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I want to write an algorithm to add pitch-shifted version of audio signal to itself (like Electro-Harmonix POG does). And I need to do it in realtime. How to implement it?
My guess is: break signal into pieces, get FFT of each piece, copy each of the frequencies in the original signal (for example, for 1 octave up shift copy 20hz to 40hz, 40hz to 80hz and so on), get the iFFT, play it back. But I don't know if I got the right idea. Will it work?
okay, try googling "pitch shifting AMDF" or "pitch shifting TDHS" and see what you hit there. also check out DSP Dimension and maybe music-dsp and the mailing list.
you need three concurrent processes:
a good pitch detector that gives you a good estimate of the period (to a precision of a fraction of a sample). this period estimate should be updated as least as often as every 15 or 20 ms.
a real-time resampling and interpolation alg with a pointer that has precision at the fractional sample. this pointer will have both integer and fractional parts, the integer part tells you where in your delay buffer you are interpolating, the fractional part tells you how you're interpolating (like the interpolation coefficients). try linear interpolation at first, it's easier and you can improve it later. now this pointer will advance faster than the write pointer for upshifting and slower for downshifting. either way, that pointer will run into the edge of the delay buffer (and glitch or click or pop when it wraps around) unless you do something about it.
a splicing and crossfading alg that allows you to splice or seamlessly "jump" from one instantaneous pointer position in the delay buffer to another pointer position. the difference between those two pointer positions should be one period as determined by your pitch detector.
if i hint more than that, i might be spilling beans (internal trade secrets) from one manufacturer or another. but this is the main idea. think of splicing as necessary to time-scale the input (say time-expand if you're upshifting), and then step 2 is playing that time-expanded audio back faster which both returns it to the original tempo, but bumps it up in pitch.
@robertbristow-johnson give you nice tips, i can suggest you, PSOLA algorithm (aka Keith Lent1) to pitch shift in time domain.
PSOLA assumes that, if you know the pitch of a given segment, then you'll be able to remove or add periods moving synchronously and preserving envelope information.
The main step of PSOLA algorithm are:
The representation of these steps:
Mybe can be ease to see how some steps are build using a sinusoid example, so, here an real example, the picture bellow show an 150hz sampled at 22050hz (blue signal), on top of that sign you can see the pitch mark positions marked directly on the sign, the green dots on top are the pitch marks again and the green marks below are the synthesised marks using a pitch factor of 1.5, the red line between up/down dots tell you what centred hann whindow get to overlap and add.
So this steps pitch shift 150Hz signal to 1.5*150 = 225Hz.
1 - An efficient Method for Pitch Shifting Digitally Sampled Sounds", published in the Computer Music Journal, Vol 13, No. 4, Winter 1989
There are many ways to do pitch shift. here is sample way. You can do it by re sampling (up and down). As you know down sampling and up sampling , they make your data frequency change too.
You can use lib Sox.
