Hello, I have irregular spaced samples in 3 dimensions: f(xi, yi, zi) that I wish to take the FFT of. Therefore I need to resample the data to...

Hello, I have irregular spaced samples in 3 dimensions: f(xi, yi, zi) that I wish to take the FFT of. Therefore I need to resample the data to a regular grid. Since the data will be Fourier transformed; it will also be good to smooth the data somewhat to avoid Gibbs noise ("ringing")? Therefore I am considering using a convolution kernel to resample the data to a regular grid. A convolutio...

I've made an overlap-add/FFT convolution function, the problem with it is that it mostly outputs samples which value is infinite. I isolated the...

I've made an overlap-add/FFT convolution function, the problem with it is that it mostly outputs samples which value is infinite. I isolated the problem in the function and it appears to be at the level of my complex multiplication context : i have cut the long input signal into nblocks and put each part into an array block[], i of course zeropadded each block, then FFTed each block, and h...

Hi all, Suppose the forced input to the system is f(t), the step response of the system is a(t) and the output is y(t). Now we want to find...

Hi all, Suppose the forced input to the system is f(t), the step response of the system is a(t) and the output is y(t). Now we want to find y(t), I am confused: Which of the following is the correct output y(t)? (1) y(t)=convolution(differentiate(a(t)), f(t)) and (2) y(t)=differentiate(convolution(a(t), f(t)) ??? All "differentiate" and "convolve" operations...

Hello, I would like to apply an FIR filter (at least 512 points) to a 1D array of audio data (512 points). This convolution would take place...

Hello, I would like to apply an FIR filter (at least 512 points) to a 1D array of audio data (512 points). This convolution would take place within a audio callback function so the input and output data arrays both need to be the same length. My problem is (if I understand correctly) convolving the filter kernel and input data would result in an output signal N + M -1 long, i.e. different i...

Hi All, I'm doing some more research into digital filtering, and would like to present a poll to the DSP community. More specifically, I've...

Hi All, I'm doing some more research into digital filtering, and would like to present a poll to the DSP community. More specifically, I've read a lot about fast convolution/overlap add/save (OLA/OLS), and was wondering about real world applications of these techniques. From previous posts, it seems like the crossover point between FIRs and OLA/ OLS is anywhere from between 30 taps to 1k t...

Hello, I have a program that performs FFT convolution to search for occurrences of a small image inside another image. I feel like I...

Hello, I have a program that performs FFT convolution to search for occurrences of a small image inside another image. I feel like I understand the concept, but my program doesn't do what it should. The small search image is simply cut out from the larger image, so I should see a bright spike in the resultant FFT convolution. But I am seeing a very ambiguous result. I created a web...

Hi, I am starting to learn DSP. I have one fundemental question. I understood the sampling process if the input signal(Fin) is a sinewave...

Hi, I am starting to learn DSP. I have one fundemental question. I understood the sampling process if the input signal(Fin) is a sinewave and an impulse train is used as a clock(Fclk), it is mulitplication in time domain, which is convolution in frequency domain. The input signal is preserved becuase of the convolution with the clock signal at DC. (Fin+/-N*Fclk when N=0 we still have F...

i am currently implementing the 'Undecimated Wavelet Transform' with the 'a trous' algorithm. See...

i am currently implementing the 'Undecimated Wavelet Transform' with the 'a trous' algorithm. See e.g. http://www.znu.ac.ir/data/members/fazli_saeid/DIP/Paper/ISSUE2/04060954_2.pdf, section II-A. As far as I understand, the implementation (in 1D) for the 'decomposition' and 'reconstruction' phase is very simple, it boils down to the convolution with a low-pass and high-pass filter (where for le...

Hi all, I have recently implemented a fast convolution algorithm using the FFT/iFFT to convert input signal and filter into the frequency...

Hi all, I have recently implemented a fast convolution algorithm using the FFT/iFFT to convert input signal and filter into the frequency domain, multiply them out, and iFFT them back to the time domain to get my result. It all seems to work fine, however, I recently got a little confused on one point. I need to determine the gain that my filter will be applying to the signal. I've done thi...

Looking for a succinct example showing convolution processing (in the time domain) is notoriously computationally intensive; e.g., the...

Looking for a succinct example showing convolution processing (in the time domain) is notoriously computationally intensive; e.g., the typical reverberation time of a room is approximately 0.3 seconds which corresponds to 2400 samples, i.e., taps (filter coefficients), for an 8 kHz sampled sound. Because the sound is sampled at 8 kHz, the "delay steps" are each of length 1/8000 (0.000125 seconds)...

Hello People I want to compare Reed Solomon code [255,239] and rate 1/2 convolution code with constraint length 5, i.e. generator polynomials...

Hello People I want to compare Reed Solomon code [255,239] and rate 1/2 convolution code with constraint length 5, i.e. generator polynomials are [23 35]_8; Now, my channel is frequency selective, doppler effect and noise. Also the way I am scaling the noise is sigma=sqrt(0.5*Eb/(Rc*snr_lin)) where for RS, the Rc=239/255 and for convolution encoder Rc=1/2; Is this fair compari...

I have 2 serially connected multirate filters. One has 18 taps and the other has 61 taps. The decimation factors are 2, 2. Now I want to...

I have 2 serially connected multirate filters. One has 18 taps and the other has 61 taps. The decimation factors are 2, 2. Now I want to combine the 2 filters into one. The new coefficient file has 78 taps by using convolution. In FDAtool, the order is 137 after cascading. I think it is from 137=17+60*2 . So I guess I cannot just use convolution to combine 2 multirate filters. The problem is th...

Hello! I have been trying to implement an image in image search algorithm using fft, as described in the dspguide, chapter 24...

Hello! I have been trying to implement an image in image search algorithm using fft, as described in the dspguide, chapter 24 (http://www.dspguide.com/), using FFT Convolution. As far as I have understood it, the search image is used as some kind of a filter (like sharpen, etc.) on the original image (in which I search) in order to give the resulting image a distribution of similarity of...

Hi Folks, According to both [proakiscomm] and [bracewell], the convolution integral is defined to be y(t)= \int_{-\infty}^{+\infty}...

Hi Folks, According to both [proakiscomm] and [bracewell], the convolution integral is defined to be y(t)= \int_{-\infty}^{+\infty} x(\tau) g(t - \tau) d\tau. Notice that there is no dependence in the limits of integration on the independent variable "t." HOWEVER, in Proakis' second equation in deriving the matched filter, equation 5.1-15, which is describing the matched filter ou...

Hai, I have to write the response of convolution to a file as fixed point arithmetic values but the following MATLAB code generate...

Hai, I have to write the response of convolution to a file as fixed point arithmetic values but the following MATLAB code generate floating point arithmetic.How to get fixed point arithmetic? h=[-0.01711328125 0.016012451171875 -0.030242919921875 0.046417236328125]; Inp = zeros(300,1); Inp(1)=1; output = conv(h, Inp); outfile1 = fopen('imp_in.txt','w'); fprintf(outfile1, '%...

Hi there! I'm working on Pulse compression for radar signal using FFT, the echo is N points, and the length of match filter is also N points, then...

Hi there! I'm working on Pulse compression for radar signal using FFT, the echo is N points, and the length of match filter is also N points, then I use N points FFT to calculate the pulse compression result, it seems that I can indeed get the right result. I just get a little confused, since two N-points signals for linear convolution needs L(L> =2N-1) points FFT to get the right thing. Is there

hello forum. convolution between the function x(t), with units of Volts, and h(t), the system impulse response. h(t) needs to have units of...

hello forum. convolution between the function x(t), with units of Volts, and h(t), the system impulse response. h(t) needs to have units of inverse time, in order for g(t), the output of the convolution, to have units of Volts: g(t)=summation {x(tau)*h(t-tau)}dtau Correct? 2) If correct, take the signal exp(i*theta) theta has units of radians. If theta=(2*pi)*rand(N)-pi (unif noi...

Hello I am new to this forum so at first I want to say hello to everyone :) I am trying to make a fast fft convolution (FFT_Blocksize=1024...

Hello I am new to this forum so at first I want to say hello to everyone :) I am trying to make a fast fft convolution (FFT_Blocksize=1024 samples) of an headpone related impulse response (L=512 samples) with an sine wave audio signal. Here you can see the plots of the time Signals: impulse response: http://fs2.directupload.net/images/150617/fc9j6cs7.png audio signal block: http://fs1....

I am trying to determine the pdf of a function which is a convolution of two random variables. How do I determine the pdf of the function? Let...

I am trying to determine the pdf of a function which is a convolution of two random variables. How do I determine the pdf of the function? Let me illustrate with a simple case cited in many text books: Z=X+Y (sum of two random variables) The final pdf, fz(z)=fx(x)*fy(y) which is convolution of the individual pdf's, if they are independent. I am looking for the pdf fz(z) when, Z=X*Y (c...

I recently implemented fast convolution filtering for the "tools" portion of KISS FFT ( http://sourceforge.net/projects/kissfft ) I've seen a...

I recently implemented fast convolution filtering for the "tools" portion of KISS FFT ( http://sourceforge.net/projects/kissfft ) I've seen a few realizations of the overlap-scrap (aka overlap save) method. They all have the scrapped output samples at the beginning of the output buffer. Those realizations require an additional buffer copy to write continuous output. I found a way to...