TPV Tracking Phase Vocoder
resynthesis = TPV.ar(chain, windowsize=1024, hopsize=512, maxpeaks=80, currentpeaks, freqmult=1.0, tolerance=4, noisefloor= 0.01, mul=1.0, add=0.0)
Implementation of the McAulay and Quatieri 1986 sinusoidal model, as described in:
McAulay, R. and Quatieri, T. (1986) Speech analysis/Synthesis based on a sinusoidal representation. IEEE Transactions on Acoustics, Speech, and Signal Processing 34(4): 744--754
chain [fft] - Audio input to track, which has been pre-analysed by the FFT UGen; see examples below for the expected FFT size
windowsize- Window size used for FFT
hopsize- hop size for FFT, typically half window size
maxpeaks- Absolute maximum number of allowed peaks to be detected in the spectrum
currentpeaks- Current number of allowed peaks to be detected in the spectrum
freqmult- Resynthesis parameter to change frequency; currently causes a gross multiplication of frequency of all sinusoidal components, no envelope compensation
tolerance- Search area for matching peaks; within tolerance spectral bins
noisefloor- Minimum magnitude for a candidate peak
b.free
//assumes hop of half fftsize, fine
b = Buffer.alloc(s,1024,1); //for sampling rates 44100 and 48000
//b = Buffer.alloc(s,2048,1); //for sampling rates 88200 and 96000
//d=Buffer.read(s,"sounds/a11wlk01.wav");
d=Buffer.read(s,"sounds/break");
(
{
var in, fft, output;
in= SoundIn.ar(0); //PlayBuf.ar(1,d,BufRateScale.kr(d),1,0,1);
fft = FFT(b, in, wintype:1);
output=TPV.ar(fft, 1024, 512, 70,MouseX.kr(1,70), MouseY.kr(0.25,3),4,0.2);
//Out.ar(0,Pan2.ar(output));
output
}.play
)
(
{
var in, fft, output;
in= PlayBuf.ar(1,d,BufRateScale.kr(d),1,0,1);
fft = FFT(b, in, wintype:1);
output=TPV.ar(fft, 1024, 512, 50,50,1.0,MouseX.kr(0.1,100),MouseY.kr(-20,40).dbamp);
//Out.ar(0,Pan2.ar(output));
output
}.play
)
(
{
var in, fft, output;
in= PlayBuf.ar(1,d,BufRateScale.kr(d),1,0,1);
fft = FFT(b, in);
output=TPV.ar(fft, 1024, 512, 1,1);
output
//Out.ar(0,output);
}.plot(0.05, s, nil, -1.5, 1.5);
)
//residual
b = Buffer.alloc(s,1024,1); //for sampling rates 44100 and 48000
d=Buffer.read(s,"sounds/break");
(
{
var in, fft, output;
var input,inputAmp,threshhold,gate;
in= SoundIn.ar(0); //PlayBuf.ar(1,d,BufRateScale.kr(d),1,0,1);
//inputAmp = Amplitude.kr(input);
//threshhold = 0.001; // noise gating threshold
//gate = Lag.kr(inputAmp > threshhold, 0.01);
//in= (input * gate);
fft = FFT(b, in, wintype:1);
output=TPV.ar(fft, 1024, 512, 50,50,1.0,4,0.2);
//Out.ar(0,Pan2.ar(output));
//delay by 512 samples for output, then phase measurement is from centre of frame, so
output - DelayN.ar(in,0.1, (512+512)/44100)
}.play
)