fft

问题 I have an assignment to implement a Ram-Lak filter, but nearly no information given on it (except look at fft, ifft, fftshift, ifftshift). I have a sinogram that I have to filter via Ram-Lak. Also the number of projections is given. I try to use the filter 1/4 if I == 0 (b^2)/(2*pi^2) * 0 if I even -1/(pi^2 * I^2) if I odd b seems to be the cut-off frequency, I has something to do with the sampling rate? Also it is said that the convolution of two functions is a simple multiplication in

问题 I have found for several times the following guidelines for getting the power spectrum of an audio signal: collect N samples, where N is a power of 2 apply a suitable window function to the samples, e.g. Hanning pass the windowed samples to an FFT routine - ideally you want a real-to-complex FFT but if all you have a is complex-to-complex FFT then pass 0 for all the imaginary input parts calculate the squared magnitude of your FFT output bins (re * re + im * im) (optional) calculate 10 *

问题 I have found for several times the following guidelines for getting the power spectrum of an audio signal: collect N samples, where N is a power of 2 apply a suitable window function to the samples, e.g. Hanning pass the windowed samples to an FFT routine - ideally you want a real-to-complex FFT but if all you have a is complex-to-complex FFT then pass 0 for all the imaginary input parts calculate the squared magnitude of your FFT output bins (re * re + im * im) (optional) calculate 10 *

问题 I've been reading a lot about Fast Fourier Transform and am trying to understand the low-level aspect of it. Unfortunately, Google and Wikipedia are not helping much at all.. and I have like 5 different algorithm books open that aren't helping much either. I'm trying to find the FFT of something simple like a vector [1,0,0,0]. Sure I could just plug it into Matlab but that won't help me understand what's going on underneath. Also, when I say I want to find the FFT of a vector, is that the

问题 I have convoluted an image I created in matlab with a 2D Gaussian function which I have also defined in matlab and now I am trying to deconvolve the resultant matrix to see if I get the 2D Gaussian function back using the fft2 and ifft2 commands. However the matrix I get as a result is incorrect (to my knowledge). Here is the code for what I have done thus far: % Code for input image (img) [300x300 array] N = 100; t = linspace(0,2*pi,50); r = (N-10)/2; circle = poly2mask(r*cos(t)+N/2+0.5, r

问题 For a FFT function I need to permutate or shuffle the elements within an array in a bit-reversed way. That's a common task with FFTs because most power of two sized FFT functions either expect or return their data in a bit-reversed way. E.g. assume that the array has 256 elements I'd like to swap each element with it's bit-reversed pattern. Here are two examples (in binary): Element 00000001b should be swapped with element 10000000b Element 00010111b should be swapped with element 11101000b

问题 First time here so sorry in advance for any butchered formatting. So I am completely new to DSP so I have only a very general understanding of the Fourier Transform. I am trying to build a visualizer app for Android SDK 9, which includes a Visualizer class in android.media.audiofx.Visualizer http://developer.android.com/reference/android/media/audiofx/Visualizer.html The javadoc for the method getFft(), which is what I am using states: "Returns a frequency capture of currently playing audio

问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 5 years ago . I'm trying to do real-time pitch detection using C++. I'm testing some code from performous (http://performous.org/), because everything else hasn't worked for me. I know for sure that this works, but i just cant get it to work. I've been trying this for a few weeks now, and I haven't been able to get any pitch

问题 I've been playing around a little with the Exocortex implementation of the FFT, but I'm having some problems. Whenever I modify the amplitudes of the frequency bins before calling the iFFT the resulting signal contains some clicks and pops, especially when low frequencies are present in the signal (like drums or basses). However, this does not happen if I attenuate all the bins by the same factor. Let me put an example of the output buffer of a 4-sample FFT: // Bin 0 (DC) FFTOut[0] = 0

问题 It's been asked a lot, but I still stuck about implement FFT class on Android I need to process my audio data using FFT... I already read the almost same question here How can I get frequency data from PCM using FFT and here How to get frequency from fft result? and more questions but still find no answer even after I tried the answers given... FFT Class I'm using: http://www.cs.princeton.edu/introcs/97data/FFT.java The complex class to go with it: http://introcs.cs.princeton.edu/java/97data