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));
Digital Signal Processing :: 05.12.2005 21:53 :: locchamp :: Replies: 3 :: Views: 7316
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: 1311
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?
Mathematics and Physics :: 20.04.2006 06:20 :: gaboor :: Replies: 2 :: Views: 986
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
Digital Signal Processing :: 06.08.2007 07:43 :: plc :: Replies: 3 :: Views: 2330
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: 2677
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
fc: cut-off frequency
Tsc: sampling period
Electronic Elementary Questions :: 09.05.2009 13:15 :: powersys :: Replies: 2 :: Views: 7298
the best way is to read matlab help. see the matlab entries for commands:
"butter" that u can set it as a lpf .
a glance will guide u through.
Digital Signal Processing :: 08.12.2005 08:46 :: mro83 :: Replies: 9 :: Views: 31906
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: 1190
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: 676
thank you for sharing the files
Added after 8 minutes:
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.
u can use thes
Digital Signal Processing :: 06.02.2007 01:34 :: amihomo :: Replies: 2 :: Views: 4542
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: 536
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: 1124
Use FDATOOL in matlab. If you type
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: 62823
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..?
Analog Circuit Design :: 17.01.2008 23:02 :: rampat :: Replies: 0 :: Views: 519
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: 893
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: 1941
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
Digital Signal Processing :: 16.02.2009 03:07 :: rramya :: Replies: 2 :: Views: 659
The resistor is the only noise component in the lpf and it's noise in the loop is bandpass. It appears between two poles at the origin one in the forward path to the output and the other in the feedback path.
Analog IC Design and Layout :: 12.11.2009 02:44 :: saro_k_82 :: Replies: 2 :: Views: 923
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: 16516
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: 604
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: 2270
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;
Digital Signal Processing :: 16.02.2006 11:45 :: seyyah :: Replies: 2 :: Views: 665
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: 1936
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: 696
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.
Analog Circuit Design :: 09.07.2008 08:27 :: gavin168 :: Replies: 4 :: Views: 3687
I am trying to simulate the behaviour of a type 1 PLL using simulink, the loop parameters are:
So the loop T.F. is T(s)=(Kvco*Kpd*Wlpf)/(S^2+Wlpf*S+Kvco*Kpd*Wlpf);
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: 1461
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: 832
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: 740
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: 349
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: 1107
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: 359
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: 573
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: 324
x1 = load('ecg3.dat');
x2=x1; fs = 1000; % Sampling rate
N = length (x2); % Silength
t = /fs; % tiidx
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: 1326
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: 1090
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: 4 :: Views: 409
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 helpful.
Thanks in Advance.
Digital Signal Processing :: 08.04.2013 03:47 :: Eminent.Engineer :: Replies: 2 :: Views: 336
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.
Digital Signal Processing :: 16.04.2013 08:56 :: ragfox :: Replies: 1 :: Views: 211
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: 447
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: 1127
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: 577
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
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.
Digital Signal Processing :: 26.05.2006 00:47 :: sanbaba :: Replies: 3 :: Views: 1668
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: 4763
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: 1365
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: 3000
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: 870
Here is the attached PDF on lpf filter design using coaxial resonators,
using AWR Microwave Office (MWO) & matlab...
RF, Microwave, Antennas and Optics :: 13.05.2008 01:22 :: Manjunatha_hv :: Replies: 2 :: Views: 2281
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: 800
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: 995
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: 1062