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2D FFTW Inverse FFT does not give original image when /n

Started by adam3914 in comp.dsp15 years ago 4 replies

I have what I have to assume is a stupid mistake on my part. I am able to take the fft of an image and get back the half of the transformed image...

I have what I have to assume is a stupid mistake on my part. I am able to take the fft of an image and get back the half of the transformed image as I should. When I try to do a inverse fft on the data from the fft I get really big numbers like 10^71 size numbers. When I would expect to get my original image times 640*480, the size of my image. My first question is do I have to create the othe...


How to zoom into a certain part of FFT?

Started by Vista in comp.dsp17 years ago 13 replies

Hi all, Suppose I have a signal f(t), t is in [0, +infinity). And I have its spectrum F(w). Let's say I found out that its main spectrum...

Hi all, Suppose I have a signal f(t), t is in [0, +infinity). And I have its spectrum F(w). Let's say I found out that its main spectrum has 99.9% in [-B, B]. So I truncate/extract out the portion of F(w), for w in [-B, B], and discretized the interval into small grids with step size deltaB. And I then do the inverse FFT on the above samples of F(w), let's call the inverse FFT r...


stft vs fft resolution

Started by Adam Chapman in comp.dsp15 years ago 9 replies

Does a short-time fourier transform have greater frequency resolution that a normal fft? the code...

Does a short-time fourier transform have greater frequency resolution that a normal fft? the code at http://ccrma.stanford.edu/~jos/sasp/STFT_Matlab.html would suggest not since it's only using a normal fft on a frame of data rather than the whole dataset.


Applying a custom filter to an FFT for IQ imbalance

Started by tomb18 in comp.dsp9 years ago 2 replies

think I am missing something here and I do not quite understand why. Background: I am collecting quadrature signals from a software...

think I am missing something here and I do not quite understand why. Background: I am collecting quadrature signals from a software defined radio. There are two signals, the I, and the Q which is 90 degrees out of phase to the I. These are passed into a complex FFT and the results plotted as Power vs frequency. This all works fine. However, the nature of the signal gives a two sided FFT. In other...


FFT output bias

Started by tinkerz in comp.dsp13 years ago 13 replies

Dear All, I put the data from the output of FFT into a power spectrum, and there is a bias in the data, changing sample speed and number or...

Dear All, I put the data from the output of FFT into a power spectrum, and there is a bias in the data, changing sample speed and number or samples there always seems to be a ratio between low and high frequencies in the data. I am not sure if this is correct with FFT, I expected to see changes independently on the low and high frequencies. Should I try some sort of filter pre FFT? The...


Wiener filter with smoothed estimation of Noise

Started by stefanb in comp.dsp17 years ago 2 replies

Hello all, I implemented a blockwise Wiener Filter with Matlab like this: H = a_priori_SNR./(a_priori_SNR+1) a_priori_SNR = S./N; S =...

Hello all, I implemented a blockwise Wiener Filter with Matlab like this: H = a_priori_SNR./(a_priori_SNR+1) a_priori_SNR = S./N; S = magnitude of fft of clean signal s; S= abs(fft(s)), of current block N = magnitude of fft of noise n; N= abs(fft(n)), of current block Y = magnitude of fft of noisy signal y; Y= abs(fft(y)), of current block Since in real life I have neither the...


Newbie Seeking Help: Question about the FFT -> Imaginary Part

Started by angusyoung1000 in comp.dsp19 years ago 6 replies
FFT

I've been reading up on how to perform an FFT. I do not understand why I am to set the imaginary part of the calculation to zero, however. Here...

I've been reading up on how to perform an FFT. I do not understand why I am to set the imaginary part of the calculation to zero, however. Here is my resource: http://www.relisoft.com/Science/Physics/fft.html See step 3 about mid-way down the page. Is this correct? If so, please explain to me why this is to be done. Thanks, AY


Question - Calculation of PSD from FFT

Started by keerthi.madyasth in comp.dsp15 years ago 2 replies

Hii How to calculate PSD from FFT?? From the FFT I have got Real and imaginary parts and i would like to know how to calculate the PSD without...

Hii How to calculate PSD from FFT?? From the FFT I have got Real and imaginary parts and i would like to know how to calculate the PSD without using correlation function. If possible i would like to know the steps involved.


Any method to FFT/DFT a long sequence

Started by m26k9 in comp.dsp15 years ago 13 replies

Hello, I was wondering if there is any method for performing a FFT/DFT on a large number of samples? For example if I have a signal with 256...

Hello, I was wondering if there is any method for performing a FFT/DFT on a large number of samples? For example if I have a signal with 256 samples but my DSP only supports 128 samples input, is there an efficient way to get the complete sequence FFT/DFT'ed? I am looking for something like the dual of the 'overlap-save/add' method for convolution, where you can use FFT to convolve a large s...


FPGA/DSP Expert - business partner for innovative FFT

Started by Seung in comp.dsp21 years ago 15 replies
FFT

Hello I have a patent and recently added one more on innovative FFT algorithm and architecture. If you're a business minded expert on FPGA...

Hello I have a patent and recently added one more on innovative FFT algorithm and architecture. If you're a business minded expert on FPGA with interests in DSP, this is a great opportunity. Our FFT is 'the' optimal HW solution as follows: 1. Minimum HW complexity: 100% HW utilization 2. Suitable for super fast pipelined FFT: only local data flow - not based on butterfly algorithm 3. ...


Some questions about radix-4 FFT algorithm using fixed-point arithmetic

Started by pallav in comp.dsp14 years ago 6 replies

Hello all, I'm very new to signal processing and want to learn to implement some basic FFT algorithms in hardware. First, I coded up the...

Hello all, I'm very new to signal processing and want to learn to implement some basic FFT algorithms in hardware. First, I coded up the radix-4 Cooley- Tukey FFT algorithm in Matlab and it is working fine (compared it against the Matlab's builtin FFT) when I use floating point number to represent my signal. For hardware implementation, the input data is a complex (a + jb). The real/ima...


Clipped before FFT

Started by Robert Scott in comp.dsp11 years ago 28 replies

I have been experimenting with some time-domain pre-processing to improve the reliability of an FFT application. The application is extracting...

I have been experimenting with some time-domain pre-processing to improve the reliability of an FFT application. The application is extracting the RPM of an engine from the sound it makes. Prior to these experiements I was using straight FFT of blocks of time-series sound data. An engine generally makes a sound rich in harmonics, so the FFT should have evenly spaced peaks. When the peaks ...


Signal Complexity effect on FFT Execution Time

Started by Anonymous in comp.dsp16 years ago 4 replies

I'm currently using the kiss FFT library in 32-bit fixed point mode in some software. I'm always using the same length FFT e.g. 1024-point. I...

I'm currently using the kiss FFT library in 32-bit fixed point mode in some software. I'm always using the same length FFT e.g. 1024-point. I want to know whether the average execution time of the FFT algorithm is likely to be dependent on the input signal waveform complexity. i.e. would you expect it to run faster for a purely DC signal, versus a complex arbitrary waveform? Thanks in ad...


PSD Calculation

Started by Saumyajit in comp.dsp20 years ago 2 replies

HI All, I am an engg student doing a project on FFT. Now I am in the process of learning DSP.I am stuck with a problem related to the...

HI All, I am an engg student doing a project on FFT. Now I am in the process of learning DSP.I am stuck with a problem related to the calculation of PSD. Could anybody please help me out ?? I have got a 16 point FFT Processor.Now after FFT Processiing I do have 16 complex o/ps which are in the frequency domain.I have to calculate the PSD over these points. How do i do that ? I should not u...


Purpose of Detrending signal before FFT?

Started by MBALOVER in comp.dsp11 years ago 16 replies
FFT

Hi all, In code of an algorithm I see that the author often detrends signal before FFT. I search for it in the Internet and the objective of...

Hi all, In code of an algorithm I see that the author often detrends signal before FFT. I search for it in the Internet and the objective of the function is "detrend removes the mean value or linear trend from a vector or matrix, usually for FFT processing". Can you explain what is the effect of detrend and what if the data is not detrended before FTT? Thank you.


Newbie FFT normalization question

Started by Yofnik in comp.dsp17 years ago 9 replies
FFT

Hi All, I am a software engineer, not a DSP guru. I have been using an open source FFT algorithm for a project I am working on. As I have seen...

Hi All, I am a software engineer, not a DSP guru. I have been using an open source FFT algorithm for a project I am working on. As I have seen in a number of examples, in order to get the results I expect, I need to "normalize" the FFT results by dividing each value by N/2 (number of samples / 2). Can someone please give a little background as to why this is necessary? I am not satisfied ...


Real time FFT?

Started by westocl in comp.dsp15 years ago 19 replies

Hello all. Is it feasable to do an fft in 'real time' on different 'blocks' of data? (i understand the nomenclature real time may be...

Hello all. Is it feasable to do an fft in 'real time' on different 'blocks' of data? (i understand the nomenclature real time may be inappropriate). Im thinking, in general usage of an FFT, some x[n] comes in one sample at a time and fills up a buffer and when that buffer (lets call it 128 time domain samples) is filled a FFT is taken and the 'output' is 128 points of frequncy domain...


FFT Chirp Problems MATLAB

Started by kyletae in comp.dsp18 years ago 4 replies

I am doing an FFT of a chirp function. I am having problems with realizing the FFT! The FFT should when p =1 look like the signal in. It...

I am doing an FFT of a chirp function. I am having problems with realizing the FFT! The FFT should when p =1 look like the signal in. It doesn't Any suggestions?! Thanks... The following is the m-code: function s = dtFMchirp(p,TW) % DCHIRP generate a sampled chirp signal % usage s = dchirp(TW,p) % s : samples of a digital "chirp" signal % exp(j(W/T)pi*t^2) -T/2


Phase of FFT compared to phase of Sinusoid

Started by Anonymous in comp.dsp18 years ago 87 replies

Hi everyone, I'm generating a sinusoid of the form cos(wt + phi), then taking the FFT. I was expecting to find the phase of the FFT (at the...

Hi everyone, I'm generating a sinusoid of the form cos(wt + phi), then taking the FFT. I was expecting to find the phase of the FFT (at the frequency peak) allows me to determine 'phi' in the generating sinusoid, but there appears to be little correlation! For example, plotting cos(2 Pi x/10) from x=0 to 2million (with step 1) , the phase at the FFT is 2.82?!! Surely the phase shou...


Partial FFT... Possible?

Started by hq9000 in comp.dsp15 years ago 16 replies
FFT

Hello people, Standard FFT lets us retrieve amplitudes of evenly distributed frequencies. What if I need at some frequency range...

Hello people, Standard FFT lets us retrieve amplitudes of evenly distributed frequencies. What if I need at some frequency range greater/smaller density of frequency points... Let's say I take fft of 1024 sample long chunk. After standard FFT it will give amplitudes of f[i]=i*samplRate/1024 the step is sampleRate/1024 now let's say I want to have step equal to sampleRate/(1024*4) fo...