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
> 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.