thanks, the problem was the phasers that keep time in the sample,  i worked around it by passing the note beginning time as the sample location indicator 


On Tuesday, April 23, 2013 at 9:17 PM, joachim heintz wrote:

> perhaps you could try ptablei instead of tablei?
> 
> 
> Am 22.04.2013 01:42, schrieb zohar argaman:
> > i thought of using tablei warp function, but i need the length of the
> > table to be a specific not-power-of-2 size. and ftgen of GEN01 is giving
> > me an error when inputing a non 2^x size. maybe i can workaround by
> > writing the sample index by index, but this seems too much. will it
> > work? or maybe there is an easier way?
> > 
> > On Monday, April 22, 2013 at 8:10 AM, zohar argaman wrote:
> > 
> > > after figuring a mistake in ftgen normalisation (should be -1) I'm
> > > getting sound and i just need to get the phasor mechanism working.
> > > changing the value of the linsey beginning point do not work as
> > > expected, if i ask to begin at the value of 1 I'm getting a short
> > > squeeki version of my sample. this is obviously not suppose to happen
> > > when only one sample offset is set. any ideas?
> > > 
> > > On Monday, April 22, 2013 at 6:03 AM, zohar argaman wrote:
> > > 
> > > > 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:
> > > > <raw>,<CsoundSynthesizer>
> > > > <CsOptions>
> > > > </CsOptions>
> > > > <CsInstruments>
> > > > 
> > > > 
> > > > 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>
> > > > <CsScore>
> > > > 
> > > > f 1 0 16384 10 1
> > > > 
> > > > s
> > > > i "Init" 0 1
> > > > s
> > > > f0 70
> > > > i "Setup" 0 0.003
> > > > i10.50362820146.828242120
> > > > 
> > > > </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.
> 
> 
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