I would like to process 12 to 20 seconds of incoming audio at a sample rate of 44100. I must process this audio in real time in an STM embedded kit (perhaps also an Android Smartphone). I'm trying to detect and count the number of occurrences of a signal of roughly 6500 samples inside the incoming audio. The maximum FFT available is of 1024 samples.

I was thinking about applying overlap-add but the number of coefficients would be 6500 and that's larger than maximum FFT size of 1024. I tried to simulate this in Matlab using fftfilt but the function help says:

If you supply a value for n, fftfilt chooses an FFT length, nfft, of 2^nextpow2(n)and a data block length of nfft - length(b) + 1. If n is less than length(b), fftfilt sets n to length(b).

This makes me think that I'm forced to use an FFT of at least 6500 samples (which I can't) and then process 1 incoming audio sample at a time (super inefficient).

What can I do?

  • $\begingroup$ So you want to replace a matched filter (of length 6500 samples) detector with FFT (of length 1024) ? $\endgroup$
    – Fat32
    Commented Aug 2, 2018 at 18:32
  • 3
    $\begingroup$ Along with the technique pointed out by Stanley in his answer below, another option is partitioned convolution. Split your long filter into shorter sections and implement each as a separate filter. Then, delay and sum the filter outputs appropriately to reconstruct the response you would have gotten from using the long filter to begin with. $\endgroup$
    – Jason R
    Commented Aug 2, 2018 at 19:26
  • $\begingroup$ are you filtering with an FIR filter of 6500 taps? is that what you're doing. is this a matched filter problem? $\endgroup$ Commented Aug 2, 2018 at 21:40
  • $\begingroup$ @robertbristow-johnson Yes i'm sorry I didn't add matched filter tag as well! $\endgroup$
    – VMMF
    Commented Aug 2, 2018 at 21:53
  • 1
    $\begingroup$ @JasonR Nice idea! $\endgroup$
    – VMMF
    Commented Aug 2, 2018 at 22:22

1 Answer 1


You can make a big FFT out of smaller FFTs

This code implements a 16384 point FFT with a 16 point FFT and 1024 point FFT.

You need only calculate the Twiddle matrix once.

clear all
x=sin(linspace(1,M*N,M*N)*2*pi*60/(M*N));  % test signal
X=reshape(x,N,M).'; % form 2D matrix read data in as rows
Twiddle=zeros(size(X));    % make Twiddle matrix
for i=1:M
for k=1:N   
X=fft(X); % fft on each column
X=X.*Twiddle;% element by element product
X=fft(X.').' ; %fft on each row
y=reshape(X,N*M,1); % read out as columns
title('Composite DFT')
title('Direct DFT')

which is based on a section in

Rabiner, Lawrence R., and Bernard Gold. "Theory and application of digital signal processing." Englewood Cliffs, NJ, Prentice-Hall, Inc., 1975. 777 p. (1975).

I don't have the book handy for a page number, but the section is the table of contents under something like 1D DFT as a 2D FFT.


section 6.8 page 371 A unified approach to the fft

  • $\begingroup$ Chapter 6 deals with FFT and discusses various decompositions. Yet no exact page shows the algorthm alone. It's distributed across the sections. $\endgroup$
    – Fat32
    Commented Aug 2, 2018 at 21:34
  • $\begingroup$ I have just downloaded the book could you please name the section $\endgroup$
    – VMMF
    Commented Aug 2, 2018 at 21:55

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