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275 Threads found on edaboard.com: Matlab Sampling Frequency
Hi all, I have a problem..I have a discrete sine with 20000 sample and with variable frequency. I need to plot this freq sweep, I expect a frequency ramp. How I can do this with matlab? Please someone help me? Thanks
Q1 : sir i want to ask you that if i am using an external ADC which have 400ksps t0 sample the frequency of 100khz. if i applied input of 100khz to ADC which converts into bits and these bits can serialy transmit to matlab through pic16fxxx, so it is possible for pic to work as fast to transmit all the bits before taking to next sample duration ti
can somebody tell me what should i do for sampling discrete time signals in matlab i think u ask for the CODE... in that case it is done like... fs = 1000 (sampling frequency) f = 10 (signal frequency) A=2(amplitude) n=0:100; wave = A*sin(2*pi*f*n/fs) this would create 100 sample points wave (...)
i need a funtion for matlab to get the fourier cn coefficients is there any link about this is it fourier series coefficients or fourier transform coefficients needed for u? the fourier transform can be obtained with a simple code as follows.. all are inbuilt commands... t = 0:1e-3:10; fs = (some value)-- sampling fr
I think you'll get a fine resolution (1/M)*fs. Then the new index should be (f0/fs)*M. Please verify it with a simple matlab code.
resolution 10bit, how to generate this binary or decimal digital sine wave using matlab command? I know litter about matlab, thanks.
Hi I tested the codes in matlab 7 and it worked well. here is the graph. cheers
hi adeel, you can give ur ECG signal to "mike" input or "line in" input (line in input must be the correct one, not sure) of the soundcard. the soundcard (which is nothing but an ADC working at a max sampling rate of 44.1Khz). now, you can use the function "wavrecord" in matlab which takes parameters like the sampling (...)
Hi guys I need to design a filter for a Phase lock loop, this is for FM system, if the carrier frequency is 20000 and sampling frequency is 44100. I beleive I need a high pass filter for PLL right? and if I do how do I set the parameters for the filter? THanks.
Hi all, I am a matlab newbie :) How to get frequency spectrum of a vector data? (something like ) I heard FFT is the time-frequency fransform. And I tried fft() function in matlab. But it return a complex number. Is it stand for both frequency and phase? Any suggestions will be appreciated! Best (...)
I have understood. you can use the hspice matlab tools.
Hi, For learning matlab there are a lot of tutorials in internet. You can also find help from . I think first of all you must learn building signals in matlab. Most of the well known signals can be created by built in functions of matlab. For example '>t=0:0.01:1; y=sin(2*pi*10*t); plot(t,y)' will give (plot) you a sinus wit
Hi all, I'm still rather new to matlab and have certain questions to ask. Is it possible to generate a signal at a specific frequency? and is it possible to control the power spectral density of the signal?
how to write codes in matlab in order to make it read a wave file. when i click on the read file button, the software i built must be able to extract the data in the file such the sampling frequency and the number of samples. the software could only analyze a singel channel PCM audio with sampling (...)
Based on 30 seconds of looking, it seems that this matlab code is coming up with the right frequency axis for a spectrum plot, and changing the FFT output from a CX to a real display. Take a look at pwelch instead, it takes care of the gory detail's for you. Dave
hi everybidy, i have simulated a code in matlab for the demodulation of ofdm signal (dab). In the synchronisation process, i found the clock offset of 13.035 khz in the spectrum after downconversion. i used a nco to remove the frequency offset. Now when i differentially decode the symbols, i get 5th and 6th symbols corrupted. I think this can be h
hi all please can any one let clear my doubt "can the sampling frequency of a digitized signal be varied i.e can we vary sampling frq after sampling by adc if so please let me know how we do it "are there any methods for doing so??? yes u can do so,u can change the sampling rate ucsing (...)
I am new to matlab and I was trying to code a sampling and aliasing example. But I have a few questions: clc; clear all; %A is the amplitude A=1; %f is the frequency(Hz) f=100; T=1/f; %fs is the sampling frequency fs=1000; %phi is the arbitraty phase, phi = 2*pi*0.75; tstart=0; (...)
can u send me ur matlab code. my email chaitubek@yahoo.co.in thank u,.. plz.. its urgent
matlab has very good on-line help. Try using the "What's this?" help button in FDATool to display info about the various controls and input fields. For example ...
AoA to ALL! I want to read a sound file in matlab, and set the sampling frequency to 8K. and represent each sample in 8-bits. play the result and also plot the frequency spectrum. i expect the spectrum should be from 0 to 4khz. plz tell me how to do all these in matlab. simple & early reply is required.
hi, I have to design some filters for processing speech in matlab, I know the basic commands but how do I adjust the cutoff frequencies in HZ , how to convert them into the filter parameters
Hi friends, I am trying to display time domain signal in frequency domain using matlab.I took frequency of signal as 100hz and sampling frequency as 1000sampels/sec I have tried using 'fft' function but not getting. Anyone of you give me the solution for displaying frequency (...)
Why not using filter design toolbox in matlab. just type fdatool, you can find everything you want there, you also can export designed filter coefficients to MALTAB or to a DSP.
I do not seem to understand how sampling affects the frequency spectrum. Help me find the frequency spectrum before and after sampling. How to do it in matlab? I am interested in frequencies from 0~integer*sampling frequency. I want to see how different (...)
I have a analog signal through the lowpass Filter(fc=40hz) then I sample it which fsample =40hz, and I recieve 200 pieces of that signal into my computer by sampling which f(sample) =40hz.And my problem is how I can analyse spectrum of those 200 pieces discrete signal in matlab. PLZ help me some code matlab or tell me which macro in (...)
I am trying to calculate the cross-spectral density for a voltage signal using matlab. I am using the fft function to do the Fourier transform. The fft function is Y=fft(X,n). But I donot know how to pick the points n. My voltage signal Y is 2seconds long. dt is 0.0001s. so sampling frequency is 1/0.0001s=10KHz. I donot know what value I (...)
Hi all, i have a problem using the pulse shaping commands in matlab i.e. rcosine( ),rcosfir( ),rcosflt( ). I can't understand the arguments of these commands. Problem: We can transmit 'Rs=2W' symbols/sec (Rs is the symbol rate,W is the available bandwidth) with zero ISI if the available bandwidth is W Hz. But this requires nyquis
anyone help me generate an IMPUSLE impulse train with matlab??? use the Difference Equation to generate a 80-120Hz impulse train with a 8k Hz sampling frequency: y=D+y p.s. any value for k is fine. Thanks!
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
i designed a pipelined adc,and i simulated it at the frequency of nyquist(about half of sampling frequency);but when i deal with the data(txt in dec) in matlab using fuction cal_snr.m(from maxim), the noise power(Pn) is negetive,why?and snr is complex.i'm really confused.i find when i change the parameter "span",the result (...)
Thank you for your code but I an need of a matlab code which actually implements an FFT algorithm, rather than a built in library function. Please send me one if you have it. Thank you :)
I want to realize a simple analog low-pass first order filter in matl ab, but i waant also obtain time domain output if an input signal is present. So: 1)input time domain signal is realized with an array of values (i'm not using symbolic) 2)fft of input signal: i consider only frequencies equal and under fs/2 (with fs=sampling frequency) 3)real
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.
hi every one, i would like to share some of the uses of fft for spectrum analysis. Hope it will be useful for those who are novice to matlab programming. DFT Notes: DFT produces a discrete frequency domain representation.Also, DFT is only defined in the region between 0 and Fs. (as we know, One period extends from f = 0 to Fs, where Fs is
Aliasing Example The program below demonstrates this problem graphically. We simulate sampling two sinusoids, 2 cos(2Π100t +Π/3) and 2 cos(2Π600t +Π/3), at 500 samples/second. They are given the same magnitude and phase for a reason: the program shows that the sampled versions are identical. % % Ex
Hi iam wrinting a simple matlab code that generates a sinusoidal signal which is then sampled ploted. when I take the sampling frequency fs=5000 and f0=4500 hz. The number of samples n=100,. I should get an aliaisnig error right because fs<2*f0.... The problem is that i get all the signal undistorted!!! This is the matlab (...)
Hi,all I am in a project of Sigma-Delta ADC. But I get in trouble when I design the digital filter in the matlab. The filter consists of a CIC, a CICCOMP and two halfband. When CIC is followed by the CICCOMP and then the twe halfbands, the CICCOMP can compensate the droop caused by the CIC in the passband well. But when I exchange the order betwee
Hi everyone I am a serious newbie in this modulation techniques :( I have been reading up and still do not understand. if i need to design a QASK modulation scheme for matlab of the following specifications, how do i do this? Bandwidth: 200 kHz Centre frequency: 1.420 MHz Signal Power available: 1 mW Noise Power Spectral density: 1.0 10
My IF is 5Mhz. I need to reduce my sampling frequency 409.8Mhz to 20.48 Mhz by using FIR lowpass filter, so i am decimating it in 2 stages , first time i am using cic filter and decimation factor 5 then i will get the new sampling frequency 81.9 which is given to half band filter by decimating factor 4 then we (...)
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 so i have to reduce to 20.49Mhz. (...)
I want to read the signal frequencies that can be seen in the spectrogram using matlab function(spectrogram). But when i give different sampling frequencies i am getting the display of my signal at different frequency points.This is explained below. I am using the below code to create a sine wave with freq = 100hz. and display its (...)
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
I designed a filter hilbert with matlab (function fdesign.hilbert and design(,'firls'). I tested it with a cosine but the results are very stranges: for my sampling frequency 100 Mhz if I try with a cosine of an interger frequency (1,2,3 .. 49) Mhz the result is ok, hilbert filter returns the 90 degree shift signal of my (...)
Help please!!!! i have sinwave with 0.1 v amplitude and 1khz frequency mixed with 5v white noise. i used FFT in matlab to simulate. 1-(Study the use of additional mean value calculation of every data set of FFT spectrum and how it affects the visibility of a present signal.). so, how can i use additional mean value??????????? 2-W
Hi, all could anyone provide me the matlab file for FSK modulation and demodulation, or any links is also welcome, I would appreciate it. Thanks in advance. Best regards
So, you have 8ms of data and you are sampling at 16kHz (128 points captured in 8ms). The FFT resolution is then 125Hz and the highest frequency is 8kHz (16kHz/2, Nyquist theorem). In matlab language: f=0:125:8000; is your frequency axis. Keep in mind that matlab will show both sides of your FFT, so you (...)
Hi friends, I am working to realize HSP50110 / 50210 architecture, used for demodulation in matlab and FPGA. I started with matlab's inbuilt example "symbol timing recovery with fixed sampling" and extended the design for HSP50110. I tried to replace Squaring loop method with either Gardner or Early-late Gate algorithm, but no success. (...)
I am generating UWB signal of 2GHz with matlab sampling frequency(for resolution) of 8GHz.Thus, each UWB pulse has 4 sample values. Now the number of samples in the received signal are 800. I do compressed sensing with Phi=randn(400,800) reconstruct the signal successfully to 800 samples. Now the question is, the compression in (...)
matlab is a discrete solver. When you write x=cos(2*pi*t1); (t1 not t, therefore f=1) you have already multiplied your cosine by an impulse train whose sampling rate is fs=6/1000... If you want to see the meaning of the Nyquist theorem just keep increasing the frequency until it is higher than fs/2. What do you see?