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246 Threads found on edaboard.com: **Calculate Fft**

Hello all,
I am looking for a software to **calculate** **fft**. In order to have an idea about SFDR .
Does anyone know a good one (free would be great)?
Thanks a lot.
fasto

Software Links :: 30.01.2010 05:52 :: fasto2008 :: Replies: **1** :: Views: **439**

Hi all,
I'm new to DSP. I'm trying to **calculate** **fft** for an ADC. I captured the digital output from the ADC by using a Digital comparator and stored the output code in a file. I need to know how i can use this digital output codes(1's and 0's) to **calculate** **fft** , as **fft** needs real and imaginary (...)

Digital Signal Processing :: 15.06.2010 08:23 :: wks_19 :: Replies: **2** :: Views: **1291**

Hi forum,
How to **calculate** **fft** core SQNR?
Sounds very easy but seems it is not.
I have a Verilog RTL core for **fft**.
I have a very accurate reference model for **fft**.
Let's assume that my RTL core outputs are R0, R1, R2.......Rn.
And reference model outputs are M0, M1, M2.......Mn.
Having these samples how can I (...)

Digital Signal Processing :: 01.03.2011 10:33 :: Syswip :: Replies: **2** :: Views: **1135**

How do we caluculate **fft** from a group of sampled values
Iam interested to **calculate** frequency from **fft**.What happens to the imaginary term in the **fft** in real time.
Thanks
Any basic DSP text book can give you the equation for DFT and **fft** algorithms like Radix2 and Radix4. Once you (...)

ASIC Design Methodologies and Tools (Digital) :: 17.01.2005 10:29 :: eda_wiz :: Replies: **7** :: Views: **1369**

Hi,
I am trying to develop a function in Matlab to **calculate** **fft** using DIF RADIX 2. In order to test it , firstly I am working with a signal with length =8 x= . Unfortunatelly it is not returning the correct result, I cant find what is wrong with the algorithm.
If somebody realise what is wrong in the code below, please let

Digital Signal Processing :: 21.10.2005 22:24 :: claudiocamera :: Replies: **1** :: Views: **5864**

Hi
thanks
now I want to take DFT.
is it true:
dft(sample("out" 2ns 22ns "linear" 100ps) 2ns 22ns 128 ...)
wil this **calculate** **fft**?
rehards

Digital Signal Processing :: 22.12.2006 02:53 :: hr_rezaee :: Replies: **10** :: Views: **1098**

I am trying to **calculate** **fft** on the PIC16F877A, since the PIC can't provide enough RAM. I have store each calculation to the SRAM.
Here is the sample code.. I still can't get it to work though...
Any help would be great...
//Source code from here:
//the adc input data is

Digital Signal Processing :: 06.12.2007 06:14 :: slickuser :: Replies: **3** :: Views: **4982**

Hi
I have been trying to **calculate** **fft** based on Radix-4 in Excel. Also I have been comparing my result with matlab. But I am stuck at the moment because I felt something did not seem to be correct. Please find the attached file.
Any help would be appreicate
MM

Digital Signal Processing :: 09.07.2009 06:01 :: Maverickmax :: Replies: **1** :: Views: **1063**

:D Hello!
I have been implementing OFDM with fixed-point **fft**. I have 16 bits at Tx output for DAC, but I have only 6 bits ADC for Rx. As you can see below
At Tx.....Outout from FPGA 16 bits then connected to DAC .........
At Rx....ADC with 6bits input to FPGA for **fft** processing.
Therefore, the problem is how to process and (...)

Digital Signal Processing :: 26.11.2009 16:00 :: kpuntsri :: Replies: **4** :: Views: **776**

i need to **calculate** **fft** of the sinusoidal output of incremental ADC with 16 bit resolution
In order to achieve sinusoidal output vs input samples, i gave ramp input voltage amplitude from 0 to 3V for 20000 samples in sin function with fixed frequency freq =1 and fs = 60K with ADC cycles 257 and veried the phase from 0 to 2pi.
algorithm:
fo

ASIC Design Methodologies and Tools (Digital) :: 06.01.2012 13:15 :: singhji0000 :: Replies: **0** :: Views: **600**

While comparing two approaches to **calculate** **fft**, better to use identical data with 2^n (say 1024) points. The outcome should agree. You can do that using your own program. availability of a sample data will also help more quantitative responses. You may try math cad too.

Electronic Elementary Questions :: 01.04.2012 18:45 :: ark5230 :: Replies: **3** :: Views: **1204**

Hi,
i am trying to use 3gsps adc of e2v's ev10as152a, the datasheet is claiming that this adc can give upto 5GHz bandwidth (better SFDR even in 2nd and 3rd Nyquist zones). but when i **calculate** **fft** for this digital data(after down sampling) i can only get Fs/2 bandwidth i.e i can only get upto 1.5Ghz frequency calculation. How can i (...)

Digital Signal Processing :: 06.08.2013 09:20 :: appalanaidu :: Replies: **0** :: Views: **437**

I want to build a harmonic current meter for power supply testing,I use AD converter to sample AC input current ,but how can i **calculate** harmonic current ? what is the formulate? how to use **fft**?

Professional Hardware and Electronics Design :: 12.02.2004 22:19 :: alphi :: Replies: **3** :: Views: **2232**

Would greately appreciate some advice.
If I have say 256 sample of a digitized waveform.
Is it possible to **calculate** the frequency from these sample ?
Any info wouls be much appreciated.
Thanks.

Digital Signal Processing :: 19.07.2004 13:01 :: jimbies :: Replies: **3** :: Views: **2809**

hello!!
Somebody can tell me !! How to **calculate** SNDR of ADC??
How to find Matlab program that to analysis dynmic parameter of A/D Converter ??
thanks !!!

Analog Circuit Design :: 03.09.2004 02:11 :: chrischen :: Replies: **3** :: Views: **3249**

THD is simulated by percent,while noise is nv^2/hz,how to get the THD+N by percent??? :) thanks first!

Analog IC Design and Layout :: 27.10.2004 07:12 :: nickooeda :: Replies: **2** :: Views: **2340**

Yeath , you are right.
In the mean while , i know the step frequency (approximating delta frequency=2/(Tstop)) , by it , we can **calculate** the point number for frequency range estimated by **fft** .
But the frequency step effects the time of estimation and the accauracy. How does we select the step as resonable as possible . Is there e

Analog Circuit Design :: 01.01.2005 00:33 :: andy1 :: Replies: **4** :: Views: **994**

Hi,
Is possible to **calculate** theorically Swtiching transients from Spectrum due to modulation by taking into account the rising and falling edge of the burst ramp? Is there any theorical calculation of Switching due to transients in GSM?
Is there any paper about this subject?
thank you for your help.
Rafouille

RF, Microwave, Antennas and Optics :: 25.04.2005 15:46 :: Rafouille :: Replies: **2** :: Views: **1001**

you should got the final value of S/H output at each hold phase. Then perform **fft** to **calculate** the SNDR & ENOB.

Analog Circuit Design :: 10.07.2005 12:23 :: Lantis :: Replies: **4** :: Views: **1772**

I need help for my pipelined ADC. I sampled the output and used matlab to get **fft**. Using the program in this forum, I **calculate** SNR,THD,etc. But I have problem during this procedure. I sampled the single sin wave, using Matlab code, SNR is just about 100dB. But when I tried to sampled the ouput from ideal DAC(I tested the example in Baker's book(mi

Analog Circuit Design :: 12.08.2005 22:50 :: tyd :: Replies: **8** :: Views: **3915**

Hi,
I have designed a second order sigma delta for sampling frequency of 12.8MHz and OSR=256 in Hspice. I don't know how to **calculate** the SNDR. I mean how to set these parameters: input frequency,transiation step and N (# of **fft** points).
Thank you in advance,
tatakt

Analog Circuit Design :: 14.08.2005 02:50 :: tatakt :: Replies: **4** :: Views: **1668**

Hi,
There is no differennce between DFT and **fft**. Both operations do the same thing, but **fft** is faster in computation than DFT. Actually, in **fft**, a divide and conquer approach is used to **calculate** the DFT. The algo for calculating **fft** is as follows:
For a length N complex sequence , , the discrete (...)

Digital Signal Processing :: 12.12.2005 00:17 :: shameem :: Replies: **3** :: Views: **2920**

Hi all,
I am design a 10bits pipeline ADC and by followingprevious articles I use Matlab to do **fft**. But here are the problems I met.
fin=100Khz, fs=10Mhz,
N=8192(number of data points in **fft**),
M=83 (integer number of cycles within the sampling window),
First I used coherent sampling method to do **fft** and I got a result, freqd_1. (...)

Analog Circuit Design :: 17.01.2006 04:17 :: yushen_yang :: Replies: **1** :: Views: **1126**

I'm not sure either.
But i do remember that Ali Niknejad, a Professor at berkeley, said something about this during one of his
lectures last semester. He said sometimes , people think they are looking at the phase
noise, but its not acutally the phase noise, its the distortion due to the windowing effect of running the simulation for only a

Analog Circuit Design :: 07.02.2006 03:24 :: eecs4ever :: Replies: **3** :: Views: **1232**

For a sine wave generated by PLL, can anyone tell me how to **calculate**d its SNR?
In the frequency spectrum , I can not distingguish what's the signal and what's the noise because the signal spectrum is not a single line but with a lot of phase noise.
As a common method, in **fft** analysis, we often pick some bins beside the signal and add them toget

Analog Circuit Design :: 05.03.2006 22:20 :: sixth :: Replies: **2** :: Views: **1286**

The SNDR can be **calculate**d by the calculator in spectreRF. For the SNR, you need to exclude the distortion products shown in the spectrum. Remember, it is not the distance from signal peak to noise floor -- this distance is usually called SFDR (spurious free dynamic range). Only knowing the noise floor is not sufficient, you also need to know how m

Analog Circuit Design :: 09.04.2006 13:56 :: willyboy19 :: Replies: **12** :: Views: **10906**

/*
This computes an in-place complex-to-complex **fft**
x and y are the real and imaginary arrays of 2^m points.
dir = 1 gives forward transform
dir = -1 gives reverse transform
*/
short **fft**(short int dir,long m,double *x,double *y)
{
long n,i,i1,j,k,i2,l,l1,l2;
double c1,c2,tx,ty,t1,t2,u1,u2,z;
/* **calculate** (...)

Digital Signal Processing :: 05.05.2006 07:05 :: sanbaba :: Replies: **0** :: Views: **658**

Hi neoflash,
performing convolution using **fft** (so-called fast convolution) is faster than direct convolution.
If you have two sequencies of length N, the complexity of direct convolution is of the order of N^2.
With fast convolution, you have to perform two **fft**'s (direct and inverse), that have complexity of the order of N*log2(N), plus spectr

Digital Signal Processing :: 17.05.2006 18:19 :: zorro :: Replies: **2** :: Views: **1654**

Hi, all,
how to **calculate** the SNR and SNDR of the bit stream (1-bit stream) of sigma-delta modulator, and the SNR and SNDR of the output data of the all ADC. need your helps. Thanks in advance.
regards,

Analog IC Design and Layout :: 01.06.2006 02:49 :: wonbef :: Replies: **3** :: Views: **1540**

Hi,
Currently, im designing 8-point **fft**. From my view, we must **calculate** two values within the stages of **fft** which is block offset and stride. Can anyone explain to me what it means by stride and block offset.
Thanx

Digital Signal Processing :: 30.06.2006 14:54 :: siva_7517 :: Replies: **0** :: Views: **894**

Yes,u can. u will get the various frequency component of signal included in the text file by **fft**.
the program below will be helpful to you
clear
load xxx.txt;
outlen=length(xxx);
ain=xxx(1:outlen,2:2);
plot(ain)
or
clear
f=fopen('xxx.txt','r');
=fscanf(f,'%f',inf);
in1=rot90(in1);
iRet=fclose(f);
plot(in1)
to **calculate** **fft**

Analog Circuit Design :: 12.10.2006 05:10 :: heartwide :: Replies: **3** :: Views: **775**

how many sample number you take to **calculate** **fft**?
for a 12 bit ADC, you should take 2^13 sample at least.

Analog Circuit Design :: 04.11.2006 04:26 :: nathanee :: Replies: **6** :: Views: **1540**

As per my understanding:
FS: is used for signals which is continous and periodic..
FT: is used for signals which is continous and aperiodic...
DFT: is used for signals which is discrete and periodic.....
DTFT: is used for signals which is discrete and aperiodic.....
To **calculate** FT for a discrete signal is done by just sampling the conti

Digital Signal Processing :: 16.11.2006 00:02 :: text2babu :: Replies: **11** :: Views: **1306**

how to **calculate** DTFT of an analog signal using matlab?

Digital Signal Processing :: 19.12.2006 12:31 :: oualkadi :: Replies: **1** :: Views: **1429**

Hi,
What is the technique to **calculate** the frequency of an input signal which is at Ghz range.
for example system1's output is given input as sys2. If I want to know the fre of sys1's output what shud be the circuit to be included?
Please help me
And also I want to disply this frequency in an LCD display....
help!!!

Digital Signal Processing :: 09.01.2007 01:31 :: sivamit :: Replies: **2** :: Views: **567**

how i **calculate** the THD of sample and hold circuits in cadence tool
actually the normal procedure as the select the net and take special function of THD and then plot .
but this is ok with the continuous wave and any amp.
but here the input signal is getting sampled at every clock cycle.
so i need to **calculate** the THD this regards.
and

Analog IC Design and Layout :: 15.03.2007 03:03 :: manissri :: Replies: **5** :: Views: **1866**

I want to **calculate** the peiod of this sampling frequency is 369hz?
how can i **calculate** it's period?

Digital Signal Processing :: 28.09.2007 08:47 :: m_eh_62 :: Replies: **23** :: Views: **8725**

Dear naresh850,
I am a little confused: you wrote that you did the processing in time domain but then you wrote that velocity recognition is based on Goertzel algorithm. But this is equivalent to saying that the processing is in Frequency domain, because the Goertzel is a simplified method to **calculate** **fft** frequency bins.
Please could you clari

Digital Signal Processing :: 13.10.2007 03:50 :: mowgli :: Replies: **14** :: Views: **8981**

i m writing a program in matlab to make the **fft** analysis of the signal which the user determine
but i dont know how to measure the period of tha signal..
how to **calculate** the frequency or period of a signal which has two component with different frequencies
like A.sin(w1.t)+B.cos(w2.t)
plz help meee

Digital Signal Processing :: 01.12.2007 13:33 :: savage67 :: Replies: **3** :: Views: **1956**

I want to **calculate** SNR for sigma-delta modulator.I know that i will require to calculating **fft** but I don't know that how I can **calculate** SNR with **fft**.:cry:
thank you

Electromagnetic Design and Simulation :: 31.12.2007 09:02 :: royayeboodan :: Replies: **1** :: Views: **1488**

plz help me . refer the following program,
....main aim. i want to generate 2000hz frequncy and plotting time and frequncy spectrum for the same. i have got all the result but frequncy plot result is not correct.
% **calculate** data
n1=2000;
n2=500;
t=0:0.1:50
x = sin(2*pi*n1*t) + sin(2*pi*n2*t);
y = **fft**(x,512);
m = y.*conj(y)/512;
f

Software Problems, Hints and Reviews :: 20.02.2008 08:47 :: suru :: Replies: **1** :: Views: **683**

Digital communication :: 02.03.2008 11:54 :: cheggy :: Replies: **5** :: Views: **1330**

I appreciate your response about the Hilbert Transform!
When I select that in the Xilinx CORE generator and build it, it gives me the I and Q parts, as you mentioned, the I part is 16 bits , but the Q part is 35 bits. I'm not sure why. It seems, however, that I am able to adjust the output width of Q somewhat based on w

Digital Signal Processing :: 06.03.2008 01:28 :: abort :: Replies: **5** :: Views: **2291**

i am doing the final year project about class D amplfier , use some algorithm to produce PWM , then pass to the power amplifier and low pass filter to generate sound.
Since i finish to build the class D amplifier in fypga and also able to see the Power spectral of the generated PWM to measure the THD of the board. The only thing i need to do now

Electronic Elementary Questions :: 15.03.2008 01:00 :: clarken :: Replies: **3** :: Views: **2279**

I wondered whether this should be in the DSP section but it involves the Xilinx Core Generator.
After looking at the Core Generator **fft** module datasheet, I see that the **fft** core is expecting the data in complex format.
The data I want to transform is from an ADC and will be held in an array, the sampling will be constant rate.
Do I have to

PLD, SPLD, GAL, CPLD, FPGA Design :: 19.03.2008 17:01 :: Rob B :: Replies: **5** :: Views: **1739**

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 (...)

Digital Signal Processing :: 24.03.2008 02:09 :: triquent :: Replies: **4** :: Views: **6403**

I design a 3-order sigma delta modulator for audio application and want to achieve
16-bits resolution(96dB). For hspice simulation, I write **fft** code :
fs=5.12MHz BW=20k fin=fs*N/np , N is prime
.tran 1/fs 12779.62us
.**fft** v(y) start=100ns stop=12779.62us window=hann np=65536 freq=fin
freq: input frequency
start

Analog Circuit Design :: 22.04.2008 04:34 :: lunbaby :: Replies: **1** :: Views: **852**

In code composer studio you can use graphs by referring x y axis variables in your code.
1) Set sampling rate of ADC
2) Take samples and copy them into array.
3) **calculate** **fft**
4) Use Graph wizards in Code Composer Studio to display.
You can use TI's code support library all three steps. For more information refer help of CSL.
Regards,
ALPER US

Digital Signal Processing :: 25.11.2010 08:00 :: alperuslu :: Replies: **2** :: Views: **736**

yes, you can **calculate** the noise of opamp firstly, and then generate the random number with the variance from the opamp's noise. the thermal noise is Gauss distribution, so it can be **calculate**d by 3-σ rule.

Analog Circuit Design :: 24.06.2008 01:06 :: jiangxb :: Replies: **3** :: Views: **834**

Hello there, anyone knows how to write a matlab code to **calculate** the discrete fourier series and its inverse given then verify the code for the following input

Digital Signal Processing :: 04.08.2008 21:07 :: aredhel :: Replies: **4** :: Views: **6256**

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