Sampler Sampler ------------ Created by Mark Van Peteghem This instrument plays audio files, and allows you to alter them. It plays both mono and stereo files (no need to specify this) and takes the sample rate of the sample in account. There are three play modes: play once, loop, and cycle. The first two should be obvious, the third means that the sample will be used as the cycle that is repeated at audio rate, at the frequency given in p5 (pitch or frequency). The first row of UI controls has the basic stuff: distortion of the table (done once to optimize, so not automatable), reversing, selecting begin and end point to be played, begin and end speed up, stutter, optional attack and release, panning and volume adjustment. Stutter length relates to the position in the original sample, so the actual stutter length may be shorter or longer due to the speedup settings. The second row of UI controls has the more advanced stuff: - Tune adjusts the pitch of the sample, either with pvscale or with delays (based on code by Victor Lazzarini). If the pvscale method is used, the preservation of formants can be selected. The amount of pitch adjustment can be selected with the knob in octaves (when 'Octaves' is selected) or by specifying the wanted fequency in p5, if the pitch of the original sample is given in the text box (when 'Pitch in p5' is selected). Note that the perceived pitch is also affected by the speedup knobs. - Shift lets you use pvshift on the sample. - Blur lets you use pvsblur on the sample. - Smooth lets you use pvsmooth on the sample. - Downsample lets you lower the sample frequency. - Auto delay is an experimental feature that uses the sample itself to add a delay. This works best with samples that only have low frequencies, like kick drums. - Chorus adds a chorus, optionally distributing the different voices to the left and right channel. - The filter settings should be obvious. If the play mode is cycle, the speedup, stutter and tune settings are ignored. Some knobs have a text field under them. These can be used to alter the value of the knob, e.g. entering +p6 when the value of the knob is 1, the used value will be 1+p6. gi_sampler_sample1_<INSTR_NAME> = 0 gi_sampler_sample1_dist_<INSTR_NAME> = 0 gi_sampler_sample2_<INSTR_NAME> = 0 gi_sampler_sample2_dist_<INSTR_NAME> = 0 gi_sampler_sr_<INSTR_NAME> = 0 gi_sampler_nr_channels_<INSTR_NAME> = 0 gi_sampler_duration_<INSTR_NAME> = 0 gi_cosine ftgen 0, 0, 65536, 11, 1, 1 gi_Hamming ftgen 0, 0, 2^16, 20, 1, 1 iamp = ((p4 > 0) ? 0dbfs*p4 : ampdbfs(p4))*ampdb(<vol_adjust>) if gi_sampler_sample1_dist_<INSTR_NAME>==0 then gi_sampler_nr_channels_<INSTR_NAME> filenchnls "<file>" gi_sampler_sample1_<INSTR_NAME>, gi_sampler_duration_<INSTR_NAME>, gi_sampler_sr_<INSTR_NAME> mvpReadSampleFromFile "<file>", 1 gi_sampler_sample1_dist_<INSTR_NAME> mvpDistortWaveTable gi_sampler_sample1_<INSTR_NAME>, <distort_type>, <distort_amount> if (<reverse>==1) then itmp mvpReverseTable gi_sampler_sample1_dist_<INSTR_NAME>, gi_sampler_duration_<INSTR_NAME>*gi_sampler_sr_<INSTR_NAME> endif istutter_reset1 table 0, gi_sampler_sample1_<INSTR_NAME> if (gi_sampler_nr_channels_<INSTR_NAME> > 1) then gi_sampler_sample2_<INSTR_NAME>, gi_sampler_duration_<INSTR_NAME>, gi_sampler_sr_<INSTR_NAME> mvpReadSampleFromFile "<file>", 2 gi_sampler_sample2_dist_<INSTR_NAME> mvpDistortWaveTable gi_sampler_sample2_<INSTR_NAME>, <distort_type>, <distort_amount> if (<reverse>==1) then itmp mvpReverseTable gi_sampler_sample2_dist_<INSTR_NAME>, gi_sampler_duration_<INSTR_NAME>*gi_sampler_sr_<INSTR_NAME> endif istutter_reset2 table 0, gi_sampler_sample2_<INSTR_NAME> endif endif ispeedup_start = <speedup_start><speedup_start_mod> ispeedup_end = <speedup_end><speedup_end_mod> istart = i(<start>) iend = (<select_end>==0) ? gi_sampler_duration_<INSTR_NAME> : i(<end>) if (iend>gi_sampler_duration_<INSTR_NAME>) then iend = gi_sampler_duration_<INSTR_NAME> endif isample_duration = iend - istart inr_samples = isample_duration*gi_sampler_sr_<INSTR_NAME> isample_start = istart*gi_sampler_sr_<INSTR_NAME> isample_end = iend *gi_sampler_sr_<INSTR_NAME> istutter_len = i(<stutter_len>)<stutter_len_mod> istutter_samples = istutter_len*gi_sampler_sr_<INSTR_NAME>+isample_start iavg_speedup = (ispeedup_start+ispeedup_end)/2 iuse_stutter = (<play_type>!=3) ? <stutter> : 0 istutters = i(<stutters>)<stutters_mod> ibasic_dur = (isample_duration+<stutter>*istutter_len*istutters)/iavg_speedup if (<play_type>!=3) then if (<play_type>==1) then p3 = ibasic_dur endif aph, astutter_env mvpSamplePosition <play_type>, ispeedup_start, ispeedup_end, gi_sampler_sr_<INSTR_NAME>/sr, isample_start, isample_end, iuse_stutter, istutters, istutter_samples itune = <tune> else ifreq getFrequency p5 aph phasor ifreq aph = isample_start + aph*(isample_end - isample_start) itune = 0 endif if (<tune>!=0 && <tune_method>==2) then idummy = p5 ; not used, but seems necessary, otherwise the UDO mvpSampler can't use p5 endif adeclick linseg 0, <attack_len>, 1, p3-<attack_len>-<release_len>, 1, <release_len>, 0 asig1 mvpSampler gi_sampler_sample1_dist_<INSTR_NAME>, aph, <auto_delay>*<auto_delay_amount>, itune, <tune_tech>, <tune_method>, <tune_formant>, <tune_oct><tune_oct_mod>, <sample_pitch>, \ <shift>, <shift_freq><shift_freq_mod>, <blur>, <blur_time>, <smooth>, <smooth_amp>, <smooth_freq>, <downsample_freq>*<downsample>, <add_attack>+<add_release>, adeclick, \ iuse_stutter, astutter_env, istutter_reset1, <filter_type>*<use_filter>, <filter_freq>, <filter_qual> kleft, kright panner <pan><pan_mod> if (gi_sampler_nr_channels_<INSTR_NAME> == 1) then ichorus_stereoize = <chorus_stereoize> if (ichorus_stereoize==0) then asig1 mvpChorus asig1, <chorus_voices>, 0.1, <chorus_depth>, <chorus_speed> asig2 = asig1*kright asig1 = asig1*kleft else asig1, asig2 mvpStereoChorus asig1, <chorus_voices>, 0.1, <chorus_depth>, <chorus_speed> asig1 = asig1*kleft asig2 = asig2*kright endif else asig2 mvpSampler gi_sampler_sample2_dist_<INSTR_NAME>, aph, <auto_delay>*<auto_delay_amount>, itune, <tune_tech>, <tune_method>, <tune_formant>, <tune_oct><tune_oct_mod>, <sample_pitch>, \ <shift>, <shift_freq><shift_freq_mod>, <blur>, <blur_time>, <smooth>, <smooth_amp>, <smooth_freq>, <downsample_freq>*<downsample>, <add_attack>+<add_release>, adeclick, \ iuse_stutter, astutter_env, istutter_reset2, <filter_type>*<use_filter>, <filter_freq>, <filter_qual> ; <chorus_stereoize> is not used in this case asig1 mvpChorus asig1, <chorus_voices>, 0.1, <chorus_depth>, <chorus_speed> asig2 mvpChorus asig2, <chorus_voices>, 0.1, <chorus_depth>, <chorus_speed> asig1 = asig1*kleft asig2 = asig2*kright endif asig1 = asig1*iamp asig2 = asig2*iamp blueMixerOut asig1, asig2 34 18 <html>File:</html> file 79 17 K:\Samples\Berklee44v1\anklung_1.wav 400 speedup_start 366 158 false 0.0 5.0 1.0 60 true 405 134 Speedup 32 134 Distort distort_amount 94 158 false -1.0 1.0 0.28721344 50 true distort_type 85 131 false None 0 Atan 1 Tan 2 Sine 3 Exp 4 Clip 5 Bitcrusher 6 Square 7 Cube 8 Split 9 0 true reverse 32 95 Reverse false true downsample_freq 556 320 true 1000.0 44100.0 36482.61 60 true downsample 524 287 <html>Downsample</html> false true auto_delay 663 287 <html>Auto delay</html> false true auto_delay_amount 692 320 true 0.0 10000.0 6442.5293 60 true tune 18 287 <html>Tune</html> false true tune_oct 29 352 true -3.0 3.0 0.46687937 60 true blur 304 287 <html>Blur</html> false true blur_time 302 320 true 0.0 1.0 0.51651406 60 true play_type 366 96 false Play once 1 Loop 2 Cycle 3 0 true speedup_end 433 158 false 0.0 5.0 1.0 60 true pan 871 158 false -1.0 1.0 0.0 60 false 890 134 Pan pan_mod 871 234 60 smooth 375 287 <html>Smooth</html> false true smooth_amp 383 320 true 0.0 1.0 0.9528363 60 true smooth_freq 453 320 true 0.0 1.0 0.46328682 60 true 401 398 amp 468 398 freq tune_method 18 321 false Octaves 1 Pitch in p5 2 0 false sample_pitch 108 393 7.00 70 107 415 sample pitch add_attack 708 130 atk false false attack_len 700 158 false 0.0 1.0 0.1 60 false add_release 773 130 rel false false release_len 773 158 false 0.0 1.0 0.1 60 false start 201 158 true 0.0 5.0 0.0 60 false end 269 158 true 0.0 5.0 2.961832 60 true 218 134 start select_end 273 130 end false true chorus_voices 864 288 false 0.0 5.0 1.0 0.0 100 true chorus_depth 810 345 true 0.0 0.1 0.017289551 60 true 826 324 Depth chorus_speed 878 345 true 0.0 1.0 0.08465153 60 true 888 324 Speed chorus_stereoize 953 324 stereoize false true stutter_len 523 158 true 0.0 1.0 0.2962486 60 true 535 135 length 962 126 <html>Volume adjust (db)</html> vol_adjust 1006 158 false -40.0 40.0 0.0 60 false stutter 534 94 <html>Stutter</html> false true stutters 588 143 true 1.0 5.0 1.0 1.0 60 true tune_tech 107 321 false PVS 1 Delays 2 1 true shift 205 287 <html>Shift</html> false true shift_freq 203 320 true -500.0 500.0 451.90448 60 true tune_formant 108 352 false None 0 Keep form 1 1 Keep form 2 2 1 true speedup_start_mod 366 237 60 speedup_end_mod 433 237 60 stutter_len_mod 523 237 60 tune_oct_mod 29 431 60 shift_freq_mod 204 398 60 pan_mod6 302 398 60 stutters_mod 588 237 50 801 291 <html>Chorus</html> filter_type 1132 292 false Lowpass 1 Highpass 2 Bandpass 3 Bandreject 4 Rezzy lowpass 5 Rezzy highpass 6 Moogvcf 7 Moogladder 8 6 true filter_freq 1068 345 true 0.0 20000.0 14301.696 50 true 1081 324 Freq filter_qual 1127 345 true 0.0 1.0 0.5409403 50 true 1134 324 Quality use_filter 1049 287 <html>Filter</html> false true mvpReadSampleFromFile iii Si Sname, ichannel xin iname_len strlen Sname Sext strsub Sname, iname_len-4 Sext strlower Sext icmp strcmp Sext, ".mp3" if (icmp!=0) then iduration filelen Sname isr filesr Sname inr_samples = iduration*isr itlen = 2^int(log(inr_samples)/log(2)+1) ; make table length power of 2 bigger than nr of samples itable ftgen 0, 0, itlen, 1, Sname, 0, 0, ichannel else iduration = 1 ;iduration mp3len Sname ;inr_samples = iduration*44100 ;itlen = 2^int(log(inr_samples)/log(2)+1) ; make table length power of 2 bigger than nr of samples print iduration itable ftgen 0, 0, 32768, 49, Sname, 0, ichannel print iduration+1 itlen = ftlen(itable) print itlen endif xout itable, iduration, isr mvpDistortWaveTable i iii iwave, idistortion_type, idistortion_amount xin i_table_len = ftlen(iwave) if (idistortion_type==1) then idistortion_amount = idistortion_amount * 20 ipos = 0 fill_loop_taninv: ival tab_i ipos, iwave ival = taninv(ival*idistortion_amount) tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_taninv elseif (idistortion_type==2) then idistortion_amount = idistortion_amount * 10 idistortion_factor = taninv(idistortion_amount) ipos = 0 fill_loop_tan: ival tab_i ipos, iwave ival = tan(ival*idistortion_factor)/idistortion_amount tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_tan elseif (idistortion_type==3) then idistortion_amount = idistortion_amount * 20 ipos = 0 fill_loop_sin: ival tab_i ipos, iwave ival = sin(ival*idistortion_amount) tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_sin elseif (idistortion_type==4) then idistortion_amount = idistortion_amount * 5 imin = exp(-idistortion_amount) imax = exp( idistortion_amount) isubtract = (imin+imax)/2 iscale = (imax-imin)/2 ipos = 0 fill_loop_exp: ival tab_i ipos, iwave ival = (exp(ival*idistortion_amount)-isubtract)/iscale tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_exp elseif (idistortion_type==5) then idistortion_amount = idistortion_amount * 20 ipos = 0 fill_loop_clip: ival tab_i ipos, iwave ival = ival*idistortion_amount if (ival>1) then ival = 1 elseif (ival<-1) then ival = -1 endif tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_clip elseif (idistortion_type==6) then idistortion_amount = 2^(16-idistortion_amount*16) ipos = 0 fill_loop_bit_crusher: ival tab_i ipos, iwave ival = round(ival*idistortion_amount)/idistortion_amount tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_bit_crusher elseif (idistortion_type==7) then ipos = 0 imin = 100 imax = -100 idry = 1-abs(idistortion_amount) fill_loop_square: ival tab_i ipos, iwave ival = idry*ival + idistortion_amount*ival*ival imin = (ival<imin) ? ival : imin imax = (ival>imax) ? ival : imax tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_square iavg = (imax+imin)/2 iscale = 2/(imax-imin) ipos = 0 fill_loop_square_normalize: ival tab_i ipos, iwave ival = (ival-iavg)*iscale tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_square_normalize elseif (idistortion_type==8) then ipos = 0 imin = 100 imax = -100 idry = 1-abs(idistortion_amount) fill_loop_cube: ival tab_i ipos, iwave ival = idry*ival + idistortion_amount*ival*ival*ival imin = (ival<imin) ? ival : imin imax = (ival>imax) ? ival : imax tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_cube iavg = (imax+imin)/2 iscale = 2/(imax-imin) ipos = 0 fill_loop_cube_normalize: ival tab_i ipos, iwave ival = (ival-iavg)*iscale tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_cube_normalize elseif (idistortion_type==9) then ; if idistortion_amount >= 0, move positive samples up and negative samples down ; opposite if idistortion_amount < 0 ; scale so that 1 and -1 are not changed, so signals between -1 and 1 stay in that range ia = 1-idistortion_amount ipos = 0 fill_loop_split: ival tab_i ipos, iwave ival = (ival>=0) ? 1-ia*(1-ival) : -1-ia*(-1-ival) tabw_i ival, ipos, iwave loop_lt ipos, 1, i_table_len, fill_loop_split endif xout iwave mvpReverseTable i ii iwave, i_table_len xin ipos = 0 reverse_loop: ival1 tab_i ipos, iwave ival2 tab_i i_table_len-ipos-1, iwave tabw_i ival2, ipos, iwave tabw_i ival1, i_table_len-ipos-1, iwave loop_lt ipos, 1, i_table_len/2, reverse_loop xout iwave getFrequency i i ipch xin iout = (ipch < 15 ? cpspch(ipch) : ipch) xout iout PitchShifter a aki setksmps 1 ; kr=sr asig,kpitch,idel xin kdelrate = (kpitch-1)/idel avdel phasor -kdelrate ; 1 to 0 avdel2 phasor -kdelrate, 0.5 ; 1/2 buffer offset afade tablei avdel, gi_Hamming, 1, 0, 1 ; crossfade windows afade2 tablei avdel2,gi_Hamming, 1, 0, 1 adump delayr idel atap1 deltapi avdel*idel ; variable delay taps atap2 deltapi avdel2*idel amix = atap1*afade + atap2*afade2 ; fade in/out the delay taps delayw asig xout amix mvpFilter a aikki asig, itype, kfreq, kqual, ibalance xin if (kfreq>sr/2) then kfreq = sr/2 endif if (itype==1) then afilt tone asig, kfreq ;afilt butterlp asig, kfreq elseif (itype==2) then afilt atone asig, kfreq elseif (itype==3) then afilt reson asig, kfreq, kfreq/(kqual*20), 2 elseif (itype==4) then afilt areson asig, kfreq, kfreq/(kqual*20), 2 elseif (itype==5) then afilt rezzy asig, kfreq, kqual*100, 0 elseif (itype==6) then afilt rezzy asig, kfreq, kqual*100, 1 elseif (itype==7) then afilt moogvcf asig, kfreq, kqual elseif (itype==8) then afilt moogladder asig, kfreq, kqual endif if (ibalance!=0) then afilt balance afilt, asig endif xout afilt mvpSampler a iakiiiikiikikikkkiaiaiikk itable, aph, kauto_delay, itune, itune_tech, itune_method, itune_form, ktune_oct, isample_pitch, ishift, kshift_freq, iblur, kblur_time, ismooth, ksmooth_amp, ksmooth_freq, \ kdownsample, iuse_declick, adeclick, iuse_stutter_env, astutter_env, istutter_reset, ifilter_type, kfilter_freq, kfilter_qual xin asig tablei aph, itable if (kauto_delay!=0) then asig tablei aph+(1+asig)*kauto_delay, itable endif if (iuse_stutter_env!=0) then asig = asig*astutter_env+istutter_reset*(1-astutter_env) endif if (itune!=0 && itune_tech==2) then if (itune_method==1) then kratio = 2^ktune_oct else isample_freq getFrequency isample_pitch ifreq getFrequency p5 kratio = ifreq/isample_freq endif asig PitchShifter asig, kratio, 0.01 endif itune_pvs = (itune==1 && itune_tech==1) ? 1 : 0 if (itune_pvs!=0 || ishift!=0 || iblur!=0 || ismooth!=0) then fsig pvsanal asig, 1024,256,2048,0 if (itune!=0) then if (itune_method==1) then kratio = 2^ktune_oct else isample_freq getFrequency isample_pitch ifreq getFrequency p5 kratio = ifreq/isample_freq endif fsig1 pvscale fsig, kratio, itune_form fsig2 pvscale fsig, kratio*2, itune_form asig1 pvsynth fsig1 asig2 pvsynth fsig2 fsig pvsanal asig1+asig2, 1024,256,2048,0 endif if (ishift!=0) then fsigo pvshift fsig, kshift_freq, 100, 0, 1 fsig = fsigo endif if (iblur!=0) then fsigo pvsblur fsig, kblur_time, 1 fsig = fsigo endif if (ismooth!=0) then fsigo pvsmooth fsig, ksmooth_amp, ksmooth_freq fsig = fsigo endif asig pvsynth fsig endif if (ifilter_type!=0) then asig mvpFilter asig, ifilter_type, kfilter_freq, kfilter_qual, 0 endif if (kdownsample!=0) then agate mpulse 1, 1/kdownsample asig samphold asig, agate endif asig dcblock asig if (iuse_declick!=0) then asig = asig*adeclick endif xout asig panner kk k ; sin/cos taper, constant power. kPanpos xin kPanpos = kPanpos/2 kPanpos = kPanpos * 3.14159265359 * .5 kLeft = sqrt(2) / 2 * (cos(kPanpos) - sin(kPanpos)) kRight = sqrt(2) / 2 * (cos(kPanpos) + sin(kPanpos)) xout kLeft,kRight mvpChorus a aiikk asig, inr_voices, imax_depth, kdepth, kspeed xin ach = asig if (inr_voices!=0) then atemp delayr imax_depth*2 alfo oscili kdepth, kspeed, gi_cosine adel deltapi alfo+kdepth ach = ach + adel if (inr_voices>1) then alfo oscili kdepth, kspeed, gi_cosine, 1/inr_voices adel deltapi alfo+kdepth ach = ach + adel endif if (inr_voices>2) then alfo oscili kdepth, kspeed, gi_cosine, 2/inr_voices adel deltapi alfo+kdepth ach = ach + adel endif if (inr_voices>3) then alfo oscili kdepth, kspeed, gi_cosine, 3/inr_voices adel deltapi alfo+kdepth ach = ach + adel endif if (inr_voices>4) then alfo oscili kdepth, kspeed, gi_cosine, 4/inr_voices adel deltapi alfo+kdepth ach = ach + adel endif delayw asig endif xout ach mvpStereoChorus aa aiikk asig, inr_voices, imax_depth, kdepth, kspeed xin if (inr_voices%2==0) then ach1 = asig ach2 = asig else ach1 = asig ach2 = 0 endif if (inr_voices!=0) then atemp delayr imax_depth*2 alfo oscili kdepth, kspeed, gi_cosine adel deltapi alfo+kdepth ach2 = ach2 + adel print 2 if (inr_voices>1) then alfo oscili kdepth, kspeed, gi_cosine, 1/inr_voices adel deltapi alfo+kdepth ach1 = ach1 + adel print 1 endif if (inr_voices>2) then alfo oscili kdepth, kspeed, gi_cosine, 2/inr_voices adel deltapi alfo+kdepth ach2 = ach2 + adel print 2 endif if (inr_voices>3) then alfo oscili kdepth, kspeed, gi_cosine, 3/inr_voices adel deltapi alfo+kdepth ach1 = ach1 + adel endif if (inr_voices>4) then alfo oscili kdepth, kspeed, gi_cosine, 4/inr_voices adel deltapi alfo+kdepth ach2 = ach2 + adel endif delayw asig endif xout ach1, ach2 mvpSamplePosition aa iiiiiiiii setksmps 1 ; need sr=kr iplay_type, ispeedup_start, ispeedup_end, isr, isample_start, isample_end, iuse_stutter, istutters, istutter_samples xin kspeedup linseg ispeedup_start, p3, ispeedup_end kstep = kspeedup*isr aph init isample_start aph = aph+kstep kph downsamp aph kstutter_count init istutters astutter_env init 1 istutter_dec = istutter_samples/5 if (iuse_stutter==1) then if (kph>istutter_samples && kstutter_count>0) then aph = isample_start astutter_env = 1 kstutter_count = kstutter_count-1 elseif (istutter_samples-kph<istutter_dec && kstutter_count>0) then astutter_env = (istutter_samples-kph)/istutter_dec else astutter_env = 1 endif endif if (iplay_type==2) then if (kph>isample_end) then aph = isample_start kstutter_count = istutters endif endif xout aph, astutter_env UDO to update aph at sr speed