i have two audio signals .wav and i want to convolve them, the trouble is that the frequency do not fit, they aren´t just the same frecuency-lenght in the wavread arguments. i am supposed to use the lesser frequency to resample the other one using interp1 and splines. I really appreciate your help. Thanks

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    $\begingroup$ What are the sampling frequencies of the two files? $\endgroup$
    – Jim Clay
    Commented Mar 14, 2013 at 15:52
  • $\begingroup$ This is a question about MATLAB programming: how to use specific tools in a MATLAB toolbox, rather than a question about signal processing. It should be closed here or migrated to stackoverflow.SE or similar site. $\endgroup$ Commented Mar 14, 2013 at 17:48
  • $\begingroup$ I think it's a DSP question, but his tool of choice seems to be Matlab. The answer depends on the sampling rates though. $\endgroup$
    – Dave C
    Commented Mar 14, 2013 at 19:42

1 Answer 1


If the ratio of the sample rates is reasonably rational (i.e can be represented as the quotient of reasonably small integers) then you can use the MATLAB function resample().

Convolving to audio signals is rather unusual. Typically and audio signal is convolved with filter impulse response. The impulse response could be stored in an aduio file, but that would be a bit unusual. I would double check if you understood the task correctly.

Both interp1() and spline() are poor choices for sample rate conversion as they don't control the mirror images in any predictable way. I recommend reading up on "sample rate conversion".


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