I am a novice in Fortran programming but I have started using code::blocks with Fortran plugin. I am trying to pass variables between my code (f03) and fftsg.f (code from http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html) to calculate fft. fftsg.f contains a subroutine called rdft (real discrete Fourier transform) that I try to use to...

I am trying to implement a 2d phase correlation algorithm in R using a recipe from Wikipedia (http://en.wikipedia.org/wiki/Phase_correlation) in order to track the movement between 2 images. These images (frames) were captured with a camera shaking in the wind and the ultimate goal is to remove the shake in these...

Hello I want to calculate realtime FFT plot of my input. With the following code i create a record and calculation. The point is that the calculation is takes to much time to get i nice plot update. Fs = 44100; % sampling frequency in Hz T = 1/5; %...

I have an audio sample, which is sampled at 22k Hz and total samples = 660k , as the duration is 30 seconds. (22k*30 = 660k) When I plot the fft of the complete sample, I get a symmetric graph with 660k x values, and corresponding y values as shown:...

I've created an FFT class/object that takes signal stored in a 2D array and produces the subsequent FFT of its input, before printing it to a matplotlib graph. After a great deal of reading, I appreciate that due to windowing, the need to have an ideally 2^x number of points...

I was trying to generate signals into the raw way but I am getting the error below could somebody help me ? And yes I need it into the raw way and after I will generate the spectrum of it. Also, could someone gimme some hints of where could I...

I'm trying to use matplotlib widget SpanSelector to make find the frequencies of a plot at certain user-clicked intervals. I'm trying to do this using matplotlib widget SpanSelector but am unsure of how to do this. I tried modifying the example given http://matplotlib.org/examples/widgets/span_selector.html but it does not work and still...

I'm using the below code in Unreal Engine 4 to capture microphone input and get the resulting FFT. I'm having trouble calculating the frequency based on this data. I've tried finding the max amplitude and taking that as the frequency, but that doesn't seem to be correct. // Additional includes:...

I have some signals and I mix respectively add up to a larger signal, where each signal is located in different frequency regions. Now, I perform the FFT operation on the big signal with FFTW and cut the concrete FFT bins (where the signals are located) out. E.g. The big...

I'm trying to convert a simple matlab code doing FFT. The code goes like this: c = fft([y; zeros(N-M, 1)]); Values for N=256, M = 100. Here are the values of y (100 points) that i'm using to test: 0.504747377917058 0.897277682272301 0.151672365249603 0.355122573738706 0.553925412440538 0.158834350258486 0.356559717393412 0.267749785433509 0.739084139656738 0.166499777727771 0.473321012784302...

I am very confused of how to use the output of FFD , here is a example output generate by result = np.abs(fftpack.fft(targetArray))[0:sample_size/2] print result will gives a ndarray : [ 4.21477326e+05 3.03444591e+04 1.80682694e+04 1.05924224e+04 1.98799134e+04 2.82620258e+04 1.39538936e+04 2.40331051e+04 4.57465472e+04 6.41241587e+04 1.88203479e+04 1.88670469e+04 5.42137198e+03 5.97675834e+03 1.23986101e+04 9.70841580e+02 2.07380817e+04 4.52632516e+04 4.49493295e+04...

I am using cufft to calculate 1D fft along each row for a matrix, and an array. The matrix size is 512 (x) X 720 (y), and the size of the array is 512 X 1. Which means the fft is applied on each row that has 512 elements for...

Here is my code for generating a triangular waveform in the time domain and for generating its corresponding fourier series/transform (I don't know whether its series or transform because matlab only has fourier transform function but since the signal is periodic, references say that the fourier counterpart must be called...

Can you help me find out why one of the FFTW's plans gives zeroes at the end of an output array? The "fftw_plan_dft_1d" yields proper result as I checked it with Matlab. The Real to Complex plan "fftw_plan_dft_r2c_1d" makes some zeroes at the end. I don't understand why. Here is...

I would like to work with percentages while doing some FFT with the web audio API. To do so I need to know the range of the values the analyser.getByteFrequencyData returns. I can't find anything about that, maybe someone knows? Thanks...

I have an issue using the FFTPACK5.1 in Fortran 90 which contains subroutines to compute discrete Fourier transforms. I manage to install it and use the routines but when I'm checking if everything is ok with a simple sine wave with a frequency A I get a non zero coefficient...

I am currently running Python's Numpy fft on 44100Hz audio samples which gives me a working frequency range of 0Hz - 22050Hz (thanks Nyquist). Once I use fft on those time domain values, I have 128 points in my fft spectrum giving me 172Hz for each frequency bin size. I...

I'm working on Matlab, I want to perform FFT on a wav file I previously recorded on Matlab as well. fs = 44100; % Hz t = 0:1/fs:1; % seconds f = 600; % Hz y = sin(2.*pi.*f.*t); audiowrite('600freq.wav',y,fs) This is the way I'm recording in the wav file. Now...

I am a university student. I am developing a music identification system for my final year project. According to the "Robust Audio Fingerprint Extraction Algorithm Based on 2-D Chroma" research paper, the following functions should need to be included in my system. Capture Audio Signal ----> Framing Window (hanning window)...

I have some binary array. For example, let my array is: int a[] = {1, 0, 0, 0, 1, 0, 1, 0, 1} I want to calculate the values based on this formula: How to calculate this function, using a fast Fourier transform? I have a large array and I...

I get how the DFT via correlation works, and use that as a basis for understanding the results of the FFT. If I have a discrete signal that was sampled at 44.1kHz, then that means if I were to take 1s of data, I would have 44,100 samples. In order...

I am trying to use the following code for finding FFT of a given list. After a lot of trials I have found that this code runs only for an input list having 2^m or 2^m+1 elements. Can you please clarify why this is so and whether it can be...

I'm trying to plot a Fourier series that should fit the original graph (which is right), but I don't know what's wrong. I also double-checked the Fourier approximation. The original graph is generated with: t=-pi:0.01:0; x=ones(size(t)); plot(t,x) axis([-3*pi 3*pi -1 4]) hold on t=0:0.01:pi; y=cos(t); plot(t,y) whereas the Fourier series...

I have noisy data (peaks with period 1.8s, 2048 bins per period) for which I want to calculate frequency and delete 50Hz. I'm pretty sure that frequency what I looking for is 50Hz, cause I find it by use originlab. When I try to do the same in python the...

I'm using MATLAB's fit function: fourier_series=(x,y,'fourier8'); to fit an 8th order Fourier series to a set of discrete data (x,y). I need the period of the Fourier series to be 2*pi. However I can't work out how to fix this so that when I call the function it fits the...

Im trying to understand how this FFT algorithm works. http://rosettacode.org/wiki/Fast_Fourier_transform#Scala def _fft(cSeq: Seq[Complex], direction: Complex, scalar: Int): Seq[Complex] = { if (cSeq.length == 1) { return cSeq } val n = cSeq.length assume(n % 2 == 0, "The Cooley-Tukey FFT algorithm only works when the length of the input is...

I am taking 32bit float audio(44.1Khz) on my PC(between -1 and +1) using Port Audio and taking fft of it with fftw. Now I need to take the 16bit int Audio and take its fft. I have converted the Audio samples to float between -1 and +1. The fft works...

The following is the code I have written to find the DFT of sine(x) over a period. program fftw_test implicit none INTEGER FFTW_MEASURE PARAMETER (FFTW_MEASURE=0) INTEGER FFTW_ESTIMATE PARAMETER (FFTW_ESTIMATE=64) INTEGER FFTW_FORWARD PARAMETER (FFTW_FORWARD=-1) integer, parameter :: n = 8 integer :: i double complex, dimension(0:n-1) :: input, output double precision,...

I have noisy data for which I want to calculate frequency and amplitude. The samples were collected every 1/100th sec. From trends, I believe frequency to be ~ 0.3 When I use numpy fft module, I end up getting very high frequency (36.32 /sec) which is clearly not correct. I...

My question is related to the explanation here by A. Levy: Analyze audio using Fast Fourier Transform How can I produce a bandpass filter on these complex numbers... [-636.00000000 +0.00000000e+00j -47.84161618 -2.80509841e+02j 30.69754505 -1.30624718e+01j -109.94022791 +7.58155488e+00j -3.18538186 +1.44880882e+01j -120.36687555 +5.45225425e+00j 50.48671763 +1.69504204e+01j 31.56751791 -7.22728042e+01j -17.96079093 -3.17853727e+01j -19.25527276 +5.08151876e+00j 18.38143611 -2.60879726e+01j...

I'm working in C with Dev-C++ I've created a 2D array of complex numbers as such: #include<complex.h> double complex **x; x = malloc(Nx * sizeof *X); if (x) { for (i = 0; i < Nx; i++) { x[i] = malloc(Nx * sizeof *x[i]); } And filled it with data,...

I have a lot of csv files with data. I want to perform the same action on all my files, but I do not know how to do this without doing it individually for all the files. I read in all the csv files and saved the data in my...

I'm doing a real-to-complex FFT with the org.apache.commons.math3.transform library as following: private Complex[] fft(double[] values) { FastFourierTransformer ffTransformer = new FastFourierTransformer(DftNormalization.STANDARD); Complex[] result = ffTransformer.transform(values, TransformType.FORWARD); return result; } This gives me a org.apache.commons.math3.complex array with the result. This works fine. Now I want to perform exactly the same with...

My goal is to compare the FFT of similar signals. For some reason, when I take the magnitude spectrum of two signals of the same length, the frequencies are different... I can't do a simple side by side comparison of two signals because of this. Anyone have any tips on...

I have a signal ts which has rougly mean 40 and applied fft on that with code ts = array([25, 40, 30, 40, 29, 48, 36, 32, 34, 38, 15, 33, 40, 32, 41, 25, 37,49, 41, 35, 23, 22, 36, 44, 28, 36, 32, 37, 39, 51]) index =...

I am trying to make a frequency spectrum up to 30 Hz of a Sine wave with period pi. I wrote a code but I keep getting the error : Undefined function 'fft' for input arguments of type 'sym' sint = sin(t); Tmax = 2*pi; %the end sample value Ts...

The real inverse FFT gives me an array full of NaNs instead of floats. kiss_fftri(conf,complex_array,output); The complex_array is normal, nothing wrong with the values i guess. kiss_fftr_cfg conf = kiss_fftr_alloc(size,1,NULL,NULL); The conf should be fine too as far as I know. Anything wrong with the size? I know that the...

I am trying to filter some data based on the the following code using Arduino FFT library for FFT (fast Fourier transform) /* fft_adc_serial.pde guest openmusiclabs.com 7.7.14 example sketch for testing the fft library. it takes in data on ADC0 (Analog0) and processes them with the fft. the data is...

Be warned, this is a newbie question. I acquired some noisy data (a 1x200 pixel sclice from a grayscale image), for which I am trying to build a simple FFT low-pass filter. I do understand the general principle of the Fourier Transform, but I ran into trouble trying to implement...

i'm kinda new to python and i had problem getting this to work, so since the deadline is for tomorrow, might as well ask the question here. I have two lists of float values, one for time and other for voltage values taken from an oscilloscope(i assume). I have to...

I am a programmer but quit bad on math.. I recently read an article which mentioned frequency-domain entropy,he calculated that thing from a FFT power spectrum but didn't tell me how to do that. I cannot find enough information online to understand what it is . I'm appreciate for any...

I am trying to make a music visualizer in Processing, not that that part is super important, and I'm using a fast fourier transform through Minim. It's working perfectly (reading the data), but there is a large spike on the left (bass) end. What's the best way to 'level' this...

To interpolate a signal in frequency domain, one can pad zeros in time domain and do an FFT. Suppose the number of elements in a given vector X is N and Y is the same as X but padded one sided with N zeros. Then the following give the same...

I got the following syntax error while I want to plot values: syntax error >>> plot(freq1, abs(fft1/max(fft1)),xlabel('f(Hz)'), ylabel('Amplitude I(f)'); ^ My definitions are as follows: a=x+y+z; % a is a sinus mixture of different curves/functions n1 = fa/0.05; % N is 50 ms fft1=fft(a,n1); freq1 = [0:deltaF1:fa-fft1]; plot(freq1, abs(fft1/max(fft1)),xlabel('f(Hz)'), ylabel('Amplitude...

since i don't have sinc function in my MATLAB, I implemented that function as shown below %% time specificactions: Fs=10000; dt=1/Fs; t=(-0.1:dt:0.1-dt)'; N=size(t,1); %message signal mt=(sin(pi*100*t))./(pi*100*t); %% frequency specifications dF=Fs/N; f=-Fs/2:dF:Fs/2-dF; M=fftshift(fft(mt)); plot(f,abs(M)/N); but the figure shows me nothing but blank, so i looked up the variable table and it...

I'm trying to compare two large sets of wav files to remove duplicates. The issue is that one set is PCM, the other has been u-law'd. When I try to read in PCM wav, no problem, but the u-law files give the following error: >>> wav = wave.open("C:\\soundfiles\\Olympus Recordings\\1019.wav") Traceback...

I have a simple one-dimensional array like [0,0,0,0,0,1,1,1,1,1,0,0,0,0,0] which describes a square impulse. I would like to transform this impulse to the frequency domain and plot its magnitude spectrum by using the code below (I got it from OpenCV Python Tutorials): squareimpulse = np.array([0,0,0,0,0,1,1,1,1,1,0,0,0,0,0]) img = (squareimpulse) f = np.fft.fft(img)...

I implemented a simple low pass filter in matlab using a forward and backward fft. It works in principle, but the minimum and maximum values differ from the original. signal = data; %% fourier spectrum % number of elements in fft NFFT = 1024; % fft of data Y =...

I need to perform two-point correlation function from astroML Python module, my data is originally a jpg image, black and white, and I convert it to binary image using OpenCV image thresholding(not sure that I did it right). The question is how now I convert the 2D binary matrix or...

I have a set of experimental data s(t) which consists of a vector (with 81 points as a function of time t). From the physics, this is the result of the convolution of the system response e(t) with a probe p(t), which is a Gaussian (actually a laser pulse). In...

I'm trying to trigger an event from sound input pitch, I have this code https://webaudiodemos.appspot.com/pitchdetect/index.html but I'm searching for an easier way to de that. Please help me, thank's by advance. ...

I have a method in Objective-C that receives an array of doubles and then it uses the Fast Fourier Transform, however the exit of the FFT doesn't match to what I want. Can someone help me, I don't know what I'm doing wrong? This is my method where fftLength is...

I have a temporal signal and I calculate its Fourier Transform to get the frequencial signal. According to Parseval's theorem, the two signals have the same energy. I successfully demonstrate it with Python. However, when I calculate the inverse Fourier Transform of the frequencial signal, the energy is no longer...

I have a python code, which imports 4 column txt file with numbers first three columns are x,y,z coordinate and fourth column is a density at that coordinate. below is the code that reads, converts to ndarray, Fourier transform that field, calculate the distance from origin (k=(0,0,0)) and a transformed...

let us consider following Page : http://djj.ee.ntu.edu.tw/S_Transform.pdf paragraph 2.3 The Discrete S Transform let say that we have sampled version of signal x, and given sampling frequency fs,i have calculated discrete Fourier transform using following code function y=DFT(x); N=length(x); D=zeros(N,N); for k=1:N for n=1: N D(k,n)=exp((-j*(k-1)*2*pi*(n-1))/N); end end y=D*x'/N; end...

I'm having two zero padded signals in Matlab h_1[n] = {...,0,0,1,2,1,0,0,...} h_2[n] = {...,0,1,0,2,0,1,0,...} and below you can see their FFT plots: % N1 and N2 are just the lengths of h1 and h2. H1 = fft(h1, N1); H2 = fft(h2, N2); % ... figure; from = -floor(length(H1)/2); to =...

I was trying to implement a FFT-based multiplication algorithm in M2(R). Basically an algorithm that gets as an input two polynoms with elements given as matrices, and builds the product polynom. However, even though the algorithm should work, as it looks exactly identical to a version I wrote earlier on...

Im working on spectrum analysis of wav file. I have plotting the spectrum of the whole frequency,but how can i plot just the high frequency of my file ? this is the code : [a,fs] = wavread('ori1.wav'); ydft = fft(a); ydft = ydft(1:length(a)/2+1); freq = 0:fs/length(a):fs/2; plot(freq,abs(ydft)); ...

Good evening guys, I wanna ask you a question regarding the analysis of a function in the domain of frequencies (Fourier). I have two vectors: one containing 7700 values for pressure, and the other one containing 7700 values (same number) for time. For example, I call the firt vector "a"...

I am totally new to signal processing and am trying to make a program that shows the amplitude of low frequency signals in a PCM (WAV) file. So far, I've been able to read in the WAV file and populate an array (actually a multi-dimensional array, one for each channel,...