problem about fft-Collection of common programming errors
Mizuti
python fft gaussian
I’m trying to create a Gaussian random field, by creating a grid in Fourier space and then inverse Fourier transorming it to get the random field. For this, the inverse Fourier transformed image needs to be real valued. I seem to be getting residuals in the imaginary part of the grid of the order 10^-18 – -22, so I expected this to be numerical errors in the FFT. The real part of the image displays a weird checkerboard pattern on pixelscale though, where the pixels jump from positive to negative
123hal321
iphone objective-c audio accelerometer fft
I am currently using FFT / other spectral analysis algorithms in order to analyse sound from the iphone’s microphone. However, if the user taps the device while recording, I get unexpected results. Is there a filter I can implement on the raw sound data to remove the tap disturbances.Thank You,nononoedit: perhaps something to do with accelerometer data
Felix
java fft analysis spectrum dft
http://jvalentino2.tripod.com/dft/index.htmlMy code is really just a copy of the above:package it.vigtig.realtime.fourier;import java.io.File; import java.io.IOException;import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.AudioSystem; import javax.sound.sampled.DataLine; import javax.sound.sampled.LineUnavailableException; import javax.sound.sampled.SourceDataLine;public class Fourier {// Create a global buffer sizeprivate static final
Nfer.zhuang
fft speex
When I compile speex, the configure failed with below output:./configure: line 11964: syntax error near unexpected token `FFT,’ ./configure: line 11964: ` PKG_CHECK_MODULES(FFT, fftw3f)’I search a lot of things about the FFT and fftw3f, but I still don’t know how to resolve it. Could anyone give me a help? Thanks.
user2464424
python fft cmath
I got this python code for FFT calculation of a sound signal:from math import * from cmath import exp, pidef fft(x):N = len(x)if N <= 1: return xeven = fft(x[0::2])odd = fft(x[1::2])return ([even[k] + exp(-2j * pi * k / N) * odd[k] for k in xrange(N / 2)] + [even[k] – exp(-2j * pi * k / N) * odd[k] for k in xrange(N / 2)])N = 64res = [sin(k) for k in xrange(N)]# Window function a = 2*pi/(N-1) for k in xrange(N):z = a * res[k]res[k] = 0.42659 – 0.49656*cos(z) + 0.076849*cos(2*z)res = fft(res)
Trojan
frameworks signal-processing fft
I’m following an article on how to produce a cepstrum for use in detecting speech formants and coding it using the iPhone Accelerate framework. However, the results are not quite as the article expects. For unvoiced sections (figure 3 in the article) it shows smaller values in the first few bins. However, when my code runs, the unvoiced sections have large values (towards 1.0) which looks more like a voiced section.Here is my code:// copy buffer data into a seperate array and apply hamming windo
user1326876
fft frequency
This question is in two parts, both related in some way or another.Suppose I have a signal where the sample rate is 44100 and compute a normal 1D DFT on this signal. The size of this signal is 220500. I need to compute the Hertz for the outputted DFT.I calculate the bins using the formula below:sqrt(output[i].re * output[i]. re + output[i].im * output[i].im)I then find the maximum number in this vector and compute the following:freq = max * 44100 / fft.size() This then gives me a result of: 4526
Chuck Carlson
fft autocorrelation
I’m doing autocorrelation (via fft) on pure, synthetic generated sinusoids and getting unexpected jitter in the results. The jitter depend on where in the sine the signal buffer is started. I get the correct results if I start the buffer at 0 and 180 degrees of a sine wave, and get the most error at 90 and 270 degrees. I am padding with zeros after the signal buffer up to the next power of 2 above 2x-1 the signal buffer size.I’ve been using scipy fftconvolve, as well as doing the auto-correla
packoman
fft fourier-transform window-functions
I have the following problem, that I ran into recently, when calculating the spectra of data that I obtain from a measurement technique, we are using in our group.In short what we do in the technique, is record a movie of a fluctuating, nearly circular object and determine the contour in each frame of the movie. We then use the spectrum of these fluctuations to measure physical properties of the object.I have now refined the algorithm that we use for tracking the object, so that it is more preci
user1348051
matlab audio signal-processing fft
I am playing around with FFT and cant seem to get the expected results. I have a sound clip that i recorded from a mic placed on an analogue clock. So it has a sound every second.[wave,fs]=wavread(‘clock.wav’); t=0:1/fs:(length(wave)-1)/fs; plot(t,wave); n=length(wave)-1; f=0:fs/n:fs; wavefft=abs(fft(wave)); plot(f,wavefft);This is the code that i am using. And here are the results.I should be seeing a huge spike at 1hz but i am not. The FFT followed
endolith
frequency fft wavelet
The Fast Fourier Transform takes O(N log N) operations, while the Fast Wavelet Transform takes O(N). But what, specifically, does the FWT compute?Although they are often compared, it seems like the FFT and FWT are apples and oranges. As I understand it, it would be more appropriate to compare the STFT (FFTs of small chunks over time) with the complex Morlet WT, since they’re both time-frequency representations based on complex sinusoids (please correct me if I’m wrong). This is often shown wi
doelleri
c# fft naudio
I’m facing some problems on implementing FFTPitchDetector. What I actually want to do is to get real-time frequency from guitar input, I not so sure how to use the functions in FftPitchDetector.cs. Any idea?private void sourceStream_DataAvailable(object sender, NAudio.Wave.WaveInEventArgs e) { if (waveWriter == null) return;byte[] buffer = e.Buffer;float[] floats = new float[buffer.Length];float sample32 = 0;int bytesRecorded = e.BytesRecorded;waveWriter.Write(buffer, 0, bytesRecorded);for (i
Paul Ruane
c# signal-processing fft
I am trying to use the FFT library from Exocortex.DSP.The calling function is:Exocortex.DSP.Fourier.FFT(float[], int, Exocortex.DSP.FourierDirection)My line of code to use this function is:Fourier.FFT(floatArray, arraycount, FourierDirection.Forward);My program compiles, but why does it break at this line?**The breaking window says:An unhandled exception of type ‘System.MissingMethodException’ occurred in Exocortex.DSP.v1.dll Additional information: Method not found: Log System.Math.”*
Jagte
matlab fft physics
I have a time series which is a linear combination of damped waves. The data is real.Y(t) =SUM_w exp(- gamma t) sin(omega t)There is no analytic form but this is a closest guess. I want to fourier analyze (FFT) such data and get the real frequencies and damping rates.I am using matlab but any tool would be fine Thanks!
SolarLune
fft
I’ve been attempting to work on signal transformation and filtering for the past few months. It’s a bit difficult, but I’ve been making progress. I was wondering if anyone had any simple examples to show loading in a wave file, editing the volume and frequencies of that file, and re-outputting it to a file? I’ve already gotten up to volume, but I just can’t edit the frequencies.I thought that by using FFT, I would gain direct access to the frequencies of the sound file. Was that incorrect? I rec
Peter K.
fft discrete-signals
I’m trying to wrap my mind about what happens in Matlab/Octave when you compute the fft of a sine function. Theoretically, the Fourier Transform of a sine is given by two delta functions multiplied by the imaginary unit (one placed at the frequency of the sine and another at minus the frequency). I can see that in the modulus of the FFT (not exactly a delta because of the frequency leakage due to the windowing), but when I check the phase I get a signal that is non-zero over all the support (fro
Dave Moore
math fft frequency-analysis
I’m designing a real time Audio Analyser to be embedded on a FPGA chip. The finished system will read in a live audio stream and output frequency and amplitude pairs for the X most prevalent frequencies. I’ve managed to implement the FFT so far, but it’s current output is just the real and imaginary parts for each window, and what I want to know is, how do I convert this into the frequency and amplitude pairs?I’ve been doing some reading on the FFT, and I see how they can be turned into a magnit
digiphd
android fft dft libgdx
I have been searching everywhere to find a reliable method to calculate the FFT of an audio byte stream received by a native function in android SDK (through eclipse IDE). I have come across the libgdx fft and Jtransform. Jtransform Found here JTransform . I have downloaded them all and added the .jar files to a created libs folder in the root directory for the project. I have then linked the project to the new .jar files through project properties > java Build Path > Libraries. My src java file
Licson
php module fft
I wrote a FFT (Fast Fourier Transform) module for PHP recently. When I tried to test it, It always throws an error that the array $this->reverseTable has some index not defined. I got no clue of how to solve this problem.Here’s the PHP code:<?php class FourierTransform {public $bufferSize;public $sampleRate;public $bandwidth;public $spectrum = array();public $real = array();public $imag = array();public $peakBand = 0;public $peak = 0;public function __construct($bufferSize,$sampleRate){$th
Ben-Uri
matlab image-processing plot fft
I am using fft2 to compute the Fourier Transform of a grayscale image in MATLAB.What is the common way to plot the magnitude of the result?
Shai
matlab fft dicom data-type-conversion
I want to perform a 2 dimensional fourier transform on an image slice of a cranial MRI. I have tried the following codes and have not been successful: (error messages included below)>> clear all >> >> info = dicominfo(‘MR000026.dcm’); Y = dicomread(info); J=imadjust(Y,stretchlim(Y),[0 1]); F = fftshift(fft2(fftshift(J))); Undefined function ‘fftn’ for input arguments of type ‘int16’.Error in fft2 (line 19)f = fftn(x);>> info = dicominfo(‘MR000026.dcm’); Y = dicomread(inf
Carl Norum
c++ fft undefined-symbol
I am trying to implement the FFT in C++. However I am getting this error:Undefined symbols for architecture x86_64:”Complex::fft(std::vector<Complex*, std::allocator<Complex*> >*)”, referenced from:_main in ccMdeaRo.o ld: symbol(s) not found for architecture x86_64 collect2: ld returned 1 exit statusHere is my code:// Complex.h —————————————————————– #include <cmath> #include <string> #include <sstream> #include <vector&
Web site is in building
I discovery a place to host code、demo、 blog and websites.
Site access is fast but not money
