Hi guys,

first post for me. let me say, i love CSound!

Hans Mikelson posted on CSound magazine this interesting Time Domain TimeScale algorithm based on SOLA

http://www.csounds.com/ezine/spring2000/processing/

I love his explanation and the sound is good, but i needed a dynamic pitch shifting, that is, one that can change pitch in k-rate, i messed around with his code but i made more mess than i can cleanup, so i need your help.

this is Mikelson's version. very clear and neat

<raw>

;---------------------------------------------------------------

; Pitch and Time Scaling

;---------------------------------------------------------------

instr 11

idur = p3 ; Duration

iamp = p4 ; Amplitude ;1

ipshft = p5 ; Pitch scaling factor

itstr = p6 ; Time scaling factor

iaph = p7*sr ; Amplitude of the phasor in seconds converted to samples ;.1

itab = p8 ; Sample

ismth = p9 ; Smoothing factor ;2

ipval = (ipshft-1) ; Setup for pitch shifting

ifph = sr/iaph*((itstr-1)/itstr + ipval) ; Frequency for phasor

prints "P-Coef:%f ipval:%f, ifph:%f%R%R",ipshft,ipval,ifph

aphas1 phasor ifph ; Phasor 1

aphas2 phasor ifph, .5 ; Phasor 2 shifted by 180 degrees

apos linseg 0, idur, idur/itstr*sr ; Scan this many samples of the table.

kph1 downsamp aphas1

kph2 downsamp aphas2

printks "kph1:%f kph2:%f%R",0.1,kph1,kph2

kdclk1 oscil 1, ifph, 1 ; Declick envelope matches phasor frequency

kdclk2 oscil 1, ifph, 1, .5 ; Another one shifted by 180 degrees

kdclk1 = (tanh(kdclk1*ismth)+1)*.5 ; Flatten the sine waves and offset

kdclk2 = (tanh(kdclk2*ismth)+1)*.5 ; Same for other stream

ashft1 table3 iaph*aphas1+apos, itab ; Scan the table with cubic interpolation

ashft2 table3 iaph*aphas2+apos, itab ; second stream

aout = ashft1*kdclk1 + ashft2*kdclk2 ; Combine the two streams with declicking

outs aout*iamp, aout*iamp ; Output the result

endin</raw>

now, how would you make a auto-tuner out of this?

youll have to have a pitch detection algorithm (PDA) and the complicated thing is i want to scale a sample that runs in a loop, so... this is my code with event flow at the bottom:

</CsOptions>

sr =44100

ksmps = 64

nchnls = 2

;vocal sample

#define Filename #vocal_sample.aif#

instr Init

;Calculate important global vars and load FT

giproperLen filelen "$Filename"

giftsnd ftgen 0, 0, 2^(inbase), 1, "$Filename", 0, 0, 1

giftlen =ftlen(giftsnd)

endin

;---------------------------------------------------------------

; Pitch and Time Scaling - modified to play loop and support initial samp offset

;TIME STRETCH - DISABLED

;---------------------------------------------------------------

instr SOLA

idur = p3 ; Duration

iamp = p4 ; Amplitude ;1

inote = p5 ; note- convert to Pitch scaling factor

itstr = p6 ; Time scaling factor

iaph = p7*sr ; Amplitude of the phasor in seconds converted to samples ;.1

itab = p8 ; Sample

ismth = p9 ; Smoothing factor ;2

ilenSamp = p11 ;length of loop in samples

knote =k(p5)

;scale factor

kpshft = knote/gkAMDFcps

reinit PhsDeter ;extract phase from phasor and store it while a note is played

PhsDeter:

iSampOffset = int(i(gkeepPhs)*sr)

rireturn

printks "knote:%d kpshft:%f iSampOffset%f gkPhs:%f%R",0.1,knote,kpshft,iSampOffset,gkeepPhs

kpval = (kpshft-1) ; Setup for pitch shifting

gkfph = sr/iaph*(kpval) ; Frequency for phasor

aphas1 phasor gkfph ; Phasor 1

aphas2 phasor gkfph, .5 ; Phasor 2 shifted by 180 degrees

apos linseg 0, idur, idur*sr ; Scan this many samples of the table.

gashft1 table3 (iaph*aphas1+apos+iSampOffset)%ilenSamp, itab ; Scan the table with cubic interpolation

gashft2 table3 (iaph*aphas2+apos+iSampOffset)%ilenSamp, itab ; second stream

krms rms gashft1

outvalue "rms", krms

endin

;Notes - No Live Input, Orchestrated :(

instr 1

ivel = p5

a1 subinstr "SOLA" ,1 ,p4,1,0.1,giftsnd,2,1,giproperLen*sr

endin

;delay vocal until sample will hit the beat

instr Setup

event "i", "Process",0 , 65

turnoff

endin

instr Process

;Read loop

a1 lposcil 0dbfs, 1,0,giproperLen*sr, giftsnd

;Start Phasor to keep track with pitch, reset every sample Length in seconds

gkeepPhs phasor 1/giproperLen

;AMDF PDA - send global to instr SOLA

gkAMDFcps, krms pitchamdf a1, 50, 700 ,130

;fade mini-samples of SOLA

kdclk1 oscil 1, gkfph, 1 ; Declick envelope matches phasor frequency

kdclk2 oscil 1, gkfph, 1, .5 ; Another one shifted by 180 degrees

iamp=1

ismth=2

kdclk1 = (tanh(kdclk1*ismth)+1)*.5 ; Flatten the sine waves and offset

kdclk2 = (tanh(kdclk2*ismth)+1)*.5 ; Same for other stream

aout = gashft1*kdclk1 + gashft2*kdclk2 ; Combine the two streams with declicking

outs aout*iamp, aout*iamp ; Output the result

endin

</CsInstruments>

f 1 0 16384 10 1

i "Init" 0 1

f0 70

i "Setup" 0 0.003

i1 0.503628 20 146.828242 120

</CsScore>

</CsoundSynthesizer>

</raw>

it became kind of global variable - p-rate nightmare ;/

the Init method initialize any global variable so the compiler wont be angry.

setup initializes the Process instrument that runs as the Always On instr while performance.

Pseudo Midi given by the note in i1.

i1 trigger the SOLA instr.

the SOLA instr takes global params from Process like the pitch of the current k-frame, and creates global signal gashft1 that is picked up by Process and mixes with the fade mechanism and heads out.

the original sound is a loop, since its beeing read by a table3 opcode, i used a phasor to keep track of where is the pointer of the sample should be on the table. the thing is, that phasor must stop running when activating a note on SOLA, so the reinit section was added with hope to solve this.

now, what is the problem? simple, no sound!

i tried measuring rms of the ga coming out of SOLA, it measured 0.2 RMS. that's wierd. im i reading the table wrong? what can cause this behavior? how is best to debug this? what tools to use? is this idea possible anyway?

thanks to all readers and answerers.