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## Matlab Sampling Frequency |

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

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

hi i am using a filter of passband **frequency** of order 128 hamming filter
i am giving sine signal of **frequency** 6500 and the sine signal is not perfect because it is higher **frequency**. i am getting glitches and when test the **frequency** in fft i am getting glitches.
please help me in testing that filter

Digital Signal Processing :: 10-04-2015 09:30 :: lakshmikalyani :: Replies: **10** :: Views: **431**

Hi everyone
I have a signal and I am using **matlab** command pwelch to calculate the **frequency** of the signal, but the **frequency** I obtained is changed as I changed the **sampling** **frequency**.
pwelch(x,window,noverlap,f,fs)
For example, when using **sampling** **frequency** equal (...)

Digital Signal Processing :: 08-01-2015 19:50 :: Serwan Bamerni :: Replies: **5** :: Views: **564**

hi
my specifications are passband frequencies are 200 and 315 for a band pass filter
i have to design a filter bank with a range of frequencies. this is first **frequency** band
i have to design a fir filter bank form 200 to 7500 hz.
my **sampling** **frequency** should be more than 15000 i took it as 16000. the stop band (...)

Digital Signal Processing :: 02-28-2015 02:59 :: lakshmikalyani :: Replies: **1** :: Views: **372**

I try to get a first order notch filter at 50hz and **sampling** **frequency**(Fs) 200 Hz in **matlab** I wrote some code but didn't work
dt=1/200; //1/fs
t=0:dt:0.2;
x=cos(2*pi*25*t)+cos(2*pi*50*t);
y=filter(x,------); //I dont know what I must write here
plot(Y(w)); //I must write here laplace of y(w)
How can do that

Digital Signal Processing :: 12-25-2014 07:59 :: Eowe :: Replies: **1** :: Views: **734**

I have simple smooth time signal with different time periods. I would like to transfer period over the time to **frequency**, and to present **frequency** changing over the time, I would call it Instantaneous **frequency**?
Here is **matlab** Code, is it right?
x = data;
z = hilbert(x);
N = length(z);
fs= **sampling** (...)

Digital Signal Processing :: 12-23-2014 13:11 :: ToShare :: Replies: **0** :: Views: **384**

PLD, SPLD, GAL, CPLD, FPGA Design :: 11-24-2014 17:54 :: FvM :: Replies: **1** :: Views: **407**

Hi,
I have this filter:
d = fdesign.pandpass('n,F3dB',3,50e3, 200e6);
Hd = design(d,'butter');
which I want to plot at a central **frequency** of 868.3 MHz.
Could you pleaase help me with this?
Thank you in advance!

Digital Signal Processing :: 07-22-2014 04:34 :: Saly07 :: Replies: **1** :: Views: **440**

Hi all,
I want to design filter to implement on FPGA with a mentioned parameters in subject but when i am starting to design it on **matlab** fdatool, it gives very high order filter result, and when i decrease or set by myself upto 10 order its impulse response is not good as i want.
Any idea will be highly appreciable.
Regards,
Umair

Digital Signal Processing :: 05-18-2014 08:18 :: umaair_653 :: Replies: **2** :: Views: **387**

Hi there!
I have an assignment that goes as follows:
Pulse compression by using a Linear FM signal
1) Consider a (real) Linear FM signal with a center **frequency**, f0= 1250 MHz, a bandwidth, B = 100 MHz, and a length, T = 0.15 ?s.
- What is the required **sampling** **frequency**, fs?
- Derive an

Digital communication :: 03-26-2014 21:28 :: kviksand81 :: Replies: **2** :: Views: **1164**

Hi,
I understand u have data with 100 samples per second ( assuming dt = 0.01s is the **sampling** interval)
this will give your nyquist **frequency** as fs/2 = 50 Hz, this is important as the cut off **frequency** of your filter is normalized with respect to the nyquist **frequency**.
fc is your cutoff frquency
you need to ge

Digital Signal Processing :: 03-18-2014 04:45 :: deboleena18 :: Replies: **3** :: Views: **471**

Do you use **matlab**? If so, consider an approach like this:
close all; clear all; clc;
% Define **sampling** times
dt = 0.001;
t = (0:dt:5).';
% Sine **frequency**
f = 3;
% Amplitudes
A1 = 2.6;
A2 = 0.7;
% Phases
phi1 = 0.4;
phi2 = -0.3;
% Sinusoids
x1 = A1*cos(2*pi*f*t + phi1);
x2 = A2*cos(2*pi*f*t + phi2);
% Estimated am

Digital Signal Processing :: 03-09-2014 11:14 :: weetabixharry :: Replies: **2** :: Views: **457**

Hello All,
The code for ASK-4 is as follows:
N = 8; % The number of bits to send - Frame Length
bit_stream = round(rand(1,N)) % Random bit stream
A1 = 3; % Amplitude for 0 bit
A2 = 5; % Amplitude for 1 bit
f = 3; % **frequency** of Modulating Signal
fs = 100; % **sampling** rate
t = 0:1/fs:1; % Time for 1 bit
% This time variable is just for p

Digital communication :: 08-26-2013 05:00 :: Shruti01 :: Replies: **2** :: Views: **660**

Hello!!!
I am not able to generate Sine wave of **frequency** 200Hz when configured PWM **frequency** of PIC16F877A to 10KHz and PIC is operating at 20MhZ Crystal.
This is **matlab** Code to generate table.
f**sampling** = 10000; %10Khz **sampling** Rate
F = 200; %200hz **frequency** (...)

Microcontrollers :: 08-24-2013 02:31 :: xpress_embedo :: Replies: **11** :: Views: **1764**

Hi, I have post this question to **matlab** but no one answer :(. Hope someone help me in here.
I have tried to plot the chirp signal from 20MHz to 70MHz in time domain and **frequency** domain. The target time and sweep time is set to 3 ms. Normally I see slew rate =1s, dont know why?. **sampling** time = 140 us and samples per frame = 4096. The (...)

Digital Signal Processing :: 07-13-2013 07:20 :: suribright :: Replies: **2** :: Views: **559**

hi guys,
when i generate a random data in **matlab** and i want to apply a lowpass filter to these data , how can i identify it's maximum **frequency** in order to determine the **sampling** **frequency** that i will use as an input to my filter
for example :
x=randint(1,10); %%% random bits
a=modem.pskmod(4); %%%% QPSK object (...)

Digital communication :: 06-27-2013 13:07 :: bigstrik :: Replies: **0** :: Views: **306**

I should design a digital filter of first order ADC Sigma-Delta converter with 8 bits, an over **sampling** ratio of 64 (OSR = 64),and **sampling** **frequency** of 10.24MHz.I make it with **matlab** SIMULINK as shown in this

PLD, SPLD, GAL, CPLD, FPGA Design :: 06-27-2013 12:43 :: fasto2008 :: Replies: **6** :: Views: **1314**

Hi there,
I have a problem with the phase of my differentiator.
I designed it with two methods (**frequency** **sampling** and Remez algorithm) in **matlab** (fir2 and firpm)
when I use a filter types with odd-order everything is right.
If I take one with even order I had a phase-shift between the input and the output
which dependet on (...)

Digital Signal Processing :: 06-20-2013 07:22 :: Stefan87 :: Replies: **0** :: Views: **324**

dear;
is it a course work?
first try ur self; if u stuck post the code.
see dft help in **matlab** details.
Savor,Best

Mathematics and Physics :: 05-20-2013 02:45 :: dashkil :: Replies: **1** :: Views: **883**

i hv written code for sigma delta modulator of DAC in **matlab**, i m taking input **frequency** 20khz and nyqiust **frequency** 48khz,over **sampling** ration 128.i got spectrum of modulator using hanning window with **frequency** 3000(normalised **frequency**) but really i confused how to measure SNR of this (...)

Analog Circuit Design :: 03-06-2013 04:09 :: nishagajare :: Replies: **0** :: Views: **463**

hi,
can u please tell me how I can calculate the fir filter coefficient calculation in **matlab** using **frequency** **sampling** method and window method for 16 taps.

Digital Signal Processing :: 01-07-2013 08:31 :: sauj90 :: Replies: **0** :: Views: **857**

well in this I disagree, since increasing the **sampling** speed ofcourse helps, e.g. (and can test on **matlab** simply) if 1 Khz signal is sampled using 3 Khz the result is actually almost no use. while increasing the **sampling** rate will increase the quality of signal.
Simply put the more the **sampling** rate, the less is the (...)

Digital Signal Processing :: 12-23-2012 11:59 :: syedshan :: Replies: **30** :: Views: **1465**

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 :: 11-25-2012 07:19 :: electronicstudent :: Replies: **3** :: Views: **1016**

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 :: 10-25-2012 11:44 :: kanni1303 :: Replies: **2** :: Views: **489**

Hi everyone, I hope to be helped as soon as possible...
I'd like to sample a signal in **matlab** which i dont know the time domain formula... i know the **frequency** domain one and i used ifft to obtain the time domain and i am chasing for a command which gives me sampled signal in my desired **sampling** **frequency** and i am sure this (...)

Digital Signal Processing :: 10-05-2012 21:45 :: y.ettefagh :: Replies: **2** :: Views: **635**

Dont implement passband simulation of such a high **frequency**.
I think it is a solver problem (menu Simulation - Configuration parameters). The internal default **matlab** **sampling** **frequency**.
I placed in an empty model sine wave generator and scope. Set your **frequency** and set the configuration parameter for (...)

Digital communication :: 09-27-2012 05:15 :: Mityan :: Replies: **3** :: Views: **901**

Hello friends,
I am facing problem in implementing IIR filter. I have determined the coefficient for the first order Butterworth filter using **matlab** FDATOOL with **sampling** **frequency** 2000 Hz and cutoff **frequency** 100 Hz and used it to filter the AC signal but it isn't working.
To implement this I took the adc sample in (...)

Digital Signal Processing :: 09-07-2012 01:25 :: mshrestha789 :: Replies: **3** :: Views: **1130**

When you supply the scalar **sampling** **frequency** fs as an input argument to freqz, the **frequency** ranges from 0 to fs/2 Hz. - This is from **matlab** help on freqz function. Taking fs you may estimate 700 hz to 1600 hz by yourself.
Also you may make a FFT of filter impulse response (this will be your spectrum) and do with it what (...)

Digital Signal Processing :: 08-06-2012 02:00 :: Mityan :: Replies: **1** :: Views: **755**

Pass band 4-6 kHz
Transition width 0.5kHz
Pass band ripple 1dB
Stop band attenuation 50dB
**sampling** **frequency** 20kHZ
FFT frame length 512
a) Apply the optimal method to design and implement
(i) the pass band and stop band ripples
(ii)The filter order and filter length N
(iii) Filter realization structure
b)write **matlab** program

Digital Signal Processing :: 04-22-2012 06:50 :: asankaou :: Replies: **3** :: Views: **910**

Hi,
I need to apply the Doppler effect to a music file, I invoke the music file by using the waveread command,
How do I proceed next (**frequency** spectrum, apparent **frequency** etc.)?

Digital Signal Processing :: 04-10-2012 02:04 :: Rahp :: Replies: **2** :: Views: **2647**

Hi,
I would like to run the QPSK at 13.56 MHz of carrier **frequency**. Any one have a **matlab** code please send me one. I have tried to write this code but did not work at 13.56 MHz. Wont know what wrong is my code?. In my code, I write the carrier at 13.56Mhz and **sampling** **frequency** at 120KHz. When I take FFT of signal it (...)

Digital communication :: 04-08-2012 05:40 :: suribright :: Replies: **13** :: Views: **1135**

I have designed the SAR ADC of 8 bit with **sampling** **frequency** of 1MHz in cadence.For calculating the dynamic performance , i have used code provided by Maxim . for exporting data , i have used the tool called TABLE in cadence and saved as . csv file and for processing , i deleted the time and heading of Table and saved as txt file and fed to the mat

Analog Integrated Circuit (IC) Design, Layout and Fabrication :: 04-02-2012 07:29 :: Bishwo :: Replies: **0** :: Views: **569**

Use the **matlab** FFT command
Scale / calibrate the **frequency** data with the **sampling** rate data and number of time domain samples
Scale / calibrate the amplitude data with the number of time domain samples

Elementary Electronic Questions :: 03-06-2012 03:56 :: klystron :: Replies: **2** :: Views: **3286**

hello. i am having some problem on demodulating my signal maybe some one can help me
below is my code
%% create DATA
fc=13.56E6; % carrier **frequency**
fm=fc/32; % modulation **frequency**
fs=fc*4; %**sampling** **frequency**
tb=512/fc; %time length of 1 bit
To=8*tb; %length of 8 bit (...)

Digital Signal Processing :: 12-13-2011 06:45 :: nearxos :: Replies: **0** :: Views: **430**

Heya Scott - unevenly sampled data is a turd!
Two options spring to mind:
1. Resample the data onto a 1ms grid and go from there, or
2. Use spectral estimation techniques that don't rely on regular **sampling**, such as the Lomb-Scargle periodogram described in Numerical Recipes ( ). I see someone has kindly written a function

Digital Signal Processing :: 12-08-2011 00:46 :: thylacine1975 :: Replies: **2** :: Views: **883**

Hello
I need to design FIR filter with cutoff **frequency** 100 Hz (with attenuation 3 dB at cutoff **frequency**), **sampling** **frequency** = 1000Hz and having 3rd order Butterworth characteristics, using the various types of window functions
Can anyone give me some ideas on how to do this?
Also, could you please tell me how to do (...)

Digital Signal Processing :: 06-14-2011 21:55 :: mahaju :: Replies: **1** :: Views: **1792**

Hi everyone. This is my first thread and I really need help. I have to create 16QAM constellation diagram in **matlab**. Here are my data: **frequency** of carrier is 8000khz and **sampling** **frequency** is 32000khz. Amplitude is 5000(dont know if you need this). Here is my code in CCS (if is any help):
#include "tonecfg.h"
#include (...)

Digital communication :: 11-11-2011 07:47 :: Keklja :: Replies: **0** :: Views: **590**

Hi All,
I have done conservative transient simulation in Cadence after doing ADC. I am trying to post process the output in **matlab** using FFT. I have a 10ns period (**sampling**) but in between periods I have almost 400 points of data in between. This makes it hard to calculate the down**sampling** **frequency** since the number can (...)

Analog Integrated Circuit (IC) Design, Layout and Fabrication :: 10-16-2011 21:41 :: acbalbason :: Replies: **0** :: Views: **602**

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

Digital Signal Processing :: 10-11-2011 22:43 :: vickyuet :: Replies: **1** :: Views: **2768**

Hello Everyone
I have a speech signal that I have read using wavread, whose details are:
Sound is a vewel 'a'
**sampling** **frequency** = Fs = 44.1Khz
Sample values = data
I need to estimate the first three formants f1,f2,f3 of this vowel.
Can any body provide me some **matlab** code or routine that could estimate the formants for this (...)

Digital Signal Processing :: 09-18-2011 16:01 :: hassaanliu :: Replies: **0** :: Views: **679**

Hi bjmoloi,
If you want a Graphical User Interface, use fdatool (Filter Design & Analysis Tool).
If you need a function, use firpm (or remez in old **matlab** versions).
Regards
Z

Digital Signal Processing :: 08-11-2011 17:51 :: zorro :: Replies: **2** :: Views: **540**

Hi,
I am simulating vibrations in simulink and processing my results in **matlab**. I am trying plot the PSD for my acceleration results using pwlech method.
When I plot the PSD plots from the methods below
(NOTE:I have a **sampling** rate of 0.006 so that means I am using a sample **frequency** of Fs=1/0.006=166)
Pwelch(x,,,2024,fs)
Or =pwel

Digital Signal Processing :: 05-03-2011 20:18 :: 19dan87 :: Replies: **1** :: Views: **614**

thanks andre.
but is there some problem in my code.
my book question says
DFT
(a) generate a sinusoid 05 .5 Khz with **sampling** freq 6 Khz
(b) plot the original signal using stem
I think first two parts are done. the third part says abt complex exponentials
can u also help me to write **matlab** code to compute its complex exponentials.

Digital Signal Processing :: 04-24-2011 13:35 :: moonnightingale :: Replies: **6** :: Views: **9601**

when i want to calculate the SNR of output of sigma delta i run the design on cadence for 6 sec(as i have input=100hz &**sampling** **frequency** =50Khz)
and i take it on **matlab** to calcsnr when i take samples very close from end the snr become 97db but when i go to the middle it become 60db so how many samples i take from the file and from where

ASIC Design Methodologies and Tools (Digital) :: 04-21-2011 18:22 :: hoka_89 :: Replies: **0** :: Views: **786**

As far as I understood your question you can simply group all the signals in a single vector. The command is:
ytot = ;
Please check the synopsis in **matlab**, since I'm actually using Scilab (however I think shouldn't be different)

Digital Signal Processing :: 04-19-2011 17:27 :: albbg :: Replies: **11** :: Views: **3629**

Hello all,
Can anyone please tell how generate random data in **matlab** with a specific data rate..!!
or a binary data stream with say **sampling** **frequency** of 20MHz..???
:?:

Digital communication :: 02-12-2011 03:16 :: commworld :: Replies: **0** :: Views: **490**

Hi,
I have a project called "pitch detection using LPC parameters with **matlab** code".The input voice is lowpass filtered a cut off **frequency** of about 900Hz and then the **sampling** rate (nominally 10kHz) is reduced to 2kHz by a decimation process.The decimated output is than analyzed using the autocorrelation method with a value of p=4 for the (...)

Digital Signal Processing :: 12-24-2010 17:01 :: magnitudee :: Replies: **0** :: Views: **2963**

can anyone help me writing source code in **matlab** for FIR filter design using Genetic Algorithm
re

Heuristic methods, Machine Learning, AI, and Soft Computing :: 12-18-2010 08:56 :: gagneja :: Replies: **2** :: Views: **1475**

Hi,
I have a signal witch spectrum (calculated with **matlab** fft) is 70 Hz wide.
But I really need only the first 5 Hz. Is there any way to "concentrate" the fft only on that part of the spectrum? In this way (maybe) I can use the 1024 fft points only in the first 5 Hz, instead of 70 Hz, increasing a lot the resolution..
Thanks,
bye,
Vale

Digital Signal Processing :: 12-04-2010 06:02 :: vaka85 :: Replies: **5** :: Views: **2022**

Dear define Order of filter.
I think **matlab** is a good teacher also use **matlab** help.
cutt of **frequency** is defined as
(cutt off freq/**sampling** rate)
try this in your function defination

Digital Signal Processing :: 11-29-2010 04:29 :: shahbaz.ele :: Replies: **4** :: Views: **3598**

I am trying to replot FFT of following signal in **matlab**
But i am not getting exact result
Signal having **sampling** **frequency** is 1024 Hz and the **sampling** time is 1s
and for more description please find attached file
Its very urgent PLEASE help me
thank u in advance

Software Problems, Hints and Reviews :: 11-16-2010 04:32 :: pravinkadam2 :: Replies: **2** :: Views: **1027**

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