1khz **frequency** mixed with 5v white noise.
i used FFT in **matlab** to simulate.
1-(Study the use of additional mean value calculation of every data set of FFT spectrum and how it affects the visibility of a present signal.). so, how can i use additional mean value???????????
2-W

Digital Signal Processing :: 27.09.2009 08:19 :: program2 :: Replies: **0** :: Views: **1460**

Hi, all
could anyone provide me the **matlab** file for FSK modulation and demodulation, or any links is also welcome, I would appreciate it.
Thanks in advance.
Best regards

Digital communication :: 12.11.2009 00:56 :: abcyin :: Replies: **1** :: Views: **6280**

So, you have 8ms of data and you are **sampling** at 16kHz (128 points captured in 8ms).
The FFT resolution is then 125Hz and the highest **frequency** is 8kHz (16kHz/2, Nyquist theorem).
In **matlab** language: f=0:125:8000; is your **frequency** axis. Keep in mind that **matlab** will show both sides of your FFT, so you (...)

Digital Signal Processing :: 02.12.2009 22:51 :: JoannesPaulus :: Replies: **4** :: Views: **2527**

Hi friends,
I am working to realize HSP50110 / 50210 architecture, used for demodulation in **matlab** and FPGA. I started with **matlab**'s inbuilt example "symbol timing recovery with fixed **sampling**" and extended the design for HSP50110. I tried to replace Squaring loop method with either Gardner or Early-late Gate algorithm, but no success. (...)

Digital communication :: 02.12.2009 02:24 :: mpatel :: Replies: **0** :: Views: **558**

Perhaps you could answer a few questions...
> I am generating UWB signal of 2GHz with **matlab** **sampling** **frequency**(for >resolution) of 8GHz.Thus, each UWB pulse has 4 sample values.
>Now the number of samples in the received signal are 800.
>I do compressed sensing with Phi=randn(400,800)
So is it correct that you generate samples (...)

Digital communication :: 26.12.2009 04:19 :: bulx :: Replies: **2** :: Views: **1213**

Digital Signal Processing :: 02.01.2010 14:18 :: JoannesPaulus :: Replies: **18** :: Views: **11730**

Last searching phrases:

bjt and area | ferrite transformer winding | sram compiler | gsm with pic microcontroller | ie3d impedance | pid pic | plot farfield | electron mobility | telecom projects | export from matlab

bjt and area | ferrite transformer winding | sram compiler | gsm with pic microcontroller | ie3d impedance | pid pic | plot farfield | electron mobility | telecom projects | export from matlab