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## Low Pass Filter Code In Matlab |

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45 Threads found on edaboard.com: **Low Pass Filter Code In Matlab**

hi, how to write **matlab** **code** for RC 1st order **low** **pass** **filter**??
i saw many **matlab** **code**s from internet , like butterworth, chebychev...
but don't have the one for RC **low** **pass** **filter**??
pls advice how to do it? i was (...)

Digital communication :: 24.12.2006 18:37 :: cmppolish2006 :: Replies: **0** :: Views: **3383**

how to design -a LPF(fir) by window method sampling freq=100hz and cut off=20khz..without using FFT builtin function.
plz can u give me a **code**(program) to implement this using **matlab** softwear.please.

Digital Signal Processing :: 01.02.2007 23:38 :: bharathchandra :: Replies: **2** :: Views: **4580**

required **matlab** **code** for **low** **pass** **filter**ing of a digital sequence
for e.g
i want how to found the value of this every elemnt in it after **pass**ing through **low** **pass** **filter** i requesting **matlab** **code** for it

Digital Signal Processing :: 24.03.2009 15:22 :: sayurabh :: Replies: **1** :: Views: **2463**

hi.
I have written a **code** which emulates a qpsk tx n rx system. it works perfectly and i'm getting the BER curve exactly.
As my next task i have been asked to design an fir **low** **pass** **filter** through which i am to **pass** the qpsk signal. i'm not getting much assistance from the guide and i'm pretty much on (...)

Digital Signal Processing :: 29.01.2010 06:00 :: shubhs9 :: Replies: **5** :: Views: **4753**

Hello,
I have a signal vector and I wanted to **low** **pass** **filter** it. My knowledge of discrete **filter**s is very limited so any help would be greatly appreciated.
My signal is basically a sampled instance of a sum of sinusoids of 100Hz and 1KHz. What I want to do is basically **filter** this sampled (...)

Digital Signal Processing :: 02.03.2010 14:28 :: aryajur :: Replies: **1** :: Views: **7808**

Use the fol**low**ing **code** for creating a **low** **pass** **filter** :
Determine the cut off frequency and change Fc appropriately
Fc = 500 ; %cut off frequency
w = Fc/(2*fs); % Normalized frequency
%design a **low** **pass** fitler with above mentioned specifications (...)

Digital Signal Processing :: 25.05.2010 06:03 :: mathuranathan :: Replies: **6** :: Views: **16770**

phase is defined as -arctan(w/wc), where w is frequency and wc is cut-off frequency. Whatever we have instead of w/wc, phase should not have the value **low**er than -90 deg.
Would be true only for a first order **low**-**pass**.

Analog Circuit Design :: 25.09.2012 05:59 :: FvM :: Replies: **4** :: Views: **577**

I need a **matlab** **code** to
promt the user to enter a sinusoid signal
ex: x= 3*sin(2*pi*50*t) + 3*sin(2*pi*100*t) + 3*sin(2*pi*200*t) + 3*sin(2*pi*400*t)
and if any other variables are needed from the user.
then display the signal in time domain and frequency domain.
and then apply a **low** **pass** **filter**,
to (...)

Digital Signal Processing :: 18.04.2013 10:00 :: Prosenjit101 :: Replies: **7** :: Views: **820**

I have one problem.
I have a data vector. for example : data=rand(1,n) ; n=100;
I want to obtain discrete fourier transform of it by DCT command.
and then,by a **low**-**pass** **filter**,remove unwanted high frequencies of larger than f1.
f1 is determined.
and finally , again convert **filter**ed data from discrete frequency domain to (...)

Mathematics and Physics :: 22.04.2013 12:01 :: ghasem_008 :: Replies: **0** :: Views: **630**

Hi
How to use **matlab** write a **low**_**pass** **filter** .. I can use simulink for
simulation , but I want to know how to write a **matlab** **code**..
simulink file can novert to **matlab** **code** or not ??
a **low** **pass** (...)

PCB Routing Schematic Layout software and Simulation :: 16.12.2003 05:39 :: andy2000a :: Replies: **8** :: Views: **2514**

I just designed an FIR **filter** in **matlab**, and saw the impulse response. I obtained a **low** **pass** **filter**. Up to now there were no problems. I read the cut-off frequency(0.144cycles/sample) and had 2 null frequencies at 0.325cycles/sample and 0.5cycles/sample.
Then I inputted a discrete sine wave block(with (...)

Digital Signal Processing :: 29.11.2006 18:34 :: dav_mt :: Replies: **3** :: Views: **1600**

hello,
can anyone provide me with the **code** or implementation ideas for **low** **pass** **filter** of particular cut-off frequency and notch **filter** without using inbuilt functions on **matlab** ...
please
thanks in advance ...

Digital communication :: 26.03.2007 05:20 :: rmreddy :: Replies: **2** :: Views: **1907**

Hello,
could you guys please give me an explanation why if I **low** **pass** **filter** an upsampled (zero stuffed) ramp signal I get some oscillation which increases in amplitude with time?
The time frame that I am looking is very big in comparison to the response length of the **filter**, in other words, I am NOT seeing just the (...)

Digital Signal Processing :: 20.11.2007 11:50 :: Arturi :: Replies: **0** :: Views: **882**

here the **low** **pass** **filter** **code** with NUM and DEN designed by **matlab** (sptool) that fsample=1KHZ;Fc=40Hz;ButterWorth IIR **filter**
#define pi acos(-1)
double Numerator={0.00018321602337,0.000732864093479,0.001099296140218,0.000732864093479,0.00018321602337};
double (...)

Digital Signal Processing :: 12.04.2008 05:51 :: hbaocr :: Replies: **7** :: Views: **919**

in one symbol period,if there is no zero-crossing point,what should I do?according to the gardner algorithm,in one symbol period,we should haver two zero-crossing I right?
Nothing. PLL **low**-**pass** **filter** will do if you once have no zero-crossing.
Positive and negative zero-crossings will be accumulated by **low**-pa

Digital Signal Processing :: 25.05.2008 03:27 :: fontp :: Replies: **6** :: Views: **4302**

What exactly is wrong? As far as I can tell, you don't have a "canonical" two pole **low**-**pass** **filter** because your transfer function has a pole at zero, another pole and a zero.

Digital Signal Processing :: 18.02.2010 14:32 :: Emanuel_hr :: Replies: **1** :: Views: **1756**

Hi guys , i need some advice/tips/guidance. Pls share.
So , i am supose to design a **low** **pass** **filter** with a certain cutoff frequency
and i'm suppose to **filter** those high frequency component of the sound in time domain(convolution) and freq domain(multiplication).
So , i am not sure how to do it..i took a snippet from (...)

Digital Signal Processing :: 12.03.2011 06:46 :: microelectronics :: Replies: **0** :: Views: **492**

I'm pretty sure I'm screwing this up pretty badly due to my limited understanding of **matlab** but I am trying my best. A **pass**ive **low** **pass** **filter** was assembled and measured the output via an o-scope. At each frequency from the signal generator we wrote down the input/ouput Vpp. Using that data I am trying to (...)

RF, Microwave, Antennas and Optics :: 20.09.2012 21:48 :: chart2006 :: Replies: **2** :: Views: **2710**

I have one problem.
I have a data vector. for example : data=rand(1,n) ; n=100;
I want to obtain discrete fourier transform of it by DCT command.
and then,by a **low**-**pass** **filter**,remove unwanted high frequencies of larger than f1.
f1 is determined.
and finally , again convert **filter**ed data from discrete frequency domain to (...)

Electromagnetic Design and Simulation :: 22.04.2013 12:14 :: ghasem_008 :: Replies: **0** :: Views: **273**

Do you want to change the unwanted points to zero, or do you want to delete the unwanted points so the data set becomes smaller?
I suggest comparing the original data with **low**-**pass** **filter**ed data. Wherever the two values are significantly different, those are your important points.
If you provide a link to your data, maybe someone could (...)

Digital Signal Processing :: 01.12.2005 09:30 :: echo47 :: Replies: **5** :: Views: **1792**

Hi
There is a book named "Microwave Devices & Circuits" by Bhal & Bhartia. You can find it very usefull. I have done implementation of **low** **pass** **filter** (Butterworth & Chebyscheff). Refer the **code** attached.

RF, Microwave, Antennas and Optics :: 21.05.2006 11:25 :: louisnells :: Replies: **7** :: Views: **2524**

try using **low** **pass** **filter** or averaging ...

Digital Signal Processing :: 19.03.2009 17:16 :: ittefaq11 :: Replies: **2** :: Views: **3907**

Hi.....
you know your design better.. so asking a **code** :?:
i suggest some steps.. fol**low** that.. its easy to use **matlab**...
in command prompt type
>> fdatool
i.e **filter** design automation tool...
just fill out your required band**pass** or **low** **pass** (...)

EDA Jobs :: 14.07.2009 00:40 :: akss_here :: Replies: **1** :: Views: **2550**

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** (...)

Digital Signal Processing :: 22.06.2009 07:59 :: guest_1044 :: Replies: **1** :: Views: **955**

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: **3463**

Baseband signal can be generated in number of ways . Refer a standard communication text book to represent band **pass** signal interms of baseband signal.
One simple way to generate a arbitrary baseband signal is to generate a PRBS pattern and **pass** it through a **low** **pass** **filter** with appropriate cut off (...)

Digital communication :: 25.03.2010 22:18 :: mathuranathan :: Replies: **4** :: Views: **1680**

to implement the fol**low**ing **filter** equation
a(1)*y(n) = b(1)*x(n) + b(2)*x(n-1) + ... +b(nb+1)*x(n-nb)
use the **filter** function in **matlab**
more info @
regards
Mathuranathan

Digital Signal Processing :: 01.06.2010 05:34 :: mathuranathan :: Replies: **2** :: Views: **1232**

I think, you got integer overf**low** inside **filter**. Check that signal's amplitude and frequency are in valid range. Decrease amplitude.
Try to analyze **filter** output as integer numbers (not dB representation)

PLD, SPLD, GAL, CPLD, FPGA Design :: 21.06.2010 04:14 :: alexadmin :: Replies: **4** :: Views: **1904**

We would need to see what you mean. Can you post an image of the spikes? A simple **low** **pass** **filter** may fix it, but it depends on the underlying signal. Also, the problem may be in your circuitry so the solution would be to fix the problem rather than try to **filter** the signal afterwards. A bit more detail would therefore (...)

Analog Circuit Design :: 14.10.2010 04:50 :: keith1200rs :: Replies: **1** :: Views: **505**

Hi,
I have written a test program in **matlab** for **low****pass** fir, when i m plotting its magnitude response after fft, it seems to be a high **pass** **filter**. Why its doing so??
My **code** is as fol**low**:
N=50; %**filter** order
Length=N+1; %length
n0=N/2; %delay (...)

Electronic Elementary Questions :: 31.10.2010 04:23 :: Naveed Ahmed :: Replies: **3** :: Views: **1589**

Hi,
I have written a test program in **matlab** for **low****pass** fir, when i m plotting its magnitude response after fft, it seems to be a high **pass** **filter**. Why its doing so??
My **code** is as fol**low**:
N=50; %**filter** order
Length=N+1; %length
n0=N/2; %delay (...)

Digital Signal Processing :: 31.10.2010 04:44 :: Naveed Ahmed :: Replies: **3** :: Views: **2956**

Hey guys , help me to check this **code**.. i cant figure out how to make it run .. T.T
Its just to **filter** out some high frequency in the song file i cut ..
function **filter**_a()
= wavread('cut2');
N = length(song); %length of song is 99093
k=;
f1=k*fs/N;
cf = 2000 ;
n = ;
t = n/cf;
N = length(t);
f = n*cf/N;
hf = [o

Digital Signal Processing :: 12.03.2011 08:51 :: microelectronics :: Replies: **0** :: Views: **1434**

I have ISE but not System Generator, is a digital **filter** implementable on a 30K gate FPGA? What is the minimum number of gates to generate a decent **filter** for **low** frequencies such as a band **pass** **filter** less than 100Hz.
- Jayson

PC Programming and Interfacing :: 25.01.2003 22:52 :: Jayson :: Replies: **9** :: Views: **2991**

For my fir **filter**,I need a **low** taps fir for implementation.I had tried windows and remez method,but the taps is too high.
I heard that optimization fir **filter** design method is good,anyone can provide some program or papers about it to help finish the design.

Digital Signal Processing :: 22.03.2004 06:33 :: blueteeth :: Replies: **6** :: Views: **2140**

Hi!
I have trouble to understand how a digital PLL for discrete valus work.
Say that we sample a 1 MHz sinus wave with an ADC 12bit/15 Msps.
Then we hava a bunch of values to do signal procesing on.
What I do not understand how each sample (dscrete value) should be
processed in a digital PLL.
The first sample will be multiplied with

Digital Signal Processing :: 11.04.2004 11:15 :: Rikard :: Replies: **2** :: Views: **1450**

Thanks for your help first. Then,
I jsut used C for a reference model. Here I try to design a LPF. I mixed 1.5M and 10M as input. The LPF **pass** band is set to 1.5M.
I can see the freq. of the output is less than that of the input, but not equal to 1.5M **low**-freq input.
Is this result right? How to verify the detail performance requirement?

ASIC Design Methodologies and Tools (Digital) :: 21.02.2005 01:19 :: qjlsy :: Replies: **4** :: Views: **1137**

I came across the fol**low**ing problem in FIR design using window and convolution techniques .
Project a FIR **filter** using a Hamming window to achieve the fol**low**ing specifications:
**pass**band 0,3-3,4 kHz
Stopband 0-0,2 e 4-8 kHz
Stopband atenuation > 25 dB
Sampling frequency 32 kHz
Since the transition band is (...)

Digital Signal Processing :: 08.10.2005 22:53 :: claudiocamera :: Replies: **0** :: Views: **1109**

A script with a strip down version of a DMT Adsl system.
Description: The fol**low**ing is a simulation of the DMT system, similar to implementation as done in ADSL systems. The bandwidth has been divided into 32 sub-channels of equal width. Each sub-channel has been allocated bits in the range of 1-15 bits/symbol/Hz.
In a finite length DMT sy

Digital Signal Processing :: 25.01.2007 11:35 :: mileoung :: Replies: **1** :: Views: **2840**

Hi All,
I have this one small audio file which is corrupted by an unknown interference signal which needs to be **filter**ed 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**

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 :: 11.10.2011 22:43 :: vickyuet :: Replies: **1** :: Views: **1642**

the first test is to do time domain tests. An impulse like 0x40, 0,0,0,0,0 ... this will give scaled versions of the coefficients. This is a good test signal for systems that perform rate-changes as well. This is mainly to show the **filter** was correctly translated to HDL. Variations on this can be done. Ideally, if the coefficients are correct,

PLD, SPLD, GAL, CPLD, FPGA Design :: 17.04.2012 03:42 :: permute :: Replies: **2** :: Views: **504**

Hi,
I am working on EEG signal. At first i applied the Butterworth **low** **pass** **filter** to extract 0-64 Hz frequency. Then i applied DWT to extract BETA (16-32Hz) and ALPHA(8-16Hz) wave . So, according to theory , 2nd and 3rd level coefficient of DWT should provide the beta and alpha wave. But when i performed the fft of D3 wave i did not get (...)

Digital Signal Processing :: 25.02.2013 04:49 :: Riheen :: Replies: **0** :: Views: **283**

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 qgimol
I don't have working **code** but would like to propose a procedure.
You can start fom white noise. One with uniform PDF can be
generated quite simple using the congruent algorithma or just
use classical PRBS with shift register and xor beed backs.
Then you need pink like **filter** in order to shape this noise.
You can design one

Digital Signal Processing :: 08.07.2004 02:51 :: dora :: Replies: **2** :: Views: **1443**

Hi Manikanta
some explanation about Group delay
%Group delay:
%
% Imagine an AM signal with a sinusoidal modulation. The carrier is high-frequency and the modulation is **low**-frequency. Compare the AM signal with the unmodulated carrier: they are in phase (the zero-crossings are coincident).
% Now, let **pass** the AM signal by a linear network

Digital Signal Processing :: 20.10.2004 03:04 :: dora :: Replies: **1** :: Views: **1333**

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