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52 Threads found on edaboard.com: **Lpf In Matlab**

I am trying to design a low pass filter using butter filter in **matlab** I am confussed how to od this. If my carrier Fc is 20000,( but can be changed if needed) and Fs is 44100 (MP3 file) what do I put in the **matlab** function below? Please help me I can not get htis please??
I need to filter: outfm = Ac * cos(wc.*t + 2 * pi * kf * cumsum(snd));
Fc

Digital Signal Processing :: 05.12.2005 21:53 :: locchamp :: Replies: **3** :: Views: **7230**

alz help me in doing this problem..............
"Make an m-file to low pass filter a signal x(t) consisting of two
cosines at 5 and 20 rad/s using an RC **lpf**. Choose suitable values of R
and C to allow the cosine at 5 rad/s to pass through and suppress
(filter out) the cosine at 20 rad/s. Plot the spectrum of the input
and output signals, as we

Miscellaneous Engineering :: 06.03.2011 08:52 :: umair_ae :: Replies: **0** :: Views: **1250**

Hi,
Iam trying to do some analysis for a redar system that uses Linear Chirp modulation. I have a problem mixing the Tx and Rx signal then **lpf**. RX is a delayed version of the Tx. Normally after mixing (muliplying Tx and Rx) i dont get the low freq part!!!
I tried **matlab** Help but i cannot get it!
Any help Please?
Ho

Mathematics and Physics :: 20.04.2006 18:12 :: ahmad_abdulghany :: Replies: **2** :: Views: **900**

for example in a communications system the the signal band may only be narrow, KHz wide, but the signal band could be centered at RF frequencies, at many MHz. If the signal is sampled according to the Nyquist criterea, i.e. twice the highest frequency, then the data rate for the RF signal will be very high. Processing the data at this high
rate is

Digital Signal Processing :: 06.08.2007 07:43 :: plc :: Replies: **3** :: Views: **2214**

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: **2643**

I wish to implement a discrete High-Pass-Filter (HPF) in **matlab**/Simulink. I found a discrete Low-Pass-Filter (**lpf**) in one of the Simulink examples as z
Denominator: z-exp(-2*pi*fc*Tsc)
fc: cut-off frequency
Tsc: sampling period
Could some

Electronic Elementary Questions :: 09.05.2009 13:15 :: powersys :: Replies: **2** :: Views: **7086**

Hi
the best way is to read **matlab** help. see the **matlab** entries for commands:
"butter" that u can set it as a **lpf** .
"filter".
a glance will guide u through.

Digital Signal Processing :: 08.12.2005 08:46 :: mro83 :: Replies: **9** :: Views: **31213**

Hi.
I agree with others. If you do the D/A conversion in **matlab** so you should not use **lpf**. But I just want to mention a few things:
If fs is samplinf freq and fin is input frequency:
fin = (N/M) * fs
and GSD(M,N) = 1
in this case you don't have to use windowing and then get FFT. You just get FFT, halve it, find the main bin and the rest is noi

Analog Circuit Design :: 29.01.2006 02:30 :: ezt :: Replies: **5** :: Views: **1157**

We will design a transceiver in WLAN we should simulate the system architecture in **matlab** or ADS at first to evaluate the parameters of the all blocks(LNA,MIXER,**lpf**,etc),such as gain,p-1db,IIP3,NF,etc.If we get the parameters,we can design circuits of the blocks in CADENCE.But I am confused now,how can i model the blocks in **matlab** or A

RF, Microwave, Antennas and Optics :: 22.11.2006 00:32 :: longstar :: Replies: **3** :: Views: **641**

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: **4244**

i wan to impliment the filter(**lpf**,HPF,etc) in **matlab** ihave tried but can't done
any one who done it plz help me

Electronic Elementary Questions :: 19.02.2007 02:23 :: halee awan :: Replies: **2** :: Views: **507**

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 of filter .I have to pass those two throug

Digital Signal Processing :: 24.02.2007 13:16 :: PraveeGoud :: Replies: **5** :: Views: **1090**

Use FDATOOL in **matlab**. If you type
>> fdatool
in command window, Fda tool will be opened. There you can select FIR or IIR filter, order of filter and cutoff freq. of filter (either HPF, **lpf** or BPF). That code will automatically generate .m file for you.

Digital Signal Processing :: 18.04.2007 11:18 :: sprao :: Replies: **26** :: Views: **62176**

Hi All,
How to measure the SNR of SC **lpf** in cadence and **matlab**...Can anybode tell me the procedure for measuring SNR of system in cadence..?
thanks

Analog Circuit Design :: 17.01.2008 23:02 :: rampat :: Replies: **0** :: Views: **493**

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: **853**

I have used it, well certainly not to the most possible extreme, but to my knowledge, I have created a **lpf**, and had my resuts succesful, but had some issues with the cut-off frequency range. You first need to add FDA tool into your design. use the **matlab** tool to design your filter. After your design, just drag a n-tap FIR Filter block from DSP. Upo

PLD, SPLD, GAL, CPLD, FPGA Design :: 15.09.2008 06:24 :: xtcx :: Replies: **4** :: Views: **1801**

Hai everyone,
I have an i/p signal x = 1 +0.5* sin(pi*t) where t = , basically a signal sampled at 100 Hz. I need to filter out the d.c. component i.e. get 1 as a constant output so for which I have to design a low pass filter.
can any one tel me how to design low pass filter in **matlab**
Thank you

Digital Signal Processing :: 16.02.2009 03:07 :: rramya :: Replies: **2** :: Views: **623**

hello everyone, I am trying to model **lpf** noise contribution to PLL output. do I need to simulate the **lpf** alone to get noise profile(like CP and VCO), then put the noise in PLL closed loop sim(verilog or **matlab**)?
my question is what is the phase noise contribution of **lpf**? is it low pass(I believe so) ? what is the (...)

Analog IC Design and Layout :: 11.11.2009 20:33 :: hearter :: Replies: **2** :: Views: **887**

Use the following 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
=butter(5,w,'low'); %5th order butterworth **lpf**
=freqz(b,a,1024); %Frequency response

Digital Signal Processing :: 25.05.2010 06:03 :: mathuranathan :: Replies: **6** :: Views: **15122**

Hey,
i've line in **matlab** example code:
**lpf** = **lpf**_b***lpf**_prev + **lpf**_a*mixer_I + **lpf**_a*mixer_I_prev ;
This is part of 'for' loop where is signal 'mixer_I' filtering.
I see this is IIR but I dont know how could I write its transfer function.
This is one of **lpf** in (...)

Digital Signal Processing :: 26.10.2010 21:53 :: daniel488 :: Replies: **1** :: Views: **560**

I am need of designing a loop filter for costas loop.
I was said that loop filter is nothing but iir filter of order 2.
For example costas loop to track 25khz carrier frequency with .02% crystal oscillator deviation.
In this example a 6 bit nco is used.
After **lpf** section on both arms and phase detection we get a vector 16 bits
NCO uses a acc

Digital communication :: 15.11.2005 16:16 :: shankar :: Replies: **0** :: Views: **2193**

I'm not famliar with digital filters. I want to implement a simple one to learn. Here is an example:
In that, there is a coefficient α, which is obvious, an output, a previous value of the output and an input. I wonder, if i want to apply this how would i decide somethings like;
1-What

Digital Signal Processing :: 16.02.2006 11:45 :: seyyah :: Replies: **2** :: Views: **636**

I want 4kHz **lpf**, I design R=780K, C=10pF, and 3 same RC in series, the -3dB is about 4KHz, is the design right?
if right but the size is too big to integrate in IC, can anyone give me some instruction to design a low cost **lpf**, thanks!

Analog IC Design and Layout :: 26.07.2007 21:46 :: xiexi :: Replies: **2** :: Views: **1797**

I want to design a efficient filter for QPSk in **matlab**'s simulink DSP block set. I know that i need upsampler+**lpf**+Interpolation filter but i am looking for such a filter which can do all thses together. Is it possible? If yes or not, please suggest me and give me some IEEE paper where i can get the design. Thanks in advance.

Digital communication :: 18.02.2008 18:38 :: somik13 :: Replies: **1** :: Views: **672**

Hi, guys, If we already have the parameters for passive third-order **lpf**, how to calculate the natural frequency and damping factor?
we can find the equations to get these for second-order **lpf**. but I do not know how to get those for third-order passive **lpf**.
thanks.

Analog Circuit Design :: 09.07.2008 08:27 :: gavin168 :: Replies: **4** :: Views: **3580**

Hi,
I am trying to simulate the behaviour of a type 1 PLL using simulink, the loop parameters are:
W**lpf**=2pi*(1 MHz);
Kvco=100M
Kpd=1
So the loop T.F. is T(s)=(Kvco*Kpd*W**lpf**)/(S^2+W**lpf***S+Kvco*Kpd*W**lpf**);
I am using three blocks:
1. input block which is a step source to simulate the input phase.
2. "Transfer function" (...)

Analog IC Design and Layout :: 18.07.2008 07:25 :: yassin2705 :: Replies: **9** :: Views: **1422**

1. Any suggestions on a multiplier free **lpf** implementation?
Even if you manage to build a multiplierless **lpf** it is very likely that it won't have good performance.
2. Will a CIC work for the second stage also?
It may or may not depending on bandwidth of the signal of interest. You may end up with a situation when the order of a CIC required t

Digital Signal Processing :: 30.06.2010 03:41 :: hobgoblin :: Replies: **1** :: Views: **795**

Dear all:
I have wrote a pll **matlab** file to simulation PM, BW..., the m-file is ok when
I use **matlab** r13.
But our company update the **matlab** to new version r14, I can't us 'tf' to create
the transfer function, could you help me?
the following is the example for my m-file:
Rp = 4.82e3;
Cp = 118.1e-12;
Cs = 11.3e-12; (...)

Analog Circuit Design :: 28.07.2010 06:10 :: mpig09 :: Replies: **0** :: Views: **699**

I am having a wave file. I have to implement this operation on it. the picture is attached
I have read the wave file in **matlab**. Now i want to down sample it.kindly help me to implement second part of picture.
the **lpf** should have 6 KHz

Digital Signal Processing :: 12.03.2011 01:53 :: moonnightingale :: Replies: **0** :: Views: **308**

Hi. I have the same problem. I am trying to demodulate BPSK signal and here is my way: BPSK signal .* carrier => integral => compare with 0 (if >0 is 1 and <0 is -1)=> recover original signal.
Use can follow this link: but if we use **lpf**, the BER is really high. My teacher said "use inte

Digital Signal Processing :: 27.02.2013 02:13 :: sadsorry :: Replies: **3** :: Views: **1020**

a First order markov process: essentially the present state does not depend on the past is the highlight given the present state value. For the markov model you can model a simple Binary channel with probability P. i.e. if there is noise the 0->0 as 1-P and 0->1 with prob P. Similarly, for "1". if your using **matlab**

Digital Signal Processing :: 08.09.2011 21:10 :: kalyanasv :: Replies: **1** :: Views: **324**

hello, i have designed a **lpf** using fdatool, now i wanna use this filter to filter an input signal x , so i use filter(Hd,x) ,in the generating m file with the designed filter but it turns error, what should i do, use export instead or what? how to use the filter designed to filter my data within my m file code ? thnx

Digital Signal Processing :: 27.12.2011 19:27 :: kimo4ever :: Replies: **0** :: Views: **528**

Hi ALL
in now days i design FIXED POINT FIR that wil be implemented on FPGA
the filtering unit using cic decimation followed by 2 fir **lpf**.
the input to the unit is 32 bits - 30 fractions and 2 for real number.
the end of the unit in 57 bits and i take just the fractions -51 downto 22.
when i analyze the FILTER FREQUENCY RESPONSE ( for t

Digital Signal Processing :: 06.05.2012 03:26 :: itmr :: Replies: **0** :: Views: **298**

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: **1111**

Hi,
I am new to Digital Communications and **matlab**. I am learning to implement modulations and demodulations in **matlab**.
I have an amplitude modulated signal at the receiver input. I am trying to recover the message bits after the demodulation.
Following are the parameters of the modulated signal:
Carrier frequency = 22000 Hz (can be

Digital communication :: 21.03.2013 12:50 :: v_suma :: Replies: **0** :: Views: **1055**

Hello Everyone
I am working on Delta Sigma ADCs in **matlab**, to equalize the amplitude of input and output I have to multiply the output by a scaling factor, anyone who has experience of ADC (using or working on ADCs) share the reason for this scaling ? and Secondly the SNR after Delta Sigma Modulator ~66 dB and after Decimation it drops to ~50dB

Digital Signal Processing :: 05.04.2013 02:44 :: Eminent.Engineer :: Replies: **0** :: Views: **179**

Hello Everyone
I have got the output of Delta Sigma ADC after Decimation (FIR **lpf** and Downsampling), How to convert Delta Sigma ADC's output to its voltage equivalent ? Any suggestions/relevant material will be he**lpf**ul.
Thanks in Advance.

Digital Signal Processing :: 08.04.2013 03:47 :: Eminent.Engineer :: Replies: **2** :: Views: **279**

Hi,
Can anyone please tell how delay and multiply affects a speech signal.
I was trying to implement Harris algorithm for Vox in **matlab**.
It was a simple one in first look with a delay and multiply and an **lpf** after that.
But i could not get the expected output.
I am newbie to DSP.
Thanks,
RTV

Digital Signal Processing :: 16.04.2013 08:56 :: ragfox :: Replies: **1** :: Views: **182**

It is not clear for me if the signal has a sinx/x shape in the time domain or in the frequency domain.
If you wish attenuate the side lobules with a low pass filter use
y is you signal in the time domain
fl=100; % **lpf** length
fbe=; damps=; % design of **lpf** parameters, analyze

Digital Signal Processing :: 27.04.2013 13:49 :: danielr :: Replies: **1** :: Views: **380**

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: **1078**

Hi all,
I am newbie in wavelets..
Just read the **matlab** wavelet tutorial as I found it the simplest amongst others to understand..
I want to ask that during the DECOMPOSITION phase in the DWT, when a signal is convolved with HPF and **lpf** to yield details and approximation..
Q: How are the co-efficients chosen for the HPF and **lpf** so (...)

Digital Signal Processing :: 07.01.2006 08:47 :: sanjay :: Replies: **1** :: Views: **545**

There are two methods
1. between each samples add 7zeros and take **lpf**.
2. If u want to take interpolation for 50 samples first take fft of 64 samples by
padding zeros.
Consider only 33 values. add 224 zeros to it (224+257)
Take ifft of 512 samples.
These two methods explained in Scientist and Eng. guide to DSP.
Bye
Sanbaba

Digital Signal Processing :: 26.05.2006 00:47 :: sanbaba :: Replies: **3** :: Views: **1604**

digtal PLL: digital PFD+charge pump+**lpf**+VCO, most porpular now;
all digtal PLL: digital PFD+digital low pass filter+DCO, it have high phase noise, for example, MT4409 from Zarlink;
analog PLL: analog PFD+**lpf**+VCO.

Analog Circuit Design :: 22.01.2007 20:32 :: butterfish :: Replies: **11** :: Views: **4627**

By the look of it, it looks like a low frequency signal ADDED to a high frequncy signal. If this the low frequency one available elsewhere or known in advance, you can just subtract it from the signal. You can laso try subtracting away your **lpf** output from the signal.
But these methods assume that there is no change in the peak amplitude of the

Digital Signal Processing :: 23.05.2007 08:53 :: bulx :: Replies: **12** :: Views: **1317**

If you just want to use a black box and don't want to build the whole transmitter and receiver, you can use the FM modulator/demodulator passband present at the communication blockset=>Modulation=>Analog passband modulation.
If you would like to build the transmitter and the receiver, you can use the Voltage-controlled oscillator present at comm

Digital communication :: 08.08.2007 07:15 :: ieropsaltic :: Replies: **3** :: Views: **2933**

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 normal transient response

Digital Signal Processing :: 20.11.2007 11:50 :: Arturi :: Replies: **0** :: Views: **809**

Hello Radiohead,
Here is the attached PDF on **lpf** filter design using coaxial resonators,
using AWR Microwave Office (MWO) & **matlab**...
---manju---

RF, Microwave, Antennas and Optics :: 13.05.2008 01:22 :: Manjunatha_hv :: Replies: **2** :: Views: **2176**

Hi, I'm a licenciate student in Finland. I am looking for Analog/Mixed-Signal/RF IC Design Job in Europe or US.
I have about five years IC design experiences. I designed DAC, **lpf**, PA, Opamp. I'm familiar with most popular EDA tools like composer, virtuoso, hspice, hsim, spectra, spiceexplorer, calibre, xcalibre, protel, **matlab**/simulink etc. I ha

EDA Jobs :: 28.01.2010 15:13 :: waosai :: Replies: **0** :: Views: **721**

Hi, all..i'm new in Sigma Delta Modulator.
I try to built SDM for my final project.
I simulate for 1st order SDM which can see in attachment.
I simulate it using Circuit Maker.
I need suggestion about it.
And i have question about how to design high order (> 1) SDM.
I read some paper, and for high order SDM you can replace integrator (which

Professional Hardware and Electronics Design :: 08.02.2012 10:19 :: surz90 :: Replies: **1** :: Views: **933**

You can follow your **lpf** with an all-pass filter (or some other phase-compensating network) to adjust the phase.

Electronic Elementary Questions :: 03.07.2012 09:10 :: barry :: Replies: **4** :: Views: **946**

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