Hi, For an experiment i have a simple application that takes a quadrature signal from a radio and then does the following: Complex FFT -> ...

Hi, For an experiment i have a simple application that takes a quadrature signal from a radio and then does the following: Complex FFT -> Inverse FFT -> Cartesian to Polar -> audio out If I tune the radio to an AM signal I can demodulate it perfectly well by this but it depends on how the FFT is processed. In the FFT routine, I apply a window and if it is anything other than a rectangular wind

Hello all, i have searched the internet for information about which of the two overlap methods is better but couldnt find any info. i am...

Hello all, i have searched the internet for information about which of the two overlap methods is better but couldnt find any info. i am implementing Decimation In Time fft in matlab. the entire process is: 1. audio sampling using wavread 2. computing fft of SEGMENTS of input audio samples 3. Computing fft of filter 4. Multiplying the two above fft's and taking inverse fft to get t...

Hello DSP pros, I'm currently working on an optimized version of FFT for ARM NEON. (FHT, to be exact.) I'm neither a DSP specialist nor...

Hello DSP pros, I'm currently working on an optimized version of FFT for ARM NEON. (FHT, to be exact.) I'm neither a DSP specialist nor mathematician, and it was very hard for me to understand the way FFT works. Now it's about 50% done, and I have my doubt about this : "What's the use of this? What does it bring when my FFT runs 20times faster than some unnamed opensource ones on handheld...

Hi, When realising a FFT, can we get the frequency corresponding to each point calculated? Apparently, the results given by a FFT is only...

Hi, When realising a FFT, can we get the frequency corresponding to each point calculated? Apparently, the results given by a FFT is only the [Re,Im] of each point? If we can't know the frequency by a FFT, how can we do it? The thing i'd like to do is detecting 2 different frequencies from the same input signal. Can it be possible? Thanks.

Hello all, I am curious why such a system does NOT work. I am using a 16-bit fixed-point DSP chip: voice1 -> ADC -> DSP -> DAC -> ...

Hello all, I am curious why such a system does NOT work. I am using a 16-bit fixed-point DSP chip: voice1 -> ADC -> DSP -> DAC -> voice2 I collect an array of samples and pass them through an FFT then an inverse FFT. The sample size is 256 but obviously, that can be changed. The FFT routines are verified to be "correct". Yet, when doing and FFT -> IFFT, the voice coming out is very

Hi Friends, I am new to this. My questions is very basic. How do i decide that i need to use 1024 point FFT or 2048 point FFT or N point FFT...

Hi Friends, I am new to this. My questions is very basic. How do i decide that i need to use 1024 point FFT or 2048 point FFT or N point FFT for a system. I am working on a project, where my sampling frequency is 80mhz and i need to decide how many point FFT to be use? What are the parameters i need to take into considerations to decide the No. of points in FFT? Can anyone please help ...

I read about something like the following before, but don't know what search terms to feed Google in order to find it. If I have only the FFT...

I read about something like the following before, but don't know what search terms to feed Google in order to find it. If I have only the FFT results, length N, for 2 successive frames of data, naively I can get an FFT of length 2N by first doing two IFFT's and then an FFT on the concatenation of their output. Is there a more efficient method? IMHO. YMMV. -- Ron Nicholson rhn AT ...

I have implemented Jens Joergen Nielsen FFT code in my windows program that converts V(t) to V(f). I am using 2500 Hz sampling with N=8192...

I have implemented Jens Joergen Nielsen FFT code in my windows program that converts V(t) to V(f). I am using 2500 Hz sampling with N=8192 point FFT. I am plotting the FFT amplitude as sqrt(Re^2 + Im^2)/N. Im getting the correct peak frequency after FFT, however, I am getting only about 1/2 of the peak amplitude (of the input sine wave input amplitude) with rectangular window. (If my input wav...

I have implemented Jens Joergen Nielsen FFT code in my windows program that converts V(t) to V(f). I am using 2500 Hz sampling with N=8192...

I have implemented Jens Joergen Nielsen FFT code in my windows program that converts V(t) to V(f). I am using 2500 Hz sampling with N=8192 point FFT. I am plotting the FFT amplitude as sqrt(Re^2 + Im^2)/N. Im getting the correct peak frequency after FFT, however, I am getting only about 1/2 of the peak amplitude (of the input sine wave input amplitude) with rectangular window. (If my input wav...

Hi, I'm using FFT convolution for smoothing. Basically FFT kernel is exp(-x^2) and the signal is something positive. All of it is done in a...

Hi, I'm using FFT convolution for smoothing. Basically FFT kernel is exp(-x^2) and the signal is something positive. All of it is done in a single pass, so no need for overlap-add and stuff like that. Seems working fine, except now I tried it on quite long data - the FFT length was 262144, kernel being 32768 samples (zero-padded to 262144), calculated in 32-bit floating points, FFT implemen...

Could someone please refer to good, non-matematician-accessible discussion of decimation-in-time vs decimation-in-freq FFT's? Second, more...

Could someone please refer to good, non-matematician-accessible discussion of decimation-in-time vs decimation-in-freq FFT's? Second, more confused, question. The implementations I find online are d.in-time, but I'd like a d.in-freq implementation if available (i think). The idea is to build a tree of partial FFT's: While making a 512-pt fft I want to save the intermediate 16 32-point fft'...

Hi, We all know the FFT-based cross-correlation in cartesian coordinate is given by: C = IFFT[ FFT[ f(x,y) ] * Conj[ FFT[ h(x,y) ]...

Hi, We all know the FFT-based cross-correlation in cartesian coordinate is given by: C = IFFT[ FFT[ f(x,y) ] * Conj[ FFT[ h(x,y) ] ] Here is the question: how is it possible to do this in polar? I do not think if this is the answer: C = IFFT[ FFT[ f(r,p) ] * Conj[ FFT[ h(r,p) ] ] where f(r,p) and h(r,p) are the polar transformation of f(x,y) and h(x,y).

Hi. I am implementing some FFT software and I'm pondering different ways of averaging FFTs.. Normally most people average the amplitude of...

Hi. I am implementing some FFT software and I'm pondering different ways of averaging FFTs.. Normally most people average the amplitude of each FFT to reduce the variance of the noise. It so happens that in my application we are sampling synchronously with the signal we are interested in (and it's centre-cell) so the phase of the desired signal between each FFT should be the same - this...

Hi, I am looking for an FFT algorithm that can give me a 4096 pt. FFT using two 64 pt. FFTs. Is it the mixed radix FFT? Thank you.

Hi, I am looking for an FFT algorithm that can give me a 4096 pt. FFT using two 64 pt. FFTs. Is it the mixed radix FFT? Thank you.

Dear All, I have a simple question regarding FFT based filtering. Most text tells us to do the FFT based filtering is multiplication of "FFT of...

Dear All, I have a simple question regarding FFT based filtering. Most text tells us to do the FFT based filtering is multiplication of "FFT of signal" and "FFT of FIR filter kernel." I understand that. My question is, in some simple situation, can we directly manipulate the frequency result of signal. For example, if do a low pass filtering, I obtained FFT of a signal, then just simply set ...

I am new to DSP and FFT. I have implememented the FFT in Smith's book. What I am a little confused about is how do I determine what is...

I am new to DSP and FFT. I have implememented the FFT in Smith's book. What I am a little confused about is how do I determine what is the frequency scale of the X axis? thanks Don

Hi there, Just a quick question from a DSP beginner here. I'm doing a college project on basic speech recognition, and I have some questions. I...

Hi there, Just a quick question from a DSP beginner here. I'm doing a college project on basic speech recognition, and I have some questions. I am told to use a 256 point FFT with a hanning window and 50% overlapping. This then produces a number of FFT windows right? (for example, in a 512 sample sound file I will have 3 windows). Then I have to perform an FFT on them. I then need to apply a...

Hello, When combining two smaller FFTs to make a larger one, what are the techniques. Example..I want to combine two 64 pt FFTs into a 128 pt...

Hello, When combining two smaller FFTs to make a larger one, what are the techniques. Example..I want to combine two 64 pt FFTs into a 128 pt FFT. I have fed the first 64 data samples into the first FFT and the second 64 into the 2nd FFT. The results are output from the two FFTs into a 128 array. The problem is that the 128 pt FFT result has the same peaks as the 64 pt FFT. What is the problem ...

Hello Comp.DSP, I need to perform forward and inverse FFT of multi-dimensional, real data. I am already successful to do this using the complex...

Hello Comp.DSP, I need to perform forward and inverse FFT of multi-dimensional, real data. I am already successful to do this using the complex FFT separably. Now I would like to do this using the real FFT for computational savings. However, after I perform real FFT of the first dimension, the data is complex and it seems I must use the complex FFT for subsequent dimensions. It makes s...

Hello, I am in academia so my knowledge on FFT are limited to what I find in basic DSP books. I have a question on scalable FFT. This is mainly...

Hello, I am in academia so my knowledge on FFT are limited to what I find in basic DSP books. I have a question on scalable FFT. This is mainly regarded to OFDMA where they supports different bin numbers, ranging from 128 to 2048. Now my question is how this is implemented in a FFT hardware module. I think different methods must be using by different manufactureres. But I am interested in a ...