FFT audio input

Question

I want to apply FFT on a signal recorded by AudioRecorder and saved to a wav file. the FFT i am using has a Complex[] input parameter. I am confused, is there a difference b

Answer 1

If you are working with audio with a bit depth of 16 bits (each sample has 16 bits), then each byte will only have half of a sample.What you need to do is cast your bytes to 16 bit samples then divide the resulting number by 32768 (This is the magnitude of the smallest number a 2's complement 16 bit number can store i.e 2^15) to get the actual audio sample which is a number between -1 and 1.You will then convert this number to a complex number by setting it's imaginary component to 0.

A small C# sample can be seen below (indicative code):

    byte[] myAudioBytes = readAudio();
    int numBytes = myAudioBytes.Length;

    var myAudioSamples = new List();

    for( int i = 0; i < numBytes; i = i + 2)
    {
      //Cast to 16 bit audio and then add sample
       short sample = (short) ((myAudioBytes[i] << 8 | myAudioBytes[i + 1]) / 32768 ); 
       myAudioSamples.Add(sample);
    }

    //Change real audio to Complex audio

    var complexAudio = new Complex[myAudioSamples.Length];

    int i = 0;
    foreach(short sample in myAudioSamples)
       complexAudio[i++] = new Complex(){ Real = sample, Imaginary = 0 };

   //Now you can proceed to getting the FFT of your Audio here

Hope the code has guided you on how you should handle your audio.