Azzera filtri
Azzera filtri

Integral Of Audio Signal

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enrique128
enrique128 il 7 Nov 2020
Commentato: enrique128 il 8 Nov 2020
Hello everyone, I take my sound from microphone and my main purpose is calculating the Fourier Coefficients of it and coverge it.
But I need to calculate the integral of my audio signal because I need to calculate the coefficients. I tried xtrapz(x,y) and int functions but they don't meet my requirements. How can I calculate the integral of my sound? Thanks.
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
Error using trapz (line 15)
LENGTH(X) must equal the length of Y in dim 1.
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
Undefined function 'int' for input arguments of type 'double'.
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
Also I need some suggestions about the calculating coefficients, thanks.
My code:
recorder= audiorecorder(16000,8,2);
disp('please record your voice');
drawnow();
pause(1);
recordblocking(recorder, 5);
play(recorder);
data=getaudiodata(recorder);
x=0.2:pi/1000:2.2;
trapz(x,data)

Risposte (1)

Walter Roberson
Walter Roberson il 7 Nov 2020
Modificato: Walter Roberson il 7 Nov 2020
Fs = 16000;
trapz(data)/Fs
But if you need the fourier coefficients then just fft(data)
  4 Commenti
Walter Roberson
Walter Roberson il 7 Nov 2020
audiorecoder(16000,8,2) means to record with a sampling frequency of 16000 Hz, 8 bits per channel, 2 channels. You are getting two results because you are getting a result for each channel.
If you are calculating the coefficients for discrete fourier transform (fft) then you should not be integrating the signal, and you should be summing the signal instead, sum(data)
enrique128
enrique128 il 8 Nov 2020
sir im not calculating dft i will converge it with fourier series so i will not use fft. Is it possible?

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