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## Matlab Sampling Frequency |

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

I am trying to understand how the spectrum analyser in SIMULINK (**matlab**) works, by simulating a model which inputs a simple cosine wave into the spectrum analyser and then observing the power spectrum on the analyser. I have set the WINDOW function in the anayser to RECTANGULAR and the **sampling** frquency to 20 times the **frequency** of the (...)

Digital Signal Processing :: 03-21-2017 20:14 :: derick007 :: Replies: **0** :: Views: **515**

Hello everybody
I need to design a Bandstop filter with **matlab** , but I do not what to do :sad: .
Can anyone help me please?
Thanks

Digital Signal Processing :: 03-16-2017 05:35 :: Majid_Vatan_Parast :: Replies: **2** :: Views: **539**

please read the following code in **matlab** for chopper amplifier
Fs = 16*131072 ; % Modulator **sampling** **frequency** Fs = 2^17Hz
how did they get these values in the code when following values are applicable to the op-amp according to the design?
modulator **frequency** =16khz
amplifier bandwidth=32khz
input signal (...)

Analog Integrated Circuit (IC) Design, Layout and Fabrication :: 11-23-2016 08:15 :: annadarwin :: Replies: **0** :: Views: **3**

Hello!! Everyone
I am using Saleae Logic Analyzer to capture signals from the PS2 Keypad, and it works perfectly.
But i want to save these signals for future analysis, so i saw export option in which we can export data in **matlab**.
Channel-0 is connected to Clock and Channel-1 is connected to Data Pin of PS2 Keyboard.
**sampling** **frequency** is (...)

Digital Signal Processing :: 06-01-2016 17:01 :: xpress_embedo :: Replies: **0** :: Views: **529**

This is the file in touchstone(S1P) format . It is in **frequency**
domain .we need to convert it to time domain.we got it from Vector
Network analyzer.And **frequency** rang is from 3 to 10 GHz. Total number of
sample is 201.the S11 is in dB.the file contain three column first column
contain **frequency** second column dBS11 and third show (...)

Software Problems, Hints and Reviews :: 04-13-2016 06:18 :: sajjadwazir :: Replies: **3** :: Views: **1280**

Hello everyone.
I am trying to plot **frequency** of a sine wave Vs time in simulink.
I have changed to **frequency** from time to time to make sure that my blocks are able to detect **frequency** change. However, i get a delay response by half fundamental cycle when the **frequency** changes from magnitude to another..
My question is : (...)

Digital Signal Processing :: 03-10-2016 20:04 :: Eng.Salwa1 :: Replies: **3** :: Views: **641**

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 13:30 :: lakshmikalyani :: Replies: **10** :: Views: **1111**

Without knowing details of **matlab** command, different **frequency** results for different Fs would happen if your **sampling** freq was a sub-**frequency** of the sampled one. I.e. when you do sub-**sampling** and the aliased component only is output.
Is your example case a real case or hypothetical ? I cannot work out (...)

Digital Signal Processing :: 08-02-2015 09:21 :: kripacharya :: Replies: **5** :: Views: **832**

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 07:59 :: lakshmikalyani :: Replies: **1** :: Views: **573**

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 12:59 :: Eowe :: Replies: **1** :: Views: **4359**

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 18:11 :: ToShare :: Replies: **0** :: Views: **605**

PLD, SPLD, GAL, CPLD, FPGA Design :: 11-24-2014 22:54 :: FvM :: Replies: **1** :: Views: **701**

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 08:34 :: Saly07 :: Replies: **1** :: Views: **634**

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 12:18 :: umaair_653 :: Replies: **2** :: Views: **592**

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 :: 03-27-2014 01:28 :: kviksand81 :: Replies: **2** :: Views: **2817**

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 08:45 :: deboleena18 :: Replies: **3** :: Views: **664**

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 15:14 :: weetabixharry :: Replies: **2** :: Views: **586**

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 09:00 :: Shruti01 :: Replies: **2** :: Views: **1098**

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 06:31 :: xpress_embedo :: Replies: **11** :: Views: **2476**

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 11:20 :: suribright :: Replies: **2** :: Views: **833**

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 17:07 :: bigstrik :: Replies: **0** :: Views: **466**

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 16:43 :: fasto2008 :: Replies: **6** :: Views: **2340**

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 11:22 :: Stefan87 :: Replies: **0** :: Views: **417**

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 06:45 :: dashkil :: Replies: **1** :: Views: **1100**

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 09:09 :: nishagajare :: Replies: **0** :: Views: **722**

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 13:31 :: sauj90 :: Replies: **0** :: Views: **1165**

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 18:21 :: FvM :: Replies: **30** :: Views: **2260**

Hi there, ty for your reply. I have to be able to view that sine wave in the **frequency** domain and use stem functions to do this. I then have to do the same for the **sampling** **frequency**. I then have to convolve these two stem functions together for a result. I understand these from a theory point of view but am una

Digital Signal Processing :: 11-27-2012 15:57 :: xpress_embedo :: Replies: **3** :: Views: **1314**

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 15:44 :: kanni1303 :: Replies: **2** :: Views: **647**

Hi,
The description of your need is not completely clear to me, but what i see is that you have a signal with some **sampling** **frequency** and you need the samples at a different **sampling** **frequency**. In this case, the resample function will work.
Look at doc resample or help resample and **matlab** will suggest you (...)

Digital Signal Processing :: 10-06-2012 11:39 :: zorro :: Replies: **2** :: Views: **902**

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 09:15 :: Mityan :: Replies: **3** :: Views: **1388**

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 05:25 :: mshrestha789 :: Replies: **3** :: Views: **1797**

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 06:00 :: Mityan :: Replies: **1** :: Views: **1021**

it's all ready in signal processing toolbox of **matlab**, what you do is just input the parameters.

Digital Signal Processing :: 04-30-2012 13:13 :: opt :: Replies: **3** :: Views: **1062**

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 06:04 :: Rahp :: Replies: **2** :: Views: **4649**

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 09:40 :: suribright :: Replies: **13** :: Views: **1444**

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 11:29 :: Bishwo :: Replies: **0** :: Views: **730**

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 08:56 :: klystron :: Replies: **2** :: Views: **4927**

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 11:45 :: nearxos :: Replies: **0** :: Views: **676**

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 05:46 :: thylacine1975 :: Replies: **2** :: Views: **1110**

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 :: 06-15-2011 01:55 :: mahaju :: Replies: **1** :: Views: **2202**

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 12:47 :: Keklja :: Replies: **0** :: Views: **871**

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 :: 10-17-2011 01:41 :: acbalbason :: Replies: **0** :: Views: **882**

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 :: 10-12-2011 02:43 :: vickyuet :: Replies: **1** :: Views: **4241**

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 20:01 :: hassaanliu :: Replies: **0** :: Views: **920**

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 21:51 :: zorro :: Replies: **2** :: Views: **866**

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 :: 05-04-2011 00:18 :: 19dan87 :: Replies: **1** :: Views: **819**

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 17:35 :: moonnightingale :: Replies: **6** :: Views: **12230**

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 22:22 :: hoka_89 :: Replies: **0** :: Views: **969**

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 21:27 :: albbg :: Replies: **11** :: Views: **4409**

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