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practical FFT

Started by Sharan123 in comp.dsp8 years ago 40 replies

Hello, I am having some questions related to FFT while using inbuilt Matlab or Octave FFT functions. These functions seem to take N samples...

Hello, I am having some questions related to FFT while using inbuilt Matlab or Octave FFT functions. These functions seem to take N samples of the signal that needs to be analyzed. Are there any requirement on the length of the input signal that needs to be fed into. A signal can be made up of many frequencies, so, it is obvious that a signal of N length can have any or all of the follow...


numerical precision of FFTs when interpolating

Started by WillGlover in comp.dsp15 years ago 12 replies

Hi all, I'm running into some problems when using FFTs to interpolate data from coarse grids to finer grids. The operation that I'm doing to...

Hi all, I'm running into some problems when using FFTs to interpolate data from coarse grids to finer grids. The operation that I'm doing to achieve this is: * Forward 3D FFT on coarse grid * Copy data to fine reciprocal space grid vector with zero padding * Backward 3D FFT The backward FFT is therefore of larger size than the forward FFT. Despite using double precision FFTs, I find tha...


Question about DSP in Matlab (beginner)

Started by JCO_DSP in comp.dsp9 years ago 97 replies

Hi, I'm starting to learn DSP (Matlab )and I have two questions about it, maybe someone can help me out? I'm trying to measure a frequency...

Hi, I'm starting to learn DSP (Matlab )and I have two questions about it, maybe someone can help me out? I'm trying to measure a frequency from an Audio Device and I'm using FFT to transform it from the time domain to the frequency domain. But I am doing a downsample before the FFT. But you can also do it after the FFT. Do you know what would be the most efficient (before or after)? And why?...


FFT computation

Started by Anonymous in comp.dsp18 years ago 23 replies

For starters, pardon my simplicity. That said, given - an array of sample data such that: data [ 8 ] = { 0., 0., 1., 0., 2., 0., 3., 0....

For starters, pardon my simplicity. That said, given - an array of sample data such that: data [ 8 ] = { 0., 0., 1., 0., 2., 0., 3., 0. }; If I ran an FFT on that via matlab - so now: a = [0+0i 1+0i 2+0i, 3+0i] b= fft( a ); The result: 6.000 -2.000 - 2.000i -2.000 -2.000 + 2.00i I'm going through a matlab couse and the individual who's teaching the course happens to ...


Help with FFT for DSP

Started by Diux in comp.dsp19 years ago 11 replies
FFT

I'm working with a motorola's DSP56824. But I need a library or algorithm to calculate de FFT (256 points and 1 Dimension). The problem is that...

I'm working with a motorola's DSP56824. But I need a library or algorithm to calculate de FFT (256 points and 1 Dimension). The problem is that I'm working with codewarrior(to programm with C), but I tried to compile some algorithms but they doesn't compile. Someone knows a link for a free FFT algorithm por DSP. Than You


Is it possible to decompose FFT to accept input in sampling time?

Started by Jimm Avante in comp.dsp17 years ago 5 replies

I am new to FFT but after reading, all FFT starts butterfly operation during the middle of the frame. For example, a radix-2, it starts until...

I am new to FFT but after reading, all FFT starts butterfly operation during the middle of the frame. For example, a radix-2, it starts until N/2 sample has been read in from ADC, i.e. but(0, N/2-1), but(1, N/2), ... Is it possible to design it in such a way it performs a butterfly operation real time every two samples? but(0, 1), but(2, 3), ...


signal power

Started by lucapp in comp.dsp16 years ago 14 replies

Hi all, I have the following problem: a snapshot of a signl s[n], n=0.....N-1. I compute signal power like this: P=1/n*sum(abs(s[n])^2)...

Hi all, I have the following problem: a snapshot of a signl s[n], n=0.....N-1. I compute signal power like this: P=1/n*sum(abs(s[n])^2) n=0,...,N-1. After this I make S=fft(s) and P'=sum(abs(S)^2). Missing out a normalization parameter (depending from fft definition) I have the same result. Suppose noe that I want to use a window function with fft like hanning (for example) my n...


FFT re-scaling or interpolation

Started by Evan Olcott in comp.dsp19 years ago 11 replies

Hello everyone. Have an interesting problem. I have an FFT of an audio signal and I want to interpolate the bins into another frequency...

Hello everyone. Have an interesting problem. I have an FFT of an audio signal and I want to interpolate the bins into another frequency set... instead of a linear set of frequencies, I'd like the frequencies I get (in the end) to be based on an exponential/logarithmic formula, based on the twelfth root of 2... What I'd like to do is take the FFT results and re-interpolate (or re-sc...


Understanding inverse FFT results

Started by dkuhta in comp.dsp19 years ago 13 replies

Hi, first off I'm fairly new to DSP. That said, here is my situation. 1.I have an input array of time domain samples. 2.I feed this array into...

Hi, first off I'm fairly new to DSP. That said, here is my situation. 1.I have an input array of time domain samples. 2.I feed this array into a FFT, then do an inverse FFT. 3.From the inverse results, I extract the "real" component. 4.I plug the iFFT real components into a new array. Here is output from my program looking at one data sample: time domain sample: -256.0 fft of sample: ...


FFT symmetry

Started by cherriegeller in comp.dsp15 years ago 3 replies
FFT

Hi, appreciate if someone can explain to me why is it when i do a FFT the output spectra is symmetrical? Is this to do with the formulae we used?...

Hi, appreciate if someone can explain to me why is it when i do a FFT the output spectra is symmetrical? Is this to do with the formulae we used? Thanks.


Calibrating FFT results, amplitude in to magnitude out

Started by Brian Willoughby in comp.dsp13 years ago 99 replies

Hello all, There is one factor that seems to be missing from texts which describe the DFT/FFT (or perhaps I have missed it). That is: The...

Hello all, There is one factor that seems to be missing from texts which describe the DFT/FFT (or perhaps I have missed it). That is: The correlation between time domain signal amplitude and frequency domain bin magnitudes. On the one hand, many DSP libraries are very meticulous about documenting the differences between their FFT or IFFT implementation results versus Matlab. For ex...


General FFT question

Started by Markus Grunwald in comp.dsp10 years ago 13 replies

Hello, while working daily with FFTs, I seem to have forgotten some of my theory :( Given: - the "usual" FFT algorithm that turns N...

Hello, while working daily with FFTs, I seem to have forgotten some of my theory :( Given: - the "usual" FFT algorithm that turns N complex samples in time domain to N complex samples in frequency domain. - a real valued input signal (= some sampled data, sampling rate "fa"). If a is the result of the FFT, there are two "interesting" elements of a: a[0] : This is the DC part ...


JAVA & FFT + Sound Processing & Latency in JAVA

Started by QWERTY in comp.dsp18 years ago 3 replies

Hello all! I'm new on this group. I started following it because I need help for project I'm working on at my faculty. I need to include...

Hello all! I'm new on this group. I started following it because I need help for project I'm working on at my faculty. I need to include an FFT algorithm in my project in order to manage sound signal. Can anyone recommend me a good implementation of FFT in JAVA, because, as I have read about it, there is many of implementations which are not optimised. Code examples about mnaging sou...


Mixed-Radix 2048 point FFT algorithm

Started by Hrachya in comp.dsp17 years ago

Hi all! For the 2048 point FFT implementation I have used two 1024 point Radix-4 (Decimation-in-frequency) FFT and for final stage I have used...

Hi all! For the 2048 point FFT implementation I have used two 1024 point Radix-4 (Decimation-in-frequency) FFT and for final stage I have used 2048 point (Decimation-in-time) Radix-2 FFT. But outputs are incorrect. When input is a pulse at x[i] = {1 when i = 1, 0 otherwise} the real part of output is cosine with frequency = 2, since I have expected the cosine with frequency = 1. Likewise fo...


FFT vs discrete convoultion

Started by westocl in comp.dsp16 years ago 9 replies

Im trying to get a feel for when Taking the FFT and doing frequency domain processing is more computationally efficient than doing...

Im trying to get a feel for when Taking the FFT and doing frequency domain processing is more computationally efficient than doing discrete convolution. Does anyone have any specific examples? I understand that very long FIR filtering may have a beneifit if filter is done via FFT then IFFT. Is most real time filtering done in the time domain?


FFT vs DCT

Started by Raeldor in comp.dsp13 years ago 36 replies

Hi All, I'm trying to understand the differences between FFT and DCT and was hoping someone may be able to help me with a couple of...

Hi All, I'm trying to understand the differences between FFT and DCT and was hoping someone may be able to help me with a couple of questions. 1. Is DCT basically the same as an FFT that operates on real numbers? 2. I keep reading about 'real-odd' and 'real-even'. What does this mean? 3. The fftw documentation also talks about being 'shifted by half a sample'. If by sample it's ta...


Correlation using FFT

Started by rg in comp.dsp20 years ago 13 replies

Dear All, Can someone describe the algorithm for performing correlation analysis between two signals using fft. I tried using the simple...

Dear All, Can someone describe the algorithm for performing correlation analysis between two signals using fft. I tried using the simple correlation algorithm but as my signals are quite long (upto 45 seconds of audio), as you can imagine, the calculation takes for ever. I know there is a quicker technique to do this by transforming the signals into frequency spectrum using fft, but I am...


Re: Integer FFT

Started by Sangamanath Kalashetty in comp.dsp12 years ago 3 replies
FFT

Hi all, I am trying to compute the FFT of signal using the integer FFT method. In this I am able to represent the twiddle factors in sum of...

Hi all, I am trying to compute the FFT of signal using the integer FFT method. In this I am able to represent the twiddle factors in sum of powers of two, to do the multiplication operations using the shift and addition operations. Example how to multiply 0.7071 with 0.255 using shift and addition operators only by representing the data in binary form. But the thing is I am not able to ...


How to fft huge data arrays?

Started by Andreas Besting in comp.dsp19 years ago 18 replies
FFT

Hi! For my little project i need to fft large data arrays, that do not fit into memory. So far i know about overlap add and for some problems...

Hi! For my little project i need to fft large data arrays, that do not fit into memory. So far i know about overlap add and for some problems that works fine, but I wondered if it's possible to transform the whole data altogether. I tried to code a special memory management for the fft but due to bit reversed addressing i get so many page faults that it's really not efficient... I a...


FFT with input in real time, is it possible?

Started by Mr. Ken in comp.dsp17 years ago 1 reply
FFT

I am new to FFT, but I found that if input is from ADC, the FFT can only start when half of the data is read in (Assume 1024 point, it...

I am new to FFT, but I found that if input is from ADC, the FFT can only start when half of the data is read in (Assume 1024 point, it starts when 511st is sampled, i.e. Butter(0, 511), Butter(1, 512), Butter(2, 513), Is it possible to decompose the sequence to allow Rad-2 butterfly operations as early as possible, say Butter(0, 1), Butter(2, 3), Butter(4, 5), .. ?