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Carla/source/native-plugins/zynaddsubfx/Synth/ADnote.cpp

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  1. /*

  2.   ZynAddSubFX - a software synthesizer

  3. 

  4.   ADnote.cpp - The "additive" synthesizer

  5.   Copyright (C) 2002-2005 Nasca Octavian Paul

  6.   Author: Nasca Octavian Paul

  7. 

  8.   This program is free software; you can redistribute it and/or modify

  9.   it under the terms of version 2 of the GNU General Public License

  10.   as published by the Free Software Foundation.

  11. 

  12.   This program is distributed in the hope that it will be useful,

  13.   but WITHOUT ANY WARRANTY; without even the implied warranty of

  14.   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

  15.   GNU General Public License (version 2 or later) for more details.

  16. 

  17.   You should have received a copy of the GNU General Public License (version 2)

  18.   along with this program; if not, write to the Free Software Foundation,

  19.   Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA

  20. */

  21. 

  22. #include <cmath>

  23. #include <cstdlib>

  24. #include <cstdio>

  25. #include <cstring>

  26. #include <cassert>

  27. #include <stdint.h>

  28. 

  29. #include "../globals.h"

  30. #include "../Misc/Util.h"

  31. #include "../Misc/Allocator.h"

  32. #include "../DSP/Filter.h"

  33. #include "../Params/ADnoteParameters.h"

  34. #include "../Params/FilterParams.h"

  35. #include "OscilGen.h"

  36. #include "ADnote.h"

  37. 

  38. ADnote::ADnote(ADnoteParameters *pars_, SynthParams &spars)

  39.     :SynthNote(spars), pars(*pars_)

  40. {

  41.     tmpwavel = memory.valloc<float>(synth.buffersize);

  42.     tmpwaver = memory.valloc<float>(synth.buffersize);

  43.     bypassl  = memory.valloc<float>(synth.buffersize);

  44.     bypassr  = memory.valloc<float>(synth.buffersize);

  45. 

  46.     ADnoteParameters &pars = *pars_;

  47.     portamento  = spars.portamento;

  48.     midinote    = spars.note;

  49.     NoteEnabled = ON;

  50.     basefreq    = spars.frequency;

  51.     velocity    = spars.velocity;

  52.     stereo = pars.GlobalPar.PStereo;

  53. 

  54.     NoteGlobalPar.Detune = getdetune(pars.GlobalPar.PDetuneType,

  55.                                      pars.GlobalPar.PCoarseDetune,

  56.                                      pars.GlobalPar.PDetune);

  57.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  58. 

  59.     if(pars.GlobalPar.PPanning == 0)

  60.         NoteGlobalPar.Panning = RND;

  61.     else

  62.         NoteGlobalPar.Panning = pars.GlobalPar.PPanning / 128.0f;

  63. 

  64. 

  65.     NoteGlobalPar.FilterCenterPitch = pars.GlobalPar.GlobalFilter->getfreq() //center freq

  66.                                       + pars.GlobalPar.PFilterVelocityScale

  67.                                       / 127.0f * 6.0f       //velocity sensing

  68.                                       * (VelF(velocity,

  69.                                               pars.GlobalPar.

  70.                                               PFilterVelocityScaleFunction) - 1);

  71. 

  72.     NoteGlobalPar.Fadein_adjustment =

  73.         pars.GlobalPar.Fadein_adjustment / (float)FADEIN_ADJUSTMENT_SCALE;

  74.     NoteGlobalPar.Fadein_adjustment *= NoteGlobalPar.Fadein_adjustment;

  75.     if(pars.GlobalPar.PPunchStrength != 0) {

  76.         NoteGlobalPar.Punch.Enabled = 1;

  77.         NoteGlobalPar.Punch.t = 1.0f; //start from 1.0f and to 0.0f

  78.         NoteGlobalPar.Punch.initialvalue =

  79.             ((powf(10, 1.5f * pars.GlobalPar.PPunchStrength / 127.0f) - 1.0f)

  80.              * VelF(velocity,

  81.                     pars.GlobalPar.PPunchVelocitySensing));

  82.         float time =

  83.             powf(10, 3.0f * pars.GlobalPar.PPunchTime / 127.0f) / 10000.0f;   //0.1f .. 100 ms

  84.         float stretch = powf(440.0f / spars.frequency,

  85.                              pars.GlobalPar.PPunchStretch / 64.0f);

  86.         NoteGlobalPar.Punch.dt = 1.0f / (time * synth.samplerate_f * stretch);

  87.     }

  88.     else

  89.         NoteGlobalPar.Punch.Enabled = 0;

  90. 

  91.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  92.         for (int i = 0; i < 14; i++)

  93.             pinking[nvoice][i] = 0.0;

  94. 

  95.         pars.VoicePar[nvoice].OscilSmp->newrandseed(prng());

  96.         NoteVoicePar[nvoice].OscilSmp = NULL;

  97.         NoteVoicePar[nvoice].FMSmp    = NULL;

  98.         NoteVoicePar[nvoice].VoiceOut = NULL;

  99. 

  100.         NoteVoicePar[nvoice].FMVoice = -1;

  101.         unison_size[nvoice] = 1;

  102. 

  103.         if(!pars.VoicePar[nvoice].Enabled) {

  104.             NoteVoicePar[nvoice].Enabled = OFF;

  105.             continue; //the voice is disabled

  106.         }

  107. 

  108.         unison_stereo_spread[nvoice] =

  109.             pars.VoicePar[nvoice].Unison_stereo_spread / 127.0f;

  110.         int unison = pars.VoicePar[nvoice].Unison_size;

  111.         if(unison < 1)

  112.             unison = 1;

  113. 

  114. 	// Since noise unison of greater than two is touch goofy...

  115. 	if (pars.VoicePar[nvoice].Type != 0 && unison > 2)

  116. 	    unison = 2;

  117. 

  118.         //compute unison

  119.         unison_size[nvoice] = unison;

  120. 

  121.         unison_base_freq_rap[nvoice] = memory.valloc<float>(unison);

  122.         unison_freq_rap[nvoice]      = memory.valloc<float>(unison);

  123.         unison_invert_phase[nvoice]  = memory.valloc<bool>(unison);

  124.         float unison_spread =

  125.             pars.getUnisonFrequencySpreadCents(nvoice);

  126.         float unison_real_spread = powf(2.0f, (unison_spread * 0.5f) / 1200.0f);

  127.         float unison_vibratto_a  =

  128.             pars.VoicePar[nvoice].Unison_vibratto / 127.0f; //0.0f .. 1.0f

  129. 

  130. 

  131.         switch(unison) {

  132.             case 1:

  133.                 unison_base_freq_rap[nvoice][0] = 1.0f; //if the unison is not used, always make the only subvoice to have the default note

  134.                 break;

  135.             case 2: { //unison for 2 subvoices

  136.                 unison_base_freq_rap[nvoice][0] = 1.0f / unison_real_spread;

  137.                 unison_base_freq_rap[nvoice][1] = unison_real_spread;

  138.             };

  139.                 break;

  140.             default: { //unison for more than 2 subvoices

  141.                 float unison_values[unison];

  142.                 float min = -1e-6, max = 1e-6;

  143.                 for(int k = 0; k < unison; ++k) {

  144.                     float step = (k / (float) (unison - 1)) * 2.0f - 1.0f; //this makes the unison spread more uniform

  145.                     float val  = step + (RND * 2.0f - 1.0f) / (unison - 1);

  146.                     unison_values[k] = val;

  147.                     if (min > val) {

  148.                         min = val;

  149.                     }

  150.                     if (max < val) {

  151.                         max = val;

  152.                     }

  153.                 }

  154.                 float diff = max - min;

  155.                 for(int k = 0; k < unison; ++k) {

  156.                     unison_values[k] =

  157.                         (unison_values[k] - (max + min) * 0.5f) / diff;             //the lowest value will be -1 and the highest will be 1

  158.                     unison_base_freq_rap[nvoice][k] =

  159.                         powf(2.0f, (unison_spread * unison_values[k]) / 1200);

  160.                 }

  161.             };

  162.         }

  163. 

  164.         //unison vibrattos

  165.         if(unison > 1)

  166.             for(int k = 0; k < unison; ++k) //reduce the frequency difference for larger vibrattos

  167.                 unison_base_freq_rap[nvoice][k] = 1.0f

  168.                                                   + (unison_base_freq_rap[

  169.                                                          nvoice][k] - 1.0f)

  170.                                                   * (1.0f - unison_vibratto_a);

  171.         unison_vibratto[nvoice].step      = memory.valloc<float>(unison);

  172.         unison_vibratto[nvoice].position  = memory.valloc<float>(unison);

  173.         unison_vibratto[nvoice].amplitude =

  174.             (unison_real_spread - 1.0f) * unison_vibratto_a;

  175. 

  176.         float increments_per_second = synth.samplerate_f / synth.buffersize_f;

  177.         const float vib_speed = pars.VoicePar[nvoice].Unison_vibratto_speed / 127.0f;

  178.         float vibratto_base_period  = 0.25f * powf(2.0f, (1.0f - vib_speed) * 4.0f);

  179.         for(int k = 0; k < unison; ++k) {

  180.             unison_vibratto[nvoice].position[k] = RND * 1.8f - 0.9f;

  181.             //make period to vary randomly from 50% to 200% vibratto base period

  182.             float vibratto_period = vibratto_base_period

  183.                                     * powf(2.0f, RND * 2.0f - 1.0f);

  184. 

  185.             float m = 4.0f / (vibratto_period * increments_per_second);

  186.             if(RND < 0.5f)

  187.                 m = -m;

  188.             unison_vibratto[nvoice].step[k] = m;

  189.         }

  190. 

  191.         if(unison == 1) { //no vibratto for a single voice

  192.             unison_vibratto[nvoice].step[0]     = 0.0f;

  193.             unison_vibratto[nvoice].position[0] = 0.0f;

  194.             unison_vibratto[nvoice].amplitude   = 0.0f;

  195.         }

  196. 

  197.         //phase invert for unison

  198.         unison_invert_phase[nvoice][0] = false;

  199.         if(unison != 1) {

  200.             int inv = pars.VoicePar[nvoice].Unison_invert_phase;

  201.             switch(inv) {

  202.                 case 0: for(int k = 0; k < unison; ++k)

  203.                         unison_invert_phase[nvoice][k] = false;

  204.                     break;

  205.                 case 1: for(int k = 0; k < unison; ++k)

  206.                         unison_invert_phase[nvoice][k] = (RND > 0.5f);

  207.                     break;

  208.                 default: for(int k = 0; k < unison; ++k)

  209.                         unison_invert_phase[nvoice][k] =

  210.                             (k % inv == 0) ? true : false;

  211.                     break;

  212.             }

  213.         }

  214. 

  215. 

  216.         oscfreqhi[nvoice]   = memory.valloc<int>(unison);

  217.         oscfreqlo[nvoice]   = memory.valloc<float>(unison);

  218.         oscfreqhiFM[nvoice] = memory.valloc<unsigned int>(unison);

  219.         oscfreqloFM[nvoice] = memory.valloc<float>(unison);

  220.         oscposhi[nvoice]    = memory.valloc<int>(unison);

  221.         oscposlo[nvoice]    = memory.valloc<float>(unison);

  222.         oscposhiFM[nvoice]  = memory.valloc<unsigned int>(unison);

  223.         oscposloFM[nvoice]  = memory.valloc<float>(unison);

  224. 

  225.         NoteVoicePar[nvoice].Enabled     = ON;

  226.         NoteVoicePar[nvoice].fixedfreq   = pars.VoicePar[nvoice].Pfixedfreq;

  227.         NoteVoicePar[nvoice].fixedfreqET = pars.VoicePar[nvoice].PfixedfreqET;

  228. 

  229.         //use the Globalpars.detunetype if the detunetype is 0

  230.         if(pars.VoicePar[nvoice].PDetuneType != 0) {

  231.             NoteVoicePar[nvoice].Detune = getdetune(

  232.                 pars.VoicePar[nvoice].PDetuneType,

  233.                 pars.VoicePar[nvoice].

  234.                 PCoarseDetune,

  235.                 8192); //coarse detune

  236.             NoteVoicePar[nvoice].FineDetune = getdetune(

  237.                 pars.VoicePar[nvoice].PDetuneType,

  238.                 0,

  239.                 pars.VoicePar[nvoice].PDetune); //fine detune

  240.         }

  241.         else {

  242.             NoteVoicePar[nvoice].Detune = getdetune(

  243.                 pars.GlobalPar.PDetuneType,

  244.                 pars.VoicePar[nvoice].

  245.                 PCoarseDetune,

  246.                 8192); //coarse detune

  247.             NoteVoicePar[nvoice].FineDetune = getdetune(

  248.                 pars.GlobalPar.PDetuneType,

  249.                 0,

  250.                 pars.VoicePar[nvoice].PDetune); //fine detune

  251.         }

  252.         if(pars.VoicePar[nvoice].PFMDetuneType != 0)

  253.             NoteVoicePar[nvoice].FMDetune = getdetune(

  254.                 pars.VoicePar[nvoice].PFMDetuneType,

  255.                 pars.VoicePar[nvoice].

  256.                 PFMCoarseDetune,

  257.                 pars.VoicePar[nvoice].PFMDetune);

  258.         else

  259.             NoteVoicePar[nvoice].FMDetune = getdetune(

  260.                 pars.GlobalPar.PDetuneType,

  261.                 pars.VoicePar[nvoice].

  262.                 PFMCoarseDetune,

  263.                 pars.VoicePar[nvoice].PFMDetune);

  264. 

  265. 

  266. 

  267.         for(int k = 0; k < unison; ++k) {

  268.             oscposhi[nvoice][k]   = 0;

  269.             oscposlo[nvoice][k]   = 0.0f;

  270.             oscposhiFM[nvoice][k] = 0;

  271.             oscposloFM[nvoice][k] = 0.0f;

  272.         }

  273. 

  274.         //the extra points contains the first point

  275.         NoteVoicePar[nvoice].OscilSmp =

  276.             memory.valloc<float>(synth.oscilsize + OSCIL_SMP_EXTRA_SAMPLES);

  277. 

  278.         //Get the voice's oscil or external's voice oscil

  279.         int vc = nvoice;

  280.         if(pars.VoicePar[nvoice].Pextoscil != -1)

  281.             vc = pars.VoicePar[nvoice].Pextoscil;

  282.         if(!pars.GlobalPar.Hrandgrouping)

  283.             pars.VoicePar[vc].OscilSmp->newrandseed(prng());

  284.         int oscposhi_start =

  285.             pars.VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,

  286.                                              getvoicebasefreq(nvoice),

  287.                                              pars.VoicePar[nvoice].Presonance);

  288. 

  289.         //I store the first elments to the last position for speedups

  290.         for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)

  291.             NoteVoicePar[nvoice].OscilSmp[synth.oscilsize

  292.                                           + i] =

  293.                 NoteVoicePar[nvoice].OscilSmp[i];

  294. 

  295.         oscposhi_start +=

  296.             (int)((pars.VoicePar[nvoice].Poscilphase

  297.                    - 64.0f) / 128.0f * synth.oscilsize + synth.oscilsize * 4);

  298.         oscposhi_start %= synth.oscilsize;

  299. 

  300.         for(int k = 0; k < unison; ++k) {

  301.             oscposhi[nvoice][k] = oscposhi_start;

  302.             //put random starting point for other subvoices

  303.             oscposhi_start      =

  304.                 (int)(RND * pars.VoicePar[nvoice].Unison_phase_randomness /

  305.                         127.0f * (synth.oscilsize - 1));

  306.         }

  307. 

  308.         NoteVoicePar[nvoice].FreqLfo      = NULL;

  309.         NoteVoicePar[nvoice].FreqEnvelope = NULL;

  310. 

  311.         NoteVoicePar[nvoice].AmpLfo      = NULL;

  312.         NoteVoicePar[nvoice].AmpEnvelope = NULL;

  313. 

  314.         NoteVoicePar[nvoice].VoiceFilterL   = NULL;

  315.         NoteVoicePar[nvoice].VoiceFilterR   = NULL;

  316.         NoteVoicePar[nvoice].FilterEnvelope = NULL;

  317.         NoteVoicePar[nvoice].FilterLfo      = NULL;

  318. 

  319.         NoteVoicePar[nvoice].FilterCenterPitch =

  320.             pars.VoicePar[nvoice].VoiceFilter->getfreq();

  321.         NoteVoicePar[nvoice].filterbypass =

  322.             pars.VoicePar[nvoice].Pfilterbypass;

  323. 

  324.         switch(pars.VoicePar[nvoice].PFMEnabled) {

  325.             case 1:

  326.                 NoteVoicePar[nvoice].FMEnabled = MORPH;

  327.                 break;

  328.             case 2:

  329.                 NoteVoicePar[nvoice].FMEnabled = RING_MOD;

  330.                 break;

  331.             case 3:

  332.                 NoteVoicePar[nvoice].FMEnabled = PHASE_MOD;

  333.                 break;

  334.             case 4:

  335.                 NoteVoicePar[nvoice].FMEnabled = FREQ_MOD;

  336.                 break;

  337.             case 5:

  338.                 NoteVoicePar[nvoice].FMEnabled = PITCH_MOD;

  339.                 break;

  340.             default:

  341.                 NoteVoicePar[nvoice].FMEnabled = NONE;

  342.         }

  343. 

  344.         NoteVoicePar[nvoice].FMVoice = pars.VoicePar[nvoice].PFMVoice;

  345.         NoteVoicePar[nvoice].FMFreqEnvelope = NULL;

  346.         NoteVoicePar[nvoice].FMAmpEnvelope  = NULL;

  347. 

  348.         //Compute the Voice's modulator volume (incl. damping)

  349.         float fmvoldamp = powf(440.0f / getvoicebasefreq(

  350.                                    nvoice),

  351.                                pars.VoicePar[nvoice].PFMVolumeDamp / 64.0f

  352.                                - 1.0f);

  353.         switch(NoteVoicePar[nvoice].FMEnabled) {

  354.             case PHASE_MOD:

  355.                 fmvoldamp =

  356.                     powf(440.0f / getvoicebasefreq(

  357.                              nvoice), pars.VoicePar[nvoice].PFMVolumeDamp

  358.                          / 64.0f);

  359.                 NoteVoicePar[nvoice].FMVolume =

  360.                     (expf(pars.VoicePar[nvoice].PFMVolume / 127.0f

  361.                           * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;

  362.                 break;

  363.             case FREQ_MOD:

  364.                 NoteVoicePar[nvoice].FMVolume =

  365.                     (expf(pars.VoicePar[nvoice].PFMVolume / 127.0f

  366.                           * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;

  367.                 break;

  368.             //    case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars.VoicePar[nvoice].PFMVolume/127.0f*8.0f)*fmvoldamp;//???????????

  369.             //	          break;

  370.             default:

  371.                 if(fmvoldamp > 1.0f)

  372.                     fmvoldamp = 1.0f;

  373.                 NoteVoicePar[nvoice].FMVolume =

  374.                     pars.VoicePar[nvoice].PFMVolume

  375.                     / 127.0f * fmvoldamp;

  376.         }

  377. 

  378.         //Voice's modulator velocity sensing

  379.         NoteVoicePar[nvoice].FMVolume *=

  380.             VelF(velocity,

  381.                  pars.VoicePar[nvoice].PFMVelocityScaleFunction);

  382. 

  383.         FMoldsmp[nvoice] = memory.valloc<float>(unison);

  384.         for(int k = 0; k < unison; ++k)

  385.             FMoldsmp[nvoice][k] = 0.0f;                     //this is for FM (integration)

  386. 

  387.         firsttick[nvoice] = 1;

  388.         NoteVoicePar[nvoice].DelayTicks =

  389.             (int)((expf(pars.VoicePar[nvoice].PDelay / 127.0f

  390.                         * logf(50.0f))

  391.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  392.     }

  393. 

  394.     max_unison = 1;

  395.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)

  396.         if(unison_size[nvoice] > max_unison)

  397.             max_unison = unison_size[nvoice];

  398. 

  399. 

  400.     tmpwave_unison = memory.valloc<float*>(max_unison);

  401.     for(int k = 0; k < max_unison; ++k) {

  402.         tmpwave_unison[k] = memory.valloc<float>(synth.buffersize);

  403.         memset(tmpwave_unison[k], 0, synth.bufferbytes);

  404.     }

  405. 

  406.     initparameters();

  407. }

  408. 

  409. SynthNote *ADnote::cloneLegato(void)

  410. {

  411.     SynthParams sp{memory, ctl, synth, time, legato.param.freq, velocity, 

  412.                    (bool)portamento, legato.param.midinote, true};

  413.     return memory.alloc<ADnote>(&pars, sp);

  414. }

  415. 

  416. // ADlegatonote: This function is (mostly) a copy of ADnote(...) and

  417. // initparameters() stuck together with some lines removed so that it

  418. // only alter the already playing note (to perform legato). It is

  419. // possible I left stuff that is not required for this.

  420. void ADnote::legatonote(LegatoParams lpars)

  421. {

  422.     //ADnoteParameters &pars = *partparams;

  423. 

  424.     // Manage legato stuff

  425.     if(legato.update(lpars))

  426.         return;

  427. 

  428.     portamento = lpars.portamento;

  429.     midinote   = lpars.midinote;

  430.     basefreq   = lpars.frequency;

  431. 

  432.     if(velocity > 1.0f)

  433.         velocity = 1.0f;

  434.     velocity = lpars.velocity;

  435. 

  436.     NoteGlobalPar.Detune = getdetune(pars.GlobalPar.PDetuneType,

  437.                                      pars.GlobalPar.PCoarseDetune,

  438.                                      pars.GlobalPar.PDetune);

  439.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  440. 

  441.     if(pars.GlobalPar.PPanning == 0)

  442.         NoteGlobalPar.Panning = RND;

  443.     else

  444.         NoteGlobalPar.Panning = pars.GlobalPar.PPanning / 128.0f;

  445. 

  446.     //center freq

  447.     NoteGlobalPar.FilterCenterPitch = pars.GlobalPar.GlobalFilter->getfreq()

  448.                                       + pars.GlobalPar.PFilterVelocityScale

  449.                                       / 127.0f * 6.0f          //velocity sensing

  450.                                       * (VelF(velocity,

  451.                                               pars.GlobalPar.

  452.                                               PFilterVelocityScaleFunction) - 1);

  453. 

  454. 

  455.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  456.         if(NoteVoicePar[nvoice].Enabled == OFF)

  457.             continue;  //(gf) Stay the same as first note in legato.

  458. 

  459.         NoteVoicePar[nvoice].fixedfreq   = pars.VoicePar[nvoice].Pfixedfreq;

  460.         NoteVoicePar[nvoice].fixedfreqET = pars.VoicePar[nvoice].PfixedfreqET;

  461. 

  462.         //use the Globalpars.detunetype if the detunetype is 0

  463.         if(pars.VoicePar[nvoice].PDetuneType != 0) {

  464.             NoteVoicePar[nvoice].Detune = getdetune(

  465.                 pars.VoicePar[nvoice].PDetuneType,

  466.                 pars.VoicePar[nvoice].PCoarseDetune,

  467.                 8192); //coarse detune

  468.             NoteVoicePar[nvoice].FineDetune = getdetune(

  469.                 pars.VoicePar[nvoice].PDetuneType,

  470.                 0,

  471.                 pars.VoicePar[nvoice].PDetune); //fine detune

  472.         }

  473.         else {

  474.             NoteVoicePar[nvoice].Detune = getdetune(

  475.                 pars.GlobalPar.PDetuneType,

  476.                 pars.VoicePar[nvoice].PCoarseDetune,

  477.                 8192); //coarse detune

  478.             NoteVoicePar[nvoice].FineDetune = getdetune(

  479.                 pars.GlobalPar.PDetuneType,

  480.                 0,

  481.                 pars.VoicePar[nvoice].PDetune); //fine detune

  482.         }

  483.         if(pars.VoicePar[nvoice].PFMDetuneType != 0)

  484.             NoteVoicePar[nvoice].FMDetune = getdetune(

  485.                 pars.VoicePar[nvoice].PFMDetuneType,

  486.                 pars.VoicePar[nvoice].PFMCoarseDetune,

  487.                 pars.VoicePar[nvoice].PFMDetune);

  488.         else

  489.             NoteVoicePar[nvoice].FMDetune = getdetune(

  490.                 pars.GlobalPar.PDetuneType,

  491.                 pars.VoicePar[nvoice].PFMCoarseDetune,

  492.                 pars.VoicePar[nvoice].PFMDetune);

  493. 

  494. 

  495.         //Get the voice's oscil or external's voice oscil

  496.         int vc = nvoice;

  497.         if(pars.VoicePar[nvoice].Pextoscil != -1)

  498.             vc = pars.VoicePar[nvoice].Pextoscil;

  499.         if(!pars.GlobalPar.Hrandgrouping)

  500.             pars.VoicePar[vc].OscilSmp->newrandseed(prng());

  501. 

  502.         pars.VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,

  503.                                          getvoicebasefreq(nvoice),

  504.                                          pars.VoicePar[nvoice].Presonance); //(gf)Modif of the above line.

  505. 

  506.         //I store the first elments to the last position for speedups

  507.         for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)

  508.             NoteVoicePar[nvoice].OscilSmp[synth.oscilsize

  509.                                           + i] =

  510.                 NoteVoicePar[nvoice].OscilSmp[i];

  511. 

  512. 

  513.         NoteVoicePar[nvoice].FilterCenterPitch =

  514.             pars.VoicePar[nvoice].VoiceFilter->getfreq();

  515.         NoteVoicePar[nvoice].filterbypass =

  516.             pars.VoicePar[nvoice].Pfilterbypass;

  517. 

  518. 

  519.         NoteVoicePar[nvoice].FMVoice = pars.VoicePar[nvoice].PFMVoice;

  520. 

  521.         //Compute the Voice's modulator volume (incl. damping)

  522.         float fmvoldamp = powf(440.0f / getvoicebasefreq(nvoice),

  523.                                pars.VoicePar[nvoice].PFMVolumeDamp / 64.0f

  524.                                - 1.0f);

  525. 

  526.         switch(NoteVoicePar[nvoice].FMEnabled) {

  527.             case PHASE_MOD:

  528.                 fmvoldamp =

  529.                     powf(440.0f / getvoicebasefreq(

  530.                              nvoice), pars.VoicePar[nvoice].PFMVolumeDamp

  531.                          / 64.0f);

  532.                 NoteVoicePar[nvoice].FMVolume =

  533.                     (expf(pars.VoicePar[nvoice].PFMVolume / 127.0f

  534.                           * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;

  535.                 break;

  536.             case FREQ_MOD:

  537.                 NoteVoicePar[nvoice].FMVolume =

  538.                     (expf(pars.VoicePar[nvoice].PFMVolume / 127.0f

  539.                           * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;

  540.                 break;

  541.             //    case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars.VoicePar[nvoice].PFMVolume/127.0f*8.0f)*fmvoldamp;//???????????

  542.             //	          break;

  543.             default:

  544.                 if(fmvoldamp > 1.0f)

  545.                     fmvoldamp = 1.0f;

  546.                 NoteVoicePar[nvoice].FMVolume =

  547.                     pars.VoicePar[nvoice].PFMVolume

  548.                     / 127.0f * fmvoldamp;

  549.         }

  550. 

  551.         //Voice's modulator velocity sensing

  552.         NoteVoicePar[nvoice].FMVolume *=

  553.             VelF(velocity,

  554.                  pars.VoicePar[nvoice].PFMVelocityScaleFunction);

  555. 

  556.         NoteVoicePar[nvoice].DelayTicks =

  557.             (int)((expf(pars.VoicePar[nvoice].PDelay / 127.0f

  558.                         * logf(50.0f))

  559.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  560.     }

  561. 

  562.     ///    initparameters();

  563. 

  564.     ///////////////

  565.     // Altered content of initparameters():

  566. 

  567.     int tmp[NUM_VOICES];

  568. 

  569.     NoteGlobalPar.Volume = 4.0f

  570.                            * powf(0.1f, 3.0f

  571.                                   * (1.0f - pars.GlobalPar.PVolume

  572.                                      / 96.0f))                                      //-60 dB .. 0 dB

  573.                            * VelF(

  574.         velocity,

  575.         pars.GlobalPar.PAmpVelocityScaleFunction); //velocity sensing

  576. 

  577.     globalnewamplitude = NoteGlobalPar.Volume

  578.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

  579.                          * NoteGlobalPar.AmpLfo->amplfoout();

  580. 

  581.     NoteGlobalPar.FilterQ = pars.GlobalPar.GlobalFilter->getq();

  582.     NoteGlobalPar.FilterFreqTracking =

  583.         pars.GlobalPar.GlobalFilter->getfreqtracking(basefreq);

  584. 

  585.     // Forbids the Modulation Voice to be greater or equal than voice

  586.     for(int i = 0; i < NUM_VOICES; ++i)

  587.         if(NoteVoicePar[i].FMVoice >= i)

  588.             NoteVoicePar[i].FMVoice = -1;

  589. 

  590.     // Voice Parameter init

  591.     for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  592.         if(NoteVoicePar[nvoice].Enabled == 0)

  593.             continue;

  594. 

  595.         NoteVoicePar[nvoice].noisetype = pars.VoicePar[nvoice].Type;

  596.         /* Voice Amplitude Parameters Init */

  597.         NoteVoicePar[nvoice].Volume =

  598.             powf(0.1f, 3.0f

  599.                  * (1.0f - pars.VoicePar[nvoice].PVolume / 127.0f))             // -60 dB .. 0 dB

  600.             * VelF(velocity,

  601.                    pars.VoicePar[nvoice].PAmpVelocityScaleFunction); //velocity

  602. 

  603.         if(pars.VoicePar[nvoice].PVolumeminus != 0)

  604.             NoteVoicePar[nvoice].Volume = -NoteVoicePar[nvoice].Volume;

  605. 

  606.         if(pars.VoicePar[nvoice].PPanning == 0)

  607.             NoteVoicePar[nvoice].Panning = RND;  // random panning

  608.         else

  609.             NoteVoicePar[nvoice].Panning =

  610.                 pars.VoicePar[nvoice].PPanning / 128.0f;

  611. 

  612.         newamplitude[nvoice] = 1.0f;

  613.         if(pars.VoicePar[nvoice].PAmpEnvelopeEnabled

  614.            && NoteVoicePar[nvoice].AmpEnvelope)

  615.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpEnvelope->envout_dB();

  616. 

  617. 

  618.         if(pars.VoicePar[nvoice].PAmpLfoEnabled && NoteVoicePar[nvoice].AmpLfo)

  619.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpLfo->amplfoout();

  620. 

  621. 

  622. 

  623.         NoteVoicePar[nvoice].FilterFreqTracking =

  624.             pars.VoicePar[nvoice].VoiceFilter->getfreqtracking(basefreq);

  625. 

  626.         /* Voice Modulation Parameters Init */

  627.         if((NoteVoicePar[nvoice].FMEnabled != NONE)

  628.            && (NoteVoicePar[nvoice].FMVoice < 0)) {

  629.             pars.VoicePar[nvoice].FMSmp->newrandseed(prng());

  630. 

  631.             //Perform Anti-aliasing only on MORPH or RING MODULATION

  632. 

  633.             int vc = nvoice;

  634.             if(pars.VoicePar[nvoice].PextFMoscil != -1)

  635.                 vc = pars.VoicePar[nvoice].PextFMoscil;

  636. 

  637.             if(!pars.GlobalPar.Hrandgrouping)

  638.                 pars.VoicePar[vc].FMSmp->newrandseed(prng());

  639. 

  640.             for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)

  641.                 NoteVoicePar[nvoice].FMSmp[synth.oscilsize + i] =

  642.                     NoteVoicePar[nvoice].FMSmp[i];

  643.         }

  644. 

  645.         FMnewamplitude[nvoice] = NoteVoicePar[nvoice].FMVolume

  646.                                  * ctl.fmamp.relamp;

  647. 

  648.         if(pars.VoicePar[nvoice].PFMAmpEnvelopeEnabled

  649.            && NoteVoicePar[nvoice].FMAmpEnvelope)

  650.             FMnewamplitude[nvoice] *=

  651.                 NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();

  652.     }

  653. 

  654.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  655.         for(unsigned i = nvoice + 1; i < NUM_VOICES; ++i)

  656.             tmp[i] = 0;

  657.         for(unsigned i = nvoice + 1; i < NUM_VOICES; ++i)

  658.             if((NoteVoicePar[i].FMVoice == nvoice) && (tmp[i] == 0))

  659.                 tmp[i] = 1;

  660.     }

  661. }

  662. 

  663. 

  664. /*

  665.  * Kill a voice of ADnote

  666.  */

  667. void ADnote::KillVoice(int nvoice)

  668. {

  669.     memory.devalloc(oscfreqhi[nvoice]);

  670.     memory.devalloc(oscfreqlo[nvoice]);

  671.     memory.devalloc(oscfreqhiFM[nvoice]);

  672.     memory.devalloc(oscfreqloFM[nvoice]);

  673.     memory.devalloc(oscposhi[nvoice]);

  674.     memory.devalloc(oscposlo[nvoice]);

  675.     memory.devalloc(oscposhiFM[nvoice]);

  676.     memory.devalloc(oscposloFM[nvoice]);

  677. 

  678.     memory.devalloc(unison_base_freq_rap[nvoice]);

  679.     memory.devalloc(unison_freq_rap[nvoice]);

  680.     memory.devalloc(unison_invert_phase[nvoice]);

  681.     memory.devalloc(FMoldsmp[nvoice]);

  682.     memory.devalloc(unison_vibratto[nvoice].step);

  683.     memory.devalloc(unison_vibratto[nvoice].position);

  684. 

  685.     NoteVoicePar[nvoice].kill(memory, synth);

  686. }

  687. 

  688. /*

  689.  * Kill the note

  690.  */

  691. void ADnote::KillNote()

  692. {

  693.     for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  694.         if(NoteVoicePar[nvoice].Enabled == ON)

  695.             KillVoice(nvoice);

  696. 

  697.         if(NoteVoicePar[nvoice].VoiceOut)

  698.             memory.dealloc(NoteVoicePar[nvoice].VoiceOut);

  699.     }

  700. 

  701.     NoteGlobalPar.kill(memory);

  702. 

  703.     NoteEnabled = OFF;

  704. }

  705. 

  706. ADnote::~ADnote()

  707. {

  708.     if(NoteEnabled == ON)

  709.         KillNote();

  710.     memory.devalloc(tmpwavel);

  711.     memory.devalloc(tmpwaver);

  712.     memory.devalloc(bypassl);

  713.     memory.devalloc(bypassr);

  714.     for(int k = 0; k < max_unison; ++k)

  715.         memory.devalloc(tmpwave_unison[k]);

  716.     memory.devalloc(tmpwave_unison);

  717. }

  718. 

  719. 

  720. /*

  721.  * Init the parameters

  722.  */

  723. void ADnote::initparameters()

  724. {

  725.     int tmp[NUM_VOICES];

  726.     //ADnoteParameters &pars = *partparams;

  727. 

  728.     // Global Parameters

  729.     NoteGlobalPar.initparameters(pars.GlobalPar, synth,

  730.                                  time,

  731.                                  memory, basefreq, velocity,

  732.                                  stereo);

  733. 

  734.     NoteGlobalPar.AmpEnvelope->envout_dB(); //discard the first envelope output

  735.     globalnewamplitude = NoteGlobalPar.Volume

  736.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

  737.                          * NoteGlobalPar.AmpLfo->amplfoout();

  738. 

  739.     // Forbids the Modulation Voice to be greater or equal than voice

  740.     for(int i = 0; i < NUM_VOICES; ++i)

  741.         if(NoteVoicePar[i].FMVoice >= i)

  742.             NoteVoicePar[i].FMVoice = -1;

  743. 

  744.     // Voice Parameter init

  745.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  746.         Voice &vce = NoteVoicePar[nvoice];

  747.         ADnoteVoiceParam &param = pars.VoicePar[nvoice];

  748. 

  749.         if(vce.Enabled == 0)

  750.             continue;

  751. 

  752.         vce.noisetype = param.Type;

  753.         /* Voice Amplitude Parameters Init */

  754.         vce.Volume = powf(0.1f, 3.0f * (1.0f - param.PVolume / 127.0f)) // -60dB..0dB

  755.                      * VelF(velocity, param.PAmpVelocityScaleFunction);

  756. 

  757.         if(param.PVolumeminus)

  758.             vce.Volume = -vce.Volume;

  759. 

  760.         if(param.PPanning == 0)

  761.             vce.Panning = RND;  // random panning

  762.         else

  763.             vce.Panning = param.PPanning / 128.0f;

  764. 

  765.         newamplitude[nvoice] = 1.0f;

  766.         if(param.PAmpEnvelopeEnabled) {

  767.             vce.AmpEnvelope = memory.alloc<Envelope>(*param.AmpEnvelope, basefreq, synth.dt());

  768.             vce.AmpEnvelope->envout_dB(); //discard the first envelope sample

  769.             newamplitude[nvoice] *= vce.AmpEnvelope->envout_dB();

  770.         }

  771. 

  772.         if(param.PAmpLfoEnabled) {

  773.             vce.AmpLfo = memory.alloc<LFO>(*param.AmpLfo, basefreq, time);

  774.             newamplitude[nvoice] *= vce.AmpLfo->amplfoout();

  775.         }

  776. 

  777.         /* Voice Frequency Parameters Init */

  778.         if(param.PFreqEnvelopeEnabled)

  779.             vce.FreqEnvelope = memory.alloc<Envelope>(*param.FreqEnvelope, basefreq, synth.dt());

  780. 

  781.         if(param.PFreqLfoEnabled)

  782.             vce.FreqLfo = memory.alloc<LFO>(*param.FreqLfo, basefreq, time);

  783. 

  784.         /* Voice Filter Parameters Init */

  785.         if(param.PFilterEnabled != 0) {

  786.             vce.VoiceFilterL = Filter::generate(memory, param.VoiceFilter,

  787.                     synth.samplerate, synth.buffersize);

  788.             vce.VoiceFilterR = Filter::generate(memory, param.VoiceFilter,

  789.                     synth.samplerate, synth.buffersize);

  790.         }

  791. 

  792.         if(param.PFilterEnvelopeEnabled)

  793.             vce.FilterEnvelope = memory.alloc<Envelope>(*param.FilterEnvelope, basefreq, synth.dt());

  794. 

  795.         if(param.PFilterLfoEnabled)

  796.             vce.FilterLfo = memory.alloc<LFO>(*param.FilterLfo, basefreq, time);

  797. 

  798.         vce.FilterFreqTracking =

  799.             param.VoiceFilter->getfreqtracking(basefreq);

  800. 

  801.         /* Voice Modulation Parameters Init */

  802.         if((vce.FMEnabled != NONE) && (vce.FMVoice < 0)) {

  803.             param.FMSmp->newrandseed(prng());

  804.             vce.FMSmp = memory.valloc<float>(synth.oscilsize + OSCIL_SMP_EXTRA_SAMPLES);

  805. 

  806.             //Perform Anti-aliasing only on MORPH or RING MODULATION

  807. 

  808.             int vc = nvoice;

  809.             if(param.PextFMoscil != -1)

  810.                 vc = param.PextFMoscil;

  811. 

  812.             float tmp = 1.0f;

  813.             if((pars.VoicePar[vc].FMSmp->Padaptiveharmonics != 0)

  814.                || (vce.FMEnabled == MORPH)

  815.                || (vce.FMEnabled == RING_MOD))

  816.                 tmp = getFMvoicebasefreq(nvoice);

  817. 

  818.             if(!pars.GlobalPar.Hrandgrouping)

  819.                 pars.VoicePar[vc].FMSmp->newrandseed(prng());

  820. 

  821.             for(int k = 0; k < unison_size[nvoice]; ++k)

  822.                 oscposhiFM[nvoice][k] = (oscposhi[nvoice][k]

  823.                                          + pars.VoicePar[vc].FMSmp->get(

  824.                                              vce.FMSmp, tmp))

  825.                                         % synth.oscilsize;

  826. 

  827.             for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)

  828.                 vce.FMSmp[synth.oscilsize + i] = vce.FMSmp[i];

  829.             int oscposhiFM_add =

  830.                 (int)((param.PFMoscilphase

  831.                        - 64.0f) / 128.0f * synth.oscilsize

  832.                       + synth.oscilsize * 4);

  833.             for(int k = 0; k < unison_size[nvoice]; ++k) {

  834.                 oscposhiFM[nvoice][k] += oscposhiFM_add;

  835.                 oscposhiFM[nvoice][k] %= synth.oscilsize;

  836.             }

  837.         }

  838. 

  839.         if(param.PFMFreqEnvelopeEnabled)

  840.             vce.FMFreqEnvelope = memory.alloc<Envelope>(*param.FMFreqEnvelope, basefreq, synth.dt());

  841. 

  842.         FMnewamplitude[nvoice] = vce.FMVolume * ctl.fmamp.relamp;

  843. 

  844.         if(param.PFMAmpEnvelopeEnabled ) {

  845.             vce.FMAmpEnvelope =

  846.                 memory.alloc<Envelope>(*param.FMAmpEnvelope, basefreq, synth.dt());

  847.             FMnewamplitude[nvoice] *= vce.FMAmpEnvelope->envout_dB();

  848.         }

  849.     }

  850. 

  851.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  852.         for(int i = nvoice + 1; i < NUM_VOICES; ++i)

  853.             tmp[i] = 0;

  854.         for(int i = nvoice + 1; i < NUM_VOICES; ++i)

  855.             if((NoteVoicePar[i].FMVoice == nvoice) && (tmp[i] == 0)) {

  856.                 NoteVoicePar[nvoice].VoiceOut =

  857.                     memory.valloc<float>(synth.buffersize);

  858.                 tmp[i] = 1;

  859.             }

  860. 

  861.         if(NoteVoicePar[nvoice].VoiceOut)

  862.             memset(NoteVoicePar[nvoice].VoiceOut, 0, synth.bufferbytes);

  863.     }

  864. }

  865. 

  866. 

  867. /*

  868.  * Computes the relative frequency of each unison voice and it's vibratto

  869.  * This must be called before setfreq* functions

  870.  */

  871. void ADnote::compute_unison_freq_rap(int nvoice) {

  872.     if(unison_size[nvoice] == 1) { //no unison

  873.         unison_freq_rap[nvoice][0] = 1.0f;

  874.         return;

  875.     }

  876.     float relbw = ctl.bandwidth.relbw * bandwidthDetuneMultiplier;

  877.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  878.         float pos  = unison_vibratto[nvoice].position[k];

  879.         float step = unison_vibratto[nvoice].step[k];

  880.         pos += step;

  881.         if(pos <= -1.0f) {

  882.             pos  = -1.0f;

  883.             step = -step;

  884.         }

  885.         if(pos >= 1.0f) {

  886.             pos  = 1.0f;

  887.             step = -step;

  888.         }

  889.         float vibratto_val = (pos - 0.333333333f * pos * pos * pos) * 1.5f; //make the vibratto lfo smoother

  890.         unison_freq_rap[nvoice][k] = 1.0f

  891.                                      + ((unison_base_freq_rap[nvoice][k]

  892.                                          - 1.0f) + vibratto_val

  893.                                         * unison_vibratto[nvoice].amplitude)

  894.                                      * relbw;

  895. 

  896.         unison_vibratto[nvoice].position[k] = pos;

  897.         step = unison_vibratto[nvoice].step[k] = step;

  898.     }

  899. }

  900. 

  901. 

  902. /*

  903.  * Computes the frequency of an oscillator

  904.  */

  905. void ADnote::setfreq(int nvoice, float in_freq)

  906. {

  907.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  908.         float freq  = fabs(in_freq) * unison_freq_rap[nvoice][k];

  909.         float speed = freq * synth.oscilsize_f / synth.samplerate_f;

  910.         if(speed > synth.oscilsize_f)

  911.             speed = synth.oscilsize_f;

  912. 

  913.         F2I(speed, oscfreqhi[nvoice][k]);

  914.         oscfreqlo[nvoice][k] = speed - floor(speed);

  915.     }

  916. }

  917. 

  918. /*

  919.  * Computes the frequency of an modullator oscillator

  920.  */

  921. void ADnote::setfreqFM(int nvoice, float in_freq)

  922. {

  923.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  924.         float freq  = fabs(in_freq) * unison_freq_rap[nvoice][k];

  925.         float speed = freq * synth.oscilsize_f / synth.samplerate_f;

  926.         if(speed > synth.samplerate_f)

  927.             speed = synth.samplerate_f;

  928. 

  929.         F2I(speed, oscfreqhiFM[nvoice][k]);

  930.         oscfreqloFM[nvoice][k] = speed - floor(speed);

  931.     }

  932. }

  933. 

  934. /*

  935.  * Get Voice base frequency

  936.  */

  937. float ADnote::getvoicebasefreq(int nvoice) const

  938. {

  939.     float detune = NoteVoicePar[nvoice].Detune / 100.0f

  940.                    + NoteVoicePar[nvoice].FineDetune / 100.0f

  941.                    * ctl.bandwidth.relbw * bandwidthDetuneMultiplier

  942.                    + NoteGlobalPar.Detune / 100.0f;

  943. 

  944.     if(NoteVoicePar[nvoice].fixedfreq == 0)

  945.         return this->basefreq * powf(2, detune / 12.0f);

  946.     else { //the fixed freq is enabled

  947.         float fixedfreq   = 440.0f;

  948.         int   fixedfreqET = NoteVoicePar[nvoice].fixedfreqET;

  949.         if(fixedfreqET != 0) { //if the frequency varies according the keyboard note

  950.             float tmp =

  951.                 (midinote

  952.                  - 69.0f) / 12.0f

  953.                 * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);

  954.             if(fixedfreqET <= 64)

  955.                 fixedfreq *= powf(2.0f, tmp);

  956.             else

  957.                 fixedfreq *= powf(3.0f, tmp);

  958.         }

  959.         return fixedfreq * powf(2.0f, detune / 12.0f);

  960.     }

  961. }

  962. 

  963. /*

  964.  * Get Voice's Modullator base frequency

  965.  */

  966. float ADnote::getFMvoicebasefreq(int nvoice) const

  967. {

  968.     float detune = NoteVoicePar[nvoice].FMDetune / 100.0f;

  969.     return getvoicebasefreq(nvoice) * powf(2, detune / 12.0f);

  970. }

  971. 

  972. /*

  973.  * Computes all the parameters for each tick

  974.  */

  975. void ADnote::computecurrentparameters()

  976. {

  977.     int   nvoice;

  978.     float voicefreq, voicepitch, filterpitch, filterfreq, FMfreq,

  979.           FMrelativepitch, globalpitch, globalfilterpitch;

  980.     globalpitch = 0.01f * (NoteGlobalPar.FreqEnvelope->envout()

  981.                            + NoteGlobalPar.FreqLfo->lfoout()

  982.                            * ctl.modwheel.relmod);

  983.     globaloldamplitude = globalnewamplitude;

  984.     globalnewamplitude = NoteGlobalPar.Volume

  985.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

  986.                          * NoteGlobalPar.AmpLfo->amplfoout();

  987. 

  988.     globalfilterpitch = NoteGlobalPar.FilterEnvelope->envout()

  989.                         + NoteGlobalPar.FilterLfo->lfoout()

  990.                         + NoteGlobalPar.FilterCenterPitch;

  991. 

  992.     float tmpfilterfreq = globalfilterpitch + ctl.filtercutoff.relfreq

  993.                           + NoteGlobalPar.FilterFreqTracking;

  994. 

  995.     tmpfilterfreq = Filter::getrealfreq(tmpfilterfreq);

  996. 

  997.     float globalfilterq = NoteGlobalPar.FilterQ * ctl.filterq.relq;

  998.     NoteGlobalPar.GlobalFilterL->setfreq_and_q(tmpfilterfreq, globalfilterq);

  999.     if(stereo != 0)

  1000.         NoteGlobalPar.GlobalFilterR->setfreq_and_q(tmpfilterfreq, globalfilterq);

  1001. 

  1002.     //compute the portamento, if it is used by this note

  1003.     float portamentofreqrap = 1.0f;

  1004.     if(portamento != 0) { //this voice use portamento

  1005.         portamentofreqrap = ctl.portamento.freqrap;

  1006.         if(ctl.portamento.used == 0) //the portamento has finished

  1007.             portamento = 0;  //this note is no longer "portamented"

  1008.     }

  1009. 

  1010.     //compute parameters for all voices

  1011.     for(nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1012.         if(NoteVoicePar[nvoice].Enabled != ON)

  1013.             continue;

  1014.         NoteVoicePar[nvoice].DelayTicks -= 1;

  1015.         if(NoteVoicePar[nvoice].DelayTicks > 0)

  1016.             continue;

  1017. 

  1018.         compute_unison_freq_rap(nvoice);

  1019. 

  1020.         /*******************/

  1021.         /* Voice Amplitude */

  1022.         /*******************/

  1023.         oldamplitude[nvoice] = newamplitude[nvoice];

  1024.         newamplitude[nvoice] = 1.0f;

  1025. 

  1026.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1027.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpEnvelope->envout_dB();

  1028. 

  1029.         if(NoteVoicePar[nvoice].AmpLfo)

  1030.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpLfo->amplfoout();

  1031. 

  1032.         /****************/

  1033.         /* Voice Filter */

  1034.         /****************/

  1035.         if(NoteVoicePar[nvoice].VoiceFilterL) {

  1036.             filterpitch = NoteVoicePar[nvoice].FilterCenterPitch;

  1037. 

  1038.             if(NoteVoicePar[nvoice].FilterEnvelope)

  1039.                 filterpitch += NoteVoicePar[nvoice].FilterEnvelope->envout();

  1040. 

  1041.             if(NoteVoicePar[nvoice].FilterLfo)

  1042.                 filterpitch += NoteVoicePar[nvoice].FilterLfo->lfoout();

  1043. 

  1044.             filterfreq = filterpitch + NoteVoicePar[nvoice].FilterFreqTracking;

  1045.             filterfreq = Filter::getrealfreq(filterfreq);

  1046. 

  1047.             NoteVoicePar[nvoice].VoiceFilterL->setfreq(filterfreq);

  1048.             if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1049.                 NoteVoicePar[nvoice].VoiceFilterR->setfreq(filterfreq);

  1050.         }

  1051. 

  1052.         if(NoteVoicePar[nvoice].noisetype == 0) { //compute only if the voice isn't noise

  1053.             /*******************/

  1054.             /* Voice Frequency */

  1055.             /*******************/

  1056.             voicepitch = 0.0f;

  1057.             if(NoteVoicePar[nvoice].FreqLfo)

  1058.                 voicepitch += NoteVoicePar[nvoice].FreqLfo->lfoout() / 100.0f

  1059.                               * ctl.bandwidth.relbw;

  1060. 

  1061.             if(NoteVoicePar[nvoice].FreqEnvelope)

  1062.                 voicepitch += NoteVoicePar[nvoice].FreqEnvelope->envout()

  1063.                               / 100.0f;

  1064.             voicefreq = getvoicebasefreq(nvoice)

  1065.                         * powf(2, (voicepitch + globalpitch) / 12.0f);                //Hz frequency

  1066.             voicefreq *= ctl.pitchwheel.relfreq; //change the frequency by the controller

  1067.             setfreq(nvoice, voicefreq * portamentofreqrap);

  1068. 

  1069.             /***************/

  1070.             /*  Modulator */

  1071.             /***************/

  1072.             if(NoteVoicePar[nvoice].FMEnabled != NONE) {

  1073.                 FMrelativepitch = NoteVoicePar[nvoice].FMDetune / 100.0f;

  1074.                 if(NoteVoicePar[nvoice].FMFreqEnvelope)

  1075.                     FMrelativepitch +=

  1076.                         NoteVoicePar[nvoice].FMFreqEnvelope->envout() / 100;

  1077.                 FMfreq =

  1078.                     powf(2.0f, FMrelativepitch

  1079.                          / 12.0f) * voicefreq * portamentofreqrap;

  1080.                 setfreqFM(nvoice, FMfreq);

  1081. 

  1082.                 FMoldamplitude[nvoice] = FMnewamplitude[nvoice];

  1083.                 FMnewamplitude[nvoice] = NoteVoicePar[nvoice].FMVolume

  1084.                                          * ctl.fmamp.relamp;

  1085.                 if(NoteVoicePar[nvoice].FMAmpEnvelope)

  1086.                     FMnewamplitude[nvoice] *=

  1087.                         NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();

  1088.             }

  1089.         }

  1090.     }

  1091. }

  1092. 

  1093. 

  1094. /*

  1095.  * Fadein in a way that removes clicks but keep sound "punchy"

  1096.  */

  1097. inline void ADnote::fadein(float *smps) const

  1098. {

  1099.     int zerocrossings = 0;

  1100.     for(int i = 1; i < synth.buffersize; ++i)

  1101.         if((smps[i - 1] < 0.0f) && (smps[i] > 0.0f))

  1102.             zerocrossings++;  //this is only the possitive crossings

  1103. 

  1104.     float tmp = (synth.buffersize_f - 1.0f) / (zerocrossings + 1) / 3.0f;

  1105.     if(tmp < 8.0f)

  1106.         tmp = 8.0f;

  1107.     tmp *= NoteGlobalPar.Fadein_adjustment;

  1108. 

  1109.     int n;

  1110.     F2I(tmp, n); //how many samples is the fade-in

  1111.     if(n > synth.buffersize)

  1112.         n = synth.buffersize;

  1113.     for(int i = 0; i < n; ++i) { //fade-in

  1114.         float tmp = 0.5f - cosf((float)i / (float) n * PI) * 0.5f;

  1115.         smps[i] *= tmp;

  1116.     }

  1117. }

  1118. 

  1119. /*

  1120.  * Computes the Oscillator (Without Modulation) - LinearInterpolation

  1121.  */

  1122. 

  1123. /* As the code here is a bit odd due to optimization, here is what happens

  1124.  * First the current possition and frequency are retrieved from the running

  1125.  * state. These are broken up into high and low portions to indicate how many

  1126.  * samples are skipped in one step and how many fractional samples are skipped.

  1127.  * Outside of this method the fractional samples are just handled with floating

  1128.  * point code, but that's a bit slower than it needs to be. In this code the low

  1129.  * portions are known to exist between 0.0 and 1.0 and it is known that they are

  1130.  * stored in single precision floating point IEEE numbers. This implies that

  1131.  * a maximum of 24 bits are significant. The below code does your standard

  1132.  * linear interpolation that you'll see throughout this codebase, but by

  1133.  * sticking to integers for tracking the overflow of the low portion, around 15%

  1134.  * of the execution time was shaved off in the ADnote test.

  1135.  */

  1136. inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)

  1137. {

  1138.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1139.         int    poshi  = oscposhi[nvoice][k];

  1140.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1141.         int    freqhi = oscfreqhi[nvoice][k];

  1142.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1143.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1144.         float *tw     = tmpwave_unison[k];

  1145.         assert(oscfreqlo[nvoice][k] < 1.0f);

  1146.         for(int i = 0; i < synth.buffersize; ++i) {

  1147.             tw[i]  = (smps[poshi] * ((1<<24) - poslo) + smps[poshi + 1] * poslo)/(1.0f*(1<<24));

  1148.             poslo += freqlo;

  1149.             poshi += freqhi + (poslo>>24);

  1150.             poslo &= 0xffffff;

  1151.             poshi &= synth.oscilsize - 1;

  1152.         }

  1153.         oscposhi[nvoice][k] = poshi;

  1154.         oscposlo[nvoice][k] = poslo/(1.0f*(1<<24));

  1155.     }

  1156. }

  1157. 

  1158. 

  1159. 

  1160. /*

  1161.  * Computes the Oscillator (Without Modulation) - CubicInterpolation

  1162.  *

  1163.  The differences from the Linear are to little to deserve to be used. This is because I am using a large synth.oscilsize (>512)

  1164. inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){

  1165.     int i,poshi;

  1166.     float poslo;

  1167. 

  1168.     poshi=oscposhi[nvoice];

  1169.     poslo=oscposlo[nvoice];

  1170.     float *smps=NoteVoicePar[nvoice].OscilSmp;

  1171.     float xm1,x0,x1,x2,a,b,c;

  1172.     for (i=0;i<synth.buffersize;i++){

  1173.     xm1=smps[poshi];

  1174.     x0=smps[poshi+1];

  1175.     x1=smps[poshi+2];

  1176.     x2=smps[poshi+3];

  1177.     a=(3.0f * (x0-x1) - xm1 + x2) / 2.0f;

  1178.     b = 2.0f*x1 + xm1 - (5.0f*x0 + x2) / 2.0f;

  1179.     c = (x1 - xm1) / 2.0f;

  1180.     tmpwave[i]=(((a * poslo) + b) * poslo + c) * poslo + x0;

  1181.     printf("a\n");

  1182.     //tmpwave[i]=smps[poshi]*(1.0f-poslo)+smps[poshi+1]*poslo;

  1183.     poslo+=oscfreqlo[nvoice];

  1184.     if (poslo>=1.0f) {

  1185.             poslo-=1.0f;

  1186.         poshi++;

  1187.     };

  1188.         poshi+=oscfreqhi[nvoice];

  1189.         poshi&=synth.oscilsize-1;

  1190.     };

  1191.     oscposhi[nvoice]=poshi;

  1192.     oscposlo[nvoice]=poslo;

  1193. };

  1194. */

  1195. /*

  1196.  * Computes the Oscillator (Morphing)

  1197.  */

  1198. inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)

  1199. {

  1200.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1201.     if(FMnewamplitude[nvoice] > 1.0f)

  1202.         FMnewamplitude[nvoice] = 1.0f;

  1203.     if(FMoldamplitude[nvoice] > 1.0f)

  1204.         FMoldamplitude[nvoice] = 1.0f;

  1205. 

  1206.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1207.         //if I use VoiceOut[] as modullator

  1208.         int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1209.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1210.             float *tw = tmpwave_unison[k];

  1211.             for(int i = 0; i < synth.buffersize; ++i) {

  1212.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1213.                                             FMnewamplitude[nvoice],

  1214.                                             i,

  1215.                                             synth.buffersize);

  1216.                 tw[i] = tw[i]

  1217.                     * (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1218.             }

  1219.         }

  1220.     }

  1221.     else

  1222.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1223.             int    poshiFM  = oscposhiFM[nvoice][k];

  1224.             float  posloFM  = oscposloFM[nvoice][k];

  1225.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1226.             float  freqloFM = oscfreqloFM[nvoice][k];

  1227.             float *tw = tmpwave_unison[k];

  1228. 

  1229.             for(int i = 0; i < synth.buffersize; ++i) {

  1230.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1231.                                             FMnewamplitude[nvoice],

  1232.                                             i,

  1233.                                             synth.buffersize);

  1234.                 tw[i] = tw[i] * (1.0f - amp) + amp

  1235.                         * (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1 - posloFM)

  1236.                            + NoteVoicePar[nvoice].FMSmp[poshiFM + 1] * posloFM);

  1237.                 posloFM += freqloFM;

  1238.                 if(posloFM >= 1.0f) {

  1239.                     posloFM -= 1.0f;

  1240.                     poshiFM++;

  1241.                 }

  1242.                 poshiFM += freqhiFM;

  1243.                 poshiFM &= synth.oscilsize - 1;

  1244.             }

  1245.             oscposhiFM[nvoice][k] = poshiFM;

  1246.             oscposloFM[nvoice][k] = posloFM;

  1247.         }

  1248. }

  1249. 

  1250. /*

  1251.  * Computes the Oscillator (Ring Modulation)

  1252.  */

  1253. inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)

  1254. {

  1255.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1256.     if(FMnewamplitude[nvoice] > 1.0f)

  1257.         FMnewamplitude[nvoice] = 1.0f;

  1258.     if(FMoldamplitude[nvoice] > 1.0f)

  1259.         FMoldamplitude[nvoice] = 1.0f;

  1260.     if(NoteVoicePar[nvoice].FMVoice >= 0)

  1261.         // if I use VoiceOut[] as modullator

  1262.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1263.             float *tw = tmpwave_unison[k];

  1264.             for(int i = 0; i < synth.buffersize; ++i) {

  1265.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1266.                                             FMnewamplitude[nvoice],

  1267.                                             i,

  1268.                                             synth.buffersize);

  1269.                 int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1270.                 tw[i] *= (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1271.             }

  1272.         }

  1273.     else

  1274.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1275.             int    poshiFM  = oscposhiFM[nvoice][k];

  1276.             float  posloFM  = oscposloFM[nvoice][k];

  1277.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1278.             float  freqloFM = oscfreqloFM[nvoice][k];

  1279.             float *tw = tmpwave_unison[k];

  1280. 

  1281.             for(int i = 0; i < synth.buffersize; ++i) {

  1282.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1283.                                             FMnewamplitude[nvoice],

  1284.                                             i,

  1285.                                             synth.buffersize);

  1286.                 tw[i] *= (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1.0f - posloFM)

  1287.                           + NoteVoicePar[nvoice].FMSmp[poshiFM

  1288.                                                        + 1] * posloFM) * amp

  1289.                          + (1.0f - amp);

  1290.                 posloFM += freqloFM;

  1291.                 if(posloFM >= 1.0f) {

  1292.                     posloFM -= 1.0f;

  1293.                     poshiFM++;

  1294.                 }

  1295.                 poshiFM += freqhiFM;

  1296.                 poshiFM &= synth.oscilsize - 1;

  1297.             }

  1298.             oscposhiFM[nvoice][k] = poshiFM;

  1299.             oscposloFM[nvoice][k] = posloFM;

  1300.         }

  1301. }

  1302. 

  1303. /*

  1304.  * Computes the Oscillator (Phase Modulation or Frequency Modulation)

  1305.  */

  1306. inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,

  1307.                                                               int FMmode)

  1308. {

  1309.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1310.         //if I use VoiceOut[] as modulator

  1311.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1312.             float *tw = tmpwave_unison[k];

  1313.             const float *smps = NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut;

  1314.             memcpy(tw, smps, synth.bufferbytes);

  1315.         }

  1316.     } else {

  1317.         //Compute the modulator and store it in tmpwave_unison[][]

  1318.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1319.             int    poshiFM  = oscposhiFM[nvoice][k];

  1320.             int    posloFM  = oscposloFM[nvoice][k]  * (1<<24);

  1321.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1322.             int    freqloFM = oscfreqloFM[nvoice][k] * (1<<24);

  1323.             float *tw = tmpwave_unison[k];

  1324.             const float *smps = NoteVoicePar[nvoice].FMSmp;

  1325. 

  1326.             for(int i = 0; i < synth.buffersize; ++i) {

  1327.                 tw[i] = (smps[poshiFM] * ((1<<24) - posloFM)

  1328.                      + smps[poshiFM + 1] * posloFM) / (1.0f*(1<<24));

  1329.                 posloFM += freqloFM;

  1330.                 if(posloFM >= (1<<24)) {

  1331.                     posloFM &= 0xffffff;//fmod(posloFM, 1.0f);

  1332.                     poshiFM++;

  1333.                 }

  1334.                 poshiFM += freqhiFM;

  1335.                 poshiFM &= synth.oscilsize - 1;

  1336.             }

  1337.             oscposhiFM[nvoice][k] = poshiFM;

  1338.             oscposloFM[nvoice][k] = posloFM/((1<<24)*1.0f);

  1339.         }

  1340.     }

  1341.     // Amplitude interpolation

  1342.     if(ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],

  1343.                                  FMnewamplitude[nvoice])) {

  1344.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1345.             float *tw = tmpwave_unison[k];

  1346.             for(int i = 0; i < synth.buffersize; ++i)

  1347.                 tw[i] *= INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1348.                                                FMnewamplitude[nvoice],

  1349.                                                i,

  1350.                                                synth.buffersize);

  1351.         }

  1352.     } else {

  1353.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1354.             float *tw = tmpwave_unison[k];

  1355.             for(int i = 0; i < synth.buffersize; ++i)

  1356.                 tw[i] *= FMnewamplitude[nvoice];

  1357.         }

  1358.     }

  1359. 

  1360. 

  1361.     //normalize: makes all sample-rates, oscil_sizes to produce same sound

  1362.     if(FMmode != 0) { //Frequency modulation

  1363.         const float normalize = synth.oscilsize_f / 262144.0f * 44100.0f

  1364.                           / synth.samplerate_f;

  1365.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1366.             float *tw    = tmpwave_unison[k];

  1367.             float  fmold = FMoldsmp[nvoice][k];

  1368.             for(int i = 0; i < synth.buffersize; ++i) {

  1369.                 fmold = fmod(fmold + tw[i] * normalize, synth.oscilsize);

  1370.                 tw[i] = fmold;

  1371.             }

  1372.             FMoldsmp[nvoice][k] = fmold;

  1373.         }

  1374.     }

  1375.     else {  //Phase modulation

  1376.         const float normalize = synth.oscilsize_f / 262144.0f;

  1377.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1378.             float *tw = tmpwave_unison[k];

  1379.             for(int i = 0; i < synth.buffersize; ++i)

  1380.                 tw[i] *= normalize;

  1381.         }

  1382.     }

  1383. 

  1384.     //do the modulation

  1385.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1386.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1387.         float *tw     = tmpwave_unison[k];

  1388.         int    poshi  = oscposhi[nvoice][k];

  1389.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1390.         int    freqhi = oscfreqhi[nvoice][k];

  1391.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1392. 

  1393.         for(int i = 0; i < synth.buffersize; ++i) {

  1394.             int FMmodfreqhi = 0;

  1395.             F2I(tw[i], FMmodfreqhi);

  1396.             float FMmodfreqlo = tw[i]-FMmodfreqhi;//fmod(tw[i] /*+ 0.0000000001f*/, 1.0f);

  1397.             if(FMmodfreqhi < 0)

  1398.                 FMmodfreqlo++;

  1399. 

  1400.             //carrier

  1401.             int carposhi = poshi + FMmodfreqhi;

  1402.             int carposlo = poslo + FMmodfreqlo;

  1403. 

  1404.             if(carposlo >= (1<<24)) {

  1405.                 carposhi++;

  1406.                 carposlo &= 0xffffff;//fmod(carposlo, 1.0f);

  1407.             }

  1408.             carposhi &= (synth.oscilsize - 1);

  1409. 

  1410.             tw[i] = (smps[carposhi] * ((1<<24) - carposlo)

  1411.                     + smps[carposhi + 1] * carposlo)/(1.0f*(1<<24));

  1412. 

  1413.             poslo += freqlo;

  1414.             if(poslo >= (1<<24)) {

  1415.                 poslo &= 0xffffff;//fmod(poslo, 1.0f);

  1416.                 poshi++;

  1417.             }

  1418. 

  1419.             poshi += freqhi;

  1420.             poshi &= synth.oscilsize - 1;

  1421.         }

  1422.         oscposhi[nvoice][k] = poshi;

  1423.         oscposlo[nvoice][k] = (poslo)/((1<<24)*1.0f);

  1424.     }

  1425. }

  1426. 

  1427. 

  1428. /*Calculeaza Oscilatorul cu PITCH MODULATION*/

  1429. inline void ADnote::ComputeVoiceOscillatorPitchModulation(int /*nvoice*/)

  1430. {

  1431. //TODO

  1432. }

  1433. 

  1434. /*

  1435.  * Computes the Noise

  1436.  */

  1437. inline void ADnote::ComputeVoiceWhiteNoise(int nvoice)

  1438. {

  1439.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1440.         float *tw = tmpwave_unison[k];

  1441.         for(int i = 0; i < synth.buffersize; ++i)

  1442.             tw[i] = RND * 2.0f - 1.0f;

  1443.     }

  1444. }

  1445. 

  1446. inline void ADnote::ComputeVoicePinkNoise(int nvoice)

  1447. {

  1448.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1449.         float *tw = tmpwave_unison[k];

  1450.         float *f = &pinking[nvoice][k > 0 ? 7 : 0];

  1451.         for(int i = 0; i < synth.buffersize; ++i) {

  1452. 	    float white = (RND-0.5)/4.0;

  1453. 	    f[0] = 0.99886*f[0]+white*0.0555179;

  1454. 	    f[1] = 0.99332*f[1]+white*0.0750759;

  1455. 	    f[2] = 0.96900*f[2]+white*0.1538520;

  1456. 	    f[3] = 0.86650*f[3]+white*0.3104856;

  1457. 	    f[4] = 0.55000*f[4]+white*0.5329522;

  1458. 	    f[5] = -0.7616*f[5]-white*0.0168980;

  1459. 	    tw[i] = f[0]+f[1]+f[2]+f[3]+f[4]+f[5]+f[6]+white*0.5362;

  1460. 	    f[6] = white*0.115926;

  1461.         }

  1462.     }

  1463. }

  1464. 

  1465. 

  1466. 

  1467. /*

  1468.  * Compute the ADnote samples

  1469.  * Returns 0 if the note is finished

  1470.  */

  1471. int ADnote::noteout(float *outl, float *outr)

  1472. {

  1473.     memcpy(outl, synth.denormalkillbuf, synth.bufferbytes);

  1474.     memcpy(outr, synth.denormalkillbuf, synth.bufferbytes);

  1475. 

  1476.     if(NoteEnabled == OFF)

  1477.         return 0;

  1478. 

  1479.     memset(bypassl, 0, synth.bufferbytes);

  1480.     memset(bypassr, 0, synth.bufferbytes);

  1481.     computecurrentparameters();

  1482. 

  1483.     for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1484.         if((NoteVoicePar[nvoice].Enabled != ON)

  1485.            || (NoteVoicePar[nvoice].DelayTicks > 0))

  1486.             continue;

  1487.         switch (NoteVoicePar[nvoice].noisetype) {

  1488.             case 0: //voice mode=sound

  1489.                 switch(NoteVoicePar[nvoice].FMEnabled) {

  1490.                     case MORPH:

  1491.                         ComputeVoiceOscillatorMorph(nvoice);

  1492.                         break;

  1493.                     case RING_MOD:

  1494.                         ComputeVoiceOscillatorRingModulation(nvoice);

  1495.                         break;

  1496.                     case PHASE_MOD:

  1497.                         ComputeVoiceOscillatorFrequencyModulation(nvoice, 0);

  1498.                         break;

  1499.                     case FREQ_MOD:

  1500.                         ComputeVoiceOscillatorFrequencyModulation(nvoice, 1);

  1501.                         break;

  1502.                     //case PITCH_MOD:ComputeVoiceOscillatorPitchModulation(nvoice);break;

  1503.                     default:

  1504.                         ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1505.                         //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice);

  1506.                 }

  1507.                 break;

  1508.             case 1:

  1509.                 ComputeVoiceWhiteNoise(nvoice);

  1510.                 break;

  1511.             default:

  1512.                 ComputeVoicePinkNoise(nvoice);

  1513.                 break;

  1514.         }

  1515.         // Voice Processing

  1516. 

  1517. 

  1518.         //mix subvoices into voice

  1519.         memset(tmpwavel, 0, synth.bufferbytes);

  1520.         if(stereo)

  1521.             memset(tmpwaver, 0, synth.bufferbytes);

  1522.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1523.             float *tw = tmpwave_unison[k];

  1524.             if(stereo) {

  1525.                 float stereo_pos = 0;

  1526.                 if(unison_size[nvoice] > 1)

  1527.                     stereo_pos = k

  1528.                                  / (float)(unison_size[nvoice]

  1529.                                            - 1) * 2.0f - 1.0f;

  1530.                 float stereo_spread = unison_stereo_spread[nvoice] * 2.0f; //between 0 and 2.0f

  1531.                 if(stereo_spread > 1.0f) {

  1532.                     float stereo_pos_1 = (stereo_pos >= 0.0f) ? 1.0f : -1.0f;

  1533.                     stereo_pos =

  1534.                         (2.0f

  1535.                          - stereo_spread) * stereo_pos

  1536.                         + (stereo_spread - 1.0f) * stereo_pos_1;

  1537.                 }

  1538.                 else

  1539.                     stereo_pos *= stereo_spread;

  1540. 

  1541.                 if(unison_size[nvoice] == 1)

  1542.                     stereo_pos = 0.0f;

  1543.                 float panning = (stereo_pos + 1.0f) * 0.5f;

  1544. 

  1545. 

  1546.                 float lvol = (1.0f - panning) * 2.0f;

  1547.                 if(lvol > 1.0f)

  1548.                     lvol = 1.0f;

  1549. 

  1550.                 float rvol = panning * 2.0f;

  1551.                 if(rvol > 1.0f)

  1552.                     rvol = 1.0f;

  1553. 

  1554.                 if(unison_invert_phase[nvoice][k]) {

  1555.                     lvol = -lvol;

  1556.                     rvol = -rvol;

  1557.                 }

  1558. 

  1559.                 for(int i = 0; i < synth.buffersize; ++i)

  1560.                     tmpwavel[i] += tw[i] * lvol;

  1561.                 for(int i = 0; i < synth.buffersize; ++i)

  1562.                     tmpwaver[i] += tw[i] * rvol;

  1563.             }

  1564.             else

  1565.                 for(int i = 0; i < synth.buffersize; ++i)

  1566.                     tmpwavel[i] += tw[i];

  1567.         }

  1568. 

  1569. 

  1570.         float unison_amplitude = 1.0f / sqrt(unison_size[nvoice]); //reduce the amplitude for large unison sizes

  1571.         // Amplitude

  1572.         float oldam = oldamplitude[nvoice] * unison_amplitude;

  1573.         float newam = newamplitude[nvoice] * unison_amplitude;

  1574. 

  1575.         if(ABOVE_AMPLITUDE_THRESHOLD(oldam, newam)) {

  1576.             int rest = synth.buffersize;

  1577.             //test if the amplitude if raising and the difference is high

  1578.             if((newam > oldam) && ((newam - oldam) > 0.25f)) {

  1579.                 rest = 10;

  1580.                 if(rest > synth.buffersize)

  1581.                     rest = synth.buffersize;

  1582.                 for(int i = 0; i < synth.buffersize - rest; ++i)

  1583.                     tmpwavel[i] *= oldam;

  1584.                 if(stereo)

  1585.                     for(int i = 0; i < synth.buffersize - rest; ++i)

  1586.                         tmpwaver[i] *= oldam;

  1587.             }

  1588.             // Amplitude interpolation

  1589.             for(int i = 0; i < rest; ++i) {

  1590.                 float amp = INTERPOLATE_AMPLITUDE(oldam, newam, i, rest);

  1591.                 tmpwavel[i + (synth.buffersize - rest)] *= amp;

  1592.                 if(stereo)

  1593.                     tmpwaver[i + (synth.buffersize - rest)] *= amp;

  1594.             }

  1595.         }

  1596.         else {

  1597.             for(int i = 0; i < synth.buffersize; ++i)

  1598.                 tmpwavel[i] *= newam;

  1599.             if(stereo)

  1600.                 for(int i = 0; i < synth.buffersize; ++i)

  1601.                     tmpwaver[i] *= newam;

  1602.         }

  1603. 

  1604.         // Fade in

  1605.         if(firsttick[nvoice] != 0) {

  1606.             fadein(&tmpwavel[0]);

  1607.             if(stereo)

  1608.                 fadein(&tmpwaver[0]);

  1609.             firsttick[nvoice] = 0;

  1610.         }

  1611. 

  1612. 

  1613.         // Filter

  1614.         if(NoteVoicePar[nvoice].VoiceFilterL)

  1615.             NoteVoicePar[nvoice].VoiceFilterL->filterout(&tmpwavel[0]);

  1616.         if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1617.             NoteVoicePar[nvoice].VoiceFilterR->filterout(&tmpwaver[0]);

  1618. 

  1619.         //check if the amplitude envelope is finished, if yes, the voice will be fadeout

  1620.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1621.             if(NoteVoicePar[nvoice].AmpEnvelope->finished()) {

  1622.                 for(int i = 0; i < synth.buffersize; ++i)

  1623.                     tmpwavel[i] *= 1.0f - (float)i / synth.buffersize_f;

  1624.                 if(stereo)

  1625.                     for(int i = 0; i < synth.buffersize; ++i)

  1626.                         tmpwaver[i] *= 1.0f - (float)i / synth.buffersize_f;

  1627.             }

  1628.         //the voice is killed later

  1629. 

  1630. 

  1631.         // Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator)

  1632.         if(NoteVoicePar[nvoice].VoiceOut) {

  1633.             if(stereo)

  1634.                 for(int i = 0; i < synth.buffersize; ++i)

  1635.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i]

  1636.                                                        + tmpwaver[i];

  1637.             else   //mono

  1638.                 for(int i = 0; i < synth.buffersize; ++i)

  1639.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i];

  1640.         }

  1641. 

  1642. 

  1643.         // Add the voice that do not bypass the filter to out

  1644.         if(NoteVoicePar[nvoice].filterbypass == 0) { //no bypass

  1645.             if(stereo)

  1646.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1647.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1648.                                * NoteVoicePar[nvoice].Panning * 2.0f;

  1649.                     outr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1650.                                * (1.0f - NoteVoicePar[nvoice].Panning) * 2.0f;

  1651.                 }

  1652.             else

  1653.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1654.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1655.         }

  1656.         else {  //bypass the filter

  1657.             if(stereo)

  1658.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1659.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1660.                                   * NoteVoicePar[nvoice].Panning * 2.0f;

  1661.                     bypassr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1662.                                   * (1.0f

  1663.                                      - NoteVoicePar[nvoice].Panning) * 2.0f;

  1664.                 }

  1665.             else

  1666.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1667.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1668.         }

  1669.         // chech if there is necesary to proces the voice longer (if the Amplitude envelope isn't finished)

  1670.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1671.             if(NoteVoicePar[nvoice].AmpEnvelope->finished())

  1672.                 KillVoice(nvoice);

  1673.     }

  1674. 

  1675. 

  1676.     //Processing Global parameters

  1677.     NoteGlobalPar.GlobalFilterL->filterout(&outl[0]);

  1678. 

  1679.     if(stereo == 0) { //set the right channel=left channel

  1680.         memcpy(outr, outl, synth.bufferbytes);

  1681.         memcpy(bypassr, bypassl, synth.bufferbytes);

  1682.     }

  1683.     else

  1684.         NoteGlobalPar.GlobalFilterR->filterout(&outr[0]);

  1685. 

  1686.     for(int i = 0; i < synth.buffersize; ++i) {

  1687.         outl[i] += bypassl[i];

  1688.         outr[i] += bypassr[i];

  1689.     }

  1690. 

  1691.     if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))

  1692.         // Amplitude Interpolation

  1693.         for(int i = 0; i < synth.buffersize; ++i) {

  1694.             float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,

  1695.                                                  globalnewamplitude,

  1696.                                                  i,

  1697.                                                  synth.buffersize);

  1698.             outl[i] *= tmpvol * NoteGlobalPar.Panning;

  1699.             outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);

  1700.         }

  1701.     else

  1702.         for(int i = 0; i < synth.buffersize; ++i) {

  1703.             outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;

  1704.             outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);

  1705.         }

  1706. 

  1707.     //Apply the punch

  1708.     if(NoteGlobalPar.Punch.Enabled != 0)

  1709.         for(int i = 0; i < synth.buffersize; ++i) {

  1710.             float punchamp = NoteGlobalPar.Punch.initialvalue

  1711.                              * NoteGlobalPar.Punch.t + 1.0f;

  1712.             outl[i] *= punchamp;

  1713.             outr[i] *= punchamp;

  1714.             NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;

  1715.             if(NoteGlobalPar.Punch.t < 0.0f) {

  1716.                 NoteGlobalPar.Punch.Enabled = 0;

  1717.                 break;

  1718.             }

  1719.         }

  1720. 

  1721. 

  1722.     // Apply legato-specific sound signal modifications

  1723.     legato.apply(*this, outl, outr);

  1724. 

  1725. 

  1726.     // Check if the global amplitude is finished.

  1727.     // If it does, disable the note

  1728.     if(NoteGlobalPar.AmpEnvelope->finished()) {

  1729.         for(int i = 0; i < synth.buffersize; ++i) { //fade-out

  1730.             float tmp = 1.0f - (float)i / synth.buffersize_f;

  1731.             outl[i] *= tmp;

  1732.             outr[i] *= tmp;

  1733.         }

  1734.         KillNote();

  1735.     }

  1736.     return 1;

  1737. }

  1738. 

  1739. 

  1740. /*

  1741.  * Release the key (NoteOff)

  1742.  */

  1743. void ADnote::releasekey()

  1744. {

  1745.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)

  1746.         NoteVoicePar[nvoice].releasekey();

  1747.     NoteGlobalPar.FreqEnvelope->releasekey();

  1748.     NoteGlobalPar.FilterEnvelope->releasekey();

  1749.     NoteGlobalPar.AmpEnvelope->releasekey();

  1750. }

  1751. 

  1752. /*

  1753.  * Check if the note is finished

  1754.  */

  1755. int ADnote::finished() const

  1756. {

  1757.     if(NoteEnabled == ON)

  1758.         return 0;

  1759.     else

  1760.         return 1;

  1761. }

  1762. 

  1763. void ADnote::Voice::releasekey()

  1764. {

  1765.     if(!Enabled)

  1766.         return;

  1767.     if(AmpEnvelope)

  1768.         AmpEnvelope->releasekey();

  1769.     if(FreqEnvelope)

  1770.         FreqEnvelope->releasekey();

  1771.     if(FilterEnvelope)

  1772.         FilterEnvelope->releasekey();

  1773.     if(FMFreqEnvelope)

  1774.         FMFreqEnvelope->releasekey();

  1775.     if(FMAmpEnvelope)

  1776.         FMAmpEnvelope->releasekey();

  1777. }

  1778. 

  1779. void ADnote::Voice::kill(Allocator &memory, const SYNTH_T &synth)

  1780. {

  1781.     memory.devalloc(OscilSmp);

  1782.     memory.dealloc(FreqEnvelope);

  1783.     memory.dealloc(FreqLfo);

  1784.     memory.dealloc(AmpEnvelope);

  1785.     memory.dealloc(AmpLfo);

  1786.     memory.dealloc(VoiceFilterL);

  1787.     memory.dealloc(VoiceFilterR);

  1788.     memory.dealloc(FilterEnvelope);

  1789.     memory.dealloc(FilterLfo);

  1790.     memory.dealloc(FMFreqEnvelope);

  1791.     memory.dealloc(FMAmpEnvelope);

  1792. 

  1793.     if((FMEnabled != NONE) && (FMVoice < 0))

  1794.         memory.devalloc(FMSmp);

  1795. 

  1796.     if(VoiceOut)

  1797.         memset(VoiceOut, 0, synth.bufferbytes);

  1798.     //the buffer can't be safely deleted as it may be

  1799.     //an input to another voice

  1800. 

  1801.     Enabled = OFF;

  1802. }

  1803. 

  1804. void ADnote::Global::kill(Allocator &memory)

  1805. {

  1806.     memory.dealloc(FreqEnvelope);

  1807.     memory.dealloc(FreqLfo);

  1808.     memory.dealloc(AmpEnvelope);

  1809.     memory.dealloc(AmpLfo);

  1810.     memory.dealloc(GlobalFilterL);

  1811.     memory.dealloc(GlobalFilterR);

  1812.     memory.dealloc(FilterEnvelope);

  1813.     memory.dealloc(FilterLfo);

  1814. }

  1815. 

  1816. void ADnote::Global::initparameters(const ADnoteGlobalParam &param,

  1817.                                     const SYNTH_T &synth,

  1818.                                     const AbsTime &time,

  1819.                                     class Allocator &memory,

  1820.                                     float basefreq, float velocity,

  1821.                                     bool stereo)

  1822. {

  1823.     FreqEnvelope = memory.alloc<Envelope>(*param.FreqEnvelope, basefreq, synth.dt());

  1824.     FreqLfo      = memory.alloc<LFO>(*param.FreqLfo, basefreq, time);

  1825. 

  1826.     AmpEnvelope = memory.alloc<Envelope>(*param.AmpEnvelope, basefreq, synth.dt());

  1827.     AmpLfo      = memory.alloc<LFO>(*param.AmpLfo, basefreq, time);

  1828. 

  1829.     Volume = 4.0f * powf(0.1f, 3.0f * (1.0f - param.PVolume / 96.0f)) //-60 dB .. 0 dB

  1830.              * VelF(velocity, param.PAmpVelocityScaleFunction);     //sensing

  1831. 

  1832.     GlobalFilterL = Filter::generate(memory, param.GlobalFilter,

  1833.             synth.samplerate, synth.buffersize);

  1834.     if(stereo)

  1835.         GlobalFilterR = Filter::generate(memory, param.GlobalFilter,

  1836.                 synth.samplerate, synth.buffersize);

  1837.     else

  1838.         GlobalFilterR = NULL;

  1839. 

  1840.     FilterEnvelope = memory.alloc<Envelope>(*param.FilterEnvelope, basefreq, synth.dt());

  1841.     FilterLfo      = memory.alloc<LFO>(*param.FilterLfo, basefreq, time);

  1842.     FilterQ = param.GlobalFilter->getq();

  1843.     FilterFreqTracking = param.GlobalFilter->getfreqtracking(basefreq);

  1844. }