what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain...

what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain filtering to FFT based filtering? bharat

what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain...

what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain filtering to FFT based filtering? bharat

Hello, I am trying to do an audio filter program. I need to filter out certain frequencies from the audio data. I am doing FFT, Bandpass &...

Hello, I am trying to do an audio filter program. I need to filter out certain frequencies from the audio data. I am doing FFT, Bandpass & IFFT. But after doing FFT, I am 0'ing out the bins for my desired frequency range. After that I am doing Inverse FFT. But the playing audio is not proper and a "tick tick" is comming in the audio output. I need to do the filtering for all PCM form...

Hi, Have anyone here tried the FFT-arm9e code from this book? ARM System Developer's Guide: Designing and Optimizing...

Hi, Have anyone here tried the FFT-arm9e code from this book? ARM System Developer's Guide: Designing and Optimizing System Software http://books.elsevier.com/companions/defaultindividual.asp?isbn=9781558608740 The book says the 1024p FFT takes about 67k cycles. But in my test it took 3 times as many cycles (I am using a slightly modified version with GNU assembler). And the precision...

Dear, I have decided to use dspic 30f4012 with 48k rom and 2k ram for implementing fft/ifft. My aim is to sample the incoming voice from a mic...

Dear, I have decided to use dspic 30f4012 with 48k rom and 2k ram for implementing fft/ifft. My aim is to sample the incoming voice from a mic to the adc module for 5sec and then perform fft. I also wanted to inverse the process to play back the voice, just to check the process. So please help me regarding the SAMPLING RATE, HOW TO USE THE LIBRARY FUNCTION IN DSPIC FOR MY DEVICE, WHAT SHOULD B...

Hi all, I posted up the question in an earlier post but have not received a clear answer yet. I would like to re-post my question in...

Hi all, I posted up the question in an earlier post but have not received a clear answer yet. I would like to re-post my question in this separated post. Please help. Normally, to normalize a signal s[n] with length N such that a real sinusoid signal x[n] with the same length N and amplitude of 1 will yield an FFT magnitude of unity, we simply divide the obtained FFT spectrum S[k] b...

Hi. I use FFTW for FFT analysis. Padded my data with zeros. Before padding my data, my fft looked like what is expected with some frequencies...

Hi. I use FFTW for FFT analysis. Padded my data with zeros. Before padding my data, my fft looked like what is expected with some frequencies showing stronger power than others. After I padded my data, the resulting plot showed a decreasing exponential trend interleaved with the actual expected data. I spent hours but couldnt figure out a reason except to conclude that there is something I'm m...

Quick (and dumb) question, but just wanted to make sure I understand. Supposed I have an input vector x of length N=500. And I also have a...

Quick (and dumb) question, but just wanted to make sure I understand. Supposed I have an input vector x of length N=500. And I also have a filter vector h of length M=3000. If peforming convolution filtering via FFT, what should I set the FFT sizes to be? I want the output signal to be the same size as the input signal (N) I assume it would need to be greater than N+M-1 to not cause circular c...

Hi, I have implemented cross-correlation using FFT's. Is it possible to to normalised cross-correlation with FFT's? If so, how? Sorry if it...

Hi, I have implemented cross-correlation using FFT's. Is it possible to to normalised cross-correlation with FFT's? If so, how? Sorry if it is a basic question - but I haven't found a solution. Thanks in advance..

I would like to get some feedback on this idea. Multiplying a signal in the time domain by a window before computing its FFT is equivalent to...

I would like to get some feedback on this idea. Multiplying a signal in the time domain by a window before computing its FFT is equivalent to the convolution of the transform of the signal with the transform of the window. It seems therefore that one could "undo" the effect of a square window (i.e. FFT with no windowing at all) by applying a complex deconvolution algorithm in the frequency d...

Hello forum, I can easily calculate the fft of a 1D Gaussian function, which is a real, even function. In theory, Its FFT should be real (zero...

Hello forum, I can easily calculate the fft of a 1D Gaussian function, which is a real, even function. In theory, Its FFT should be real (zero imaginary part). Instead the simulation shows an imaginary part definitely not zero and oscillating...Why? where is my mistake? Here my brief matlab code: mu=0; %mean sigma2=.2; %variance sd = sqrt(sig...

Hello again, I am still trying to implement a band-splitter scrambler based on the FFT. By band splitting I mean I take a frame of time, take...

Hello again, I am still trying to implement a band-splitter scrambler based on the FFT. By band splitting I mean I take a frame of time, take the FFT, shuffle some "bins"(bunch of frequencies) around and take the inverse FFT. If Im correct, this is essentially a time varying filter. My question is: What is the best overlap processing method to implement this? I have been using overlap...

Hi Guys, I am using FFT for an audio programming project I'm working on, and was wondering what are the bounds of the array values returned...

Hi Guys, I am using FFT for an audio programming project I'm working on, and was wondering what are the bounds of the array values returned from the FFT function (FFTW)? The input is an array of 16-bit integers representing amplitude between -16,384 and 16,383, with a sample rate of 22k. I am passing a sample length of 2048. At the moment the return values do seem to represent some kind...

Hi The FFT famous butterfly, involves the following calculations: (1) A' = A + B*W (2) B' = A - B*W I am going to implement a 32-bit...

Hi The FFT famous butterfly, involves the following calculations: (1) A' = A + B*W (2) B' = A - B*W I am going to implement a 32-bit FFT on a 16-bit machine, i.e. it can perform 16bit X 16bit -> 32bit. For doing that, a 32bit X 32bit -> 64bit multiplication needs to be performed by four 16bit X 16bit -> 32bit operations and then adding & shifting four products in order to have a

Hi The FFT famous butterfly, involves the following calculations: (1) A' = A + B*W (2) B' = A - B*W I am going to implement a 32-bit...

Hi The FFT famous butterfly, involves the following calculations: (1) A' = A + B*W (2) B' = A - B*W I am going to implement a 32-bit FFT on a 16-bit machine, i.e. it can perform 16bit X 16bit -> 32bit. For doing that, a 32bit X 32bit -> 64bit multiplication needs to be performed by four 16bit X 16bit -> 32bit operations and then adding & shifting four products in order to have a

Hi All, I was doing some experimenting and was wondering if someone could explain my results. I have a pure sine wave at 1000hz generated...

Hi All, I was doing some experimenting and was wondering if someone could explain my results. I have a pure sine wave at 1000hz generated into a wave file. I also have some FFT code to produce a spectrograph. If I run the wave file through the FFT, the spectrograph has one white line running through it (across time), which is what I would expect. HOWEVER, if I play the same sine wave an...

Hi all, is it correct that to normalize the output of fft such that a unit sinusoid in the time domain corresponds to unit amplitude in...

Hi all, is it correct that to normalize the output of fft such that a unit sinusoid in the time domain corresponds to unit amplitude in the frequency domain, I simply divide the DFT by the length of data. for example data is x(n), I take N point DFT so the normalized output is abs(fft(x,N))/N. Can you please tell me how to prove this? Thanks AELOVER

Hi, I know there are certain cases, that people implement a barrel shifter before actually taking FFT. My understanding was that this is done...

Hi, I know there are certain cases, that people implement a barrel shifter before actually taking FFT. My understanding was that this is done to preserve phase relation? Can anyone please explain more about it? I tried to find some useful article in this regard, but I was not able to find much about it. I would appreciate if someone explain the need for a barrel shifter before FFT? thank...

In another thread (Re: FIR Bandpass filtering (C code)) the OP was doing FFT of short segments of a much larger financial data stream and was...

In another thread (Re: FIR Bandpass filtering (C code)) the OP was doing FFT of short segments of a much larger financial data stream and was told FFT was inherently meaningless. I did not understand the stated logic. I'll pose another data set. GPS position data from single frequency receiver. Such data is known to have periodic components to the position errors. (eg diurnal variat...

Hi, I think my data is successfully getting transformed to the freq. domain by using the following FFT method. However, I'm a bit confused as to...

Hi, I think my data is successfully getting transformed to the freq. domain by using the following FFT method. However, I'm a bit confused as to how I can use the same method to do an inverse FFT. Any suggestions will be greatly appreciated. //bit reversal sorting private int bitrev(int j) { int j2; int j1 = j; int k = 0; for (int i = 1; i