Programmatically increase the pitch of an array of audio samples

Question

Hello kind people of the audio computing world,

I have an array of samples that respresent a recording. Let us say that it is 5 seconds at 44100Hz. How would I play

Answer 1

If you want this done easily, see AShelly's suggestion [edit: as a matter of fact, try it first anyway]. If you need good quality, you basically need a phase vocoder.

The very basic idea of a phase vocoder is to find the frequencies that the sound consists of, change those frequencies as needed and resynthesize the sound. So a brutal simplification would be:

1. run FFT
2. change all frequencies by a factor
3. run inverse FFT

If you're going to implement this yourself, you definitely should read a thorough explanation of how a phase vocoder works. The algorithm really needs many more considerations than the three-step simplification above.

Of course, ready-made implementations exist, but from the question I gather you want to do this yourself.