CODEWITHSUNDEEP
matlabsignal-processingfftaudio-recording
Denis Davydov
Denis Davydov

Reputation: 462

Matlab Fourier transform of dsp.Audiorecorder in real-time

I'm using dsp.Audiorecord to get real-time microphone input. The sound input is a series of sinusoids with different frequencies ranging from 500 to 2000Hz. Each one sounds for a second.

I'd like to know in real-time, what's the frequency of the current sin and also make the difference between two sins with same frequency going one after the other. This is why I use dsp.Audiorecord.

This is what my code looks like now:

Microphone = dsp.AudioRecorder;
tic;
while(toc<30)
    audio = step(Microphone);
    [x, indexMax] = max(abs(fft(audio(:,1)-mean(audio(:,1)))));
    indexMax
end

All the indexMax shows are numbers ranging from around 25 to 40. There's clearly an operation left out in order to retrieve the original frequency in [500;2000].

I've tried also to apply dsp.FFT() directly to audio but it tells me:

Error using dsp.FFT/pvParse
Invalid property/value pair arguments.

If there's any other way to perform real-time FFT on the dsp.Audiorecorder I'd really like to know. Or just if you see a way to to complete what I've done here it would be great also.

Upvotes: 0

Views: 603

Answers (1)

hotpaw2
hotpaw2

Reputation: 70693

To approximately estimate what frequency goes with what index, you need to know the sample rate (Fs) of the data sent to the FFT, and the length (N) of the FFT: 

f ~= index * Fs / N

That's the operation you've left out.

Upvotes: 1

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