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 > > > > ;--------------------------------------------------------------- > > ; 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 > > > > 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: > > , > > > > > > > > > > > > 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 > > > > > > > > > > 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 > > > > > > > > > > > > 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. >