I'm trying to perform Fast Fourier Transform on a .3gpp audio file. The file contains a small 5 second recording in 44100kHz from the phones microphone.
Every Java FFT algorithm I can find only takes double[], float[] or Complex[] inputs, for obvious reasons, but I'm reading in the audio file in a byte-array, so I'm kind of confused as to where I go from here. The only thing I could find is the answer to a previous question:
Android audio FFT to retrieve specific frequency magnitude using audiorecord
But I'm unsure as to wether or not this is the correct procedure. Anyone with any insight?
There is no alternative. You have to run a loop and cast each element of the array separately.
I do the same thing for shorts that I fft as floats:
public static float[] floatMe(short[] pcms) {
float[] floaters = new float[pcms.length];
for (int i = 0; i < pcms.length; i++) {
floaters[i] = pcms[i];
}
return floaters;
}
If you really do have 16 bit PCM but have it as a byte[], then you can do this:
public static short[] shortMe(byte[] bytes) {
short[] out = new short[bytes.length / 2]; // will drop last byte if odd number
ByteBuffer bb = ByteBuffer.wrap(bytes);
for (int i = 0; i < out.length; i++) {
out[i] = bb.getShort();
}
return out;
}
then
float[] pcmAsFloats = floatMe(shortMe(bytes));
Unless you are working with a weird and badly designed class that gave you the byte array in the first place, the designers of that class should have packed the bytes to be consistent with the way Java converts bytes (2 at a time) to shorts.
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