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

  73.         NoteGlobalPar.Punch.Enabled = 1;

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

  75.         NoteGlobalPar.Punch.initialvalue =

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

  77.              * VelF(velocity,

  78.                     pars.GlobalPar.PPunchVelocitySensing));

  79.         float time =

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

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

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

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

  84.     }

  85.     else

  86.         NoteGlobalPar.Punch.Enabled = 0;

  87. 

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

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

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

  91. 

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

  93.         NoteVoicePar[nvoice].OscilSmp = NULL;

  94.         NoteVoicePar[nvoice].FMSmp    = NULL;

  95.         NoteVoicePar[nvoice].VoiceOut = NULL;

  96. 

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

  98.         unison_size[nvoice] = 1;

  99. 

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

  101.             NoteVoicePar[nvoice].Enabled = OFF;

  102.             continue; //the voice is disabled

  103.         }

  104. 

  105.         unison_stereo_spread[nvoice] =

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

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

  108.         if(unison < 1)

  109.             unison = 1;

  110. 

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

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

  113. 	    unison = 2;

  114. 

  115.         //compute unison

  116.         unison_size[nvoice] = unison;

  117. 

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

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

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

  121.         float unison_spread =

  122.             pars.getUnisonFrequencySpreadCents(nvoice);

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

  124.         float unison_vibratto_a  =

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

  126. 

  127. 

  128.         switch(unison) {

  129.             case 1:

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

  131.                 break;

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

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

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

  135.             };

  136.                 break;

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

  138.                 float unison_values[unison];

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

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

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

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

  143.                     unison_values[k] = val;

  144.                     if (min > val) {

  145.                         min = val;

  146.                     }

  147.                     if (max < val) {

  148.                         max = val;

  149.                     }

  150.                 }

  151.                 float diff = max - min;

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

  153.                     unison_values[k] =

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

  155.                     unison_base_freq_rap[nvoice][k] =

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

  157.                 }

  158.             };

  159.         }

  160. 

  161.         //unison vibrattos

  162.         if(unison > 1)

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

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

  165.                                                   + (unison_base_freq_rap[

  166.                                                          nvoice][k] - 1.0f)

  167.                                                   * (1.0f - unison_vibratto_a);

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

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

  170.         unison_vibratto[nvoice].amplitude =

  171.             (unison_real_spread - 1.0f) * unison_vibratto_a;

  172. 

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

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

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

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

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

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

  179.             float vibratto_period = vibratto_base_period

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

  181. 

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

  183.             if(RND < 0.5f)

  184.                 m = -m;

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

  186.         }

  187. 

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

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

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

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

  192.         }

  193. 

  194.         //phase invert for unison

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

  196.         if(unison != 1) {

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

  198.             switch(inv) {

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

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

  201.                     break;

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

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

  204.                     break;

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

  206.                         unison_invert_phase[nvoice][k] =

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

  208.                     break;

  209.             }

  210.         }

  211. 

  212. 

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

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

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

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

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

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

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

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

  221. 

  222.         NoteVoicePar[nvoice].Enabled     = ON;

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

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

  225. 

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

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

  228.             NoteVoicePar[nvoice].Detune = getdetune(

  229.                 pars.VoicePar[nvoice].PDetuneType,

  230.                 pars.VoicePar[nvoice].

  231.                 PCoarseDetune,

  232.                 8192); //coarse detune

  233.             NoteVoicePar[nvoice].FineDetune = getdetune(

  234.                 pars.VoicePar[nvoice].PDetuneType,

  235.                 0,

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

  237.         }

  238.         else {

  239.             NoteVoicePar[nvoice].Detune = getdetune(

  240.                 pars.GlobalPar.PDetuneType,

  241.                 pars.VoicePar[nvoice].

  242.                 PCoarseDetune,

  243.                 8192); //coarse detune

  244.             NoteVoicePar[nvoice].FineDetune = getdetune(

  245.                 pars.GlobalPar.PDetuneType,

  246.                 0,

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

  248.         }

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

  250.             NoteVoicePar[nvoice].FMDetune = getdetune(

  251.                 pars.VoicePar[nvoice].PFMDetuneType,

  252.                 pars.VoicePar[nvoice].

  253.                 PFMCoarseDetune,

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

  255.         else

  256.             NoteVoicePar[nvoice].FMDetune = getdetune(

  257.                 pars.GlobalPar.PDetuneType,

  258.                 pars.VoicePar[nvoice].

  259.                 PFMCoarseDetune,

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

  261. 

  262. 

  263. 

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

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

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

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

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

  269.         }

  270. 

  271.         //the extra points contains the first point

  272.         NoteVoicePar[nvoice].OscilSmp =

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

  274. 

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

  276.         int vc = nvoice;

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

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

  279.         if(!pars.GlobalPar.Hrandgrouping)

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

  281.         int oscposhi_start =

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

  283.                                              getvoicebasefreq(nvoice),

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

  285. 

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

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

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

  289.                                           + i] =

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

  291. 

  292.         oscposhi_start +=

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

  294.                    - 64.0f) / 128.0f * synth.oscilsize + synth.oscilsize * 4);

  295.         oscposhi_start %= synth.oscilsize;

  296. 

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

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

  299.             //put random starting point for other subvoices

  300.             oscposhi_start      =

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

  302.                         127.0f * (synth.oscilsize - 1));

  303.         }

  304. 

  305.         NoteVoicePar[nvoice].FreqLfo      = NULL;

  306.         NoteVoicePar[nvoice].FreqEnvelope = NULL;

  307. 

  308.         NoteVoicePar[nvoice].AmpLfo      = NULL;

  309.         NoteVoicePar[nvoice].AmpEnvelope = NULL;

  310. 

  311.         NoteVoicePar[nvoice].VoiceFilterL   = NULL;

  312.         NoteVoicePar[nvoice].VoiceFilterR   = NULL;

  313.         NoteVoicePar[nvoice].FilterEnvelope = NULL;

  314.         NoteVoicePar[nvoice].FilterLfo      = NULL;

  315. 

  316.         NoteVoicePar[nvoice].FilterCenterPitch =

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

  318.         NoteVoicePar[nvoice].filterbypass =

  319.             pars.VoicePar[nvoice].Pfilterbypass;

  320. 

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

  322.             case 1:

  323.                 NoteVoicePar[nvoice].FMEnabled = MORPH;

  324.                 break;

  325.             case 2:

  326.                 NoteVoicePar[nvoice].FMEnabled = RING_MOD;

  327.                 break;

  328.             case 3:

  329.                 NoteVoicePar[nvoice].FMEnabled = PHASE_MOD;

  330.                 break;

  331.             case 4:

  332.                 NoteVoicePar[nvoice].FMEnabled = FREQ_MOD;

  333.                 break;

  334.             case 5:

  335.                 NoteVoicePar[nvoice].FMEnabled = PITCH_MOD;

  336.                 break;

  337.             default:

  338.                 NoteVoicePar[nvoice].FMEnabled = NONE;

  339.         }

  340. 

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

  342.         NoteVoicePar[nvoice].FMFreqEnvelope = NULL;

  343.         NoteVoicePar[nvoice].FMAmpEnvelope  = NULL;

  344. 

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

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

  347.                                    nvoice),

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

  349.                                - 1.0f);

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

  351.             case PHASE_MOD:

  352.                 fmvoldamp =

  353.                     powf(440.0f / getvoicebasefreq(

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

  355.                          / 64.0f);

  356.                 NoteVoicePar[nvoice].FMVolume =

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

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

  359.                 break;

  360.             case FREQ_MOD:

  361.                 NoteVoicePar[nvoice].FMVolume =

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

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

  364.                 break;

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

  366.             //	          break;

  367.             default:

  368.                 if(fmvoldamp > 1.0f)

  369.                     fmvoldamp = 1.0f;

  370.                 NoteVoicePar[nvoice].FMVolume =

  371.                     pars.VoicePar[nvoice].PFMVolume

  372.                     / 127.0f * fmvoldamp;

  373.         }

  374. 

  375.         //Voice's modulator velocity sensing

  376.         NoteVoicePar[nvoice].FMVolume *=

  377.             VelF(velocity,

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

  379. 

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

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

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

  383. 

  384.         firsttick[nvoice] = 1;

  385.         NoteVoicePar[nvoice].DelayTicks =

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

  387.                         * logf(50.0f))

  388.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  389.     }

  390. 

  391.     max_unison = 1;

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

  393.         if(unison_size[nvoice] > max_unison)

  394.             max_unison = unison_size[nvoice];

  395. 

  396. 

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

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

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

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

  401.     }

  402. 

  403.     initparameters();

  404. }

  405. 

  406. SynthNote *ADnote::cloneLegato(void)

  407. {

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

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

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

  411. }

  412. 

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

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

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

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

  417. void ADnote::legatonote(LegatoParams lpars)

  418. {

  419.     //ADnoteParameters &pars = *partparams;

  420. 

  421.     // Manage legato stuff

  422.     if(legato.update(lpars))

  423.         return;

  424. 

  425.     portamento = lpars.portamento;

  426.     midinote   = lpars.midinote;

  427.     basefreq   = lpars.frequency;

  428. 

  429.     if(velocity > 1.0f)

  430.         velocity = 1.0f;

  431.     velocity = lpars.velocity;

  432. 

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

  434.                                      pars.GlobalPar.PCoarseDetune,

  435.                                      pars.GlobalPar.PDetune);

  436.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  437. 

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

  439.         NoteGlobalPar.Panning = RND;

  440.     else

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

  442. 

  443.     //center freq

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

  445.                                       + pars.GlobalPar.PFilterVelocityScale

  446.                                       / 127.0f * 6.0f          //velocity sensing

  447.                                       * (VelF(velocity,

  448.                                               pars.GlobalPar.

  449.                                               PFilterVelocityScaleFunction) - 1);

  450. 

  451. 

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

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

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

  455. 

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

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

  458. 

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

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

  461.             NoteVoicePar[nvoice].Detune = getdetune(

  462.                 pars.VoicePar[nvoice].PDetuneType,

  463.                 pars.VoicePar[nvoice].PCoarseDetune,

  464.                 8192); //coarse detune

  465.             NoteVoicePar[nvoice].FineDetune = getdetune(

  466.                 pars.VoicePar[nvoice].PDetuneType,

  467.                 0,

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

  469.         }

  470.         else {

  471.             NoteVoicePar[nvoice].Detune = getdetune(

  472.                 pars.GlobalPar.PDetuneType,

  473.                 pars.VoicePar[nvoice].PCoarseDetune,

  474.                 8192); //coarse detune

  475.             NoteVoicePar[nvoice].FineDetune = getdetune(

  476.                 pars.GlobalPar.PDetuneType,

  477.                 0,

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

  479.         }

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

  481.             NoteVoicePar[nvoice].FMDetune = getdetune(

  482.                 pars.VoicePar[nvoice].PFMDetuneType,

  483.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  485.         else

  486.             NoteVoicePar[nvoice].FMDetune = getdetune(

  487.                 pars.GlobalPar.PDetuneType,

  488.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  490. 

  491. 

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

  493.         int vc = nvoice;

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

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

  496.         if(!pars.GlobalPar.Hrandgrouping)

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

  498. 

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

  500.                                          getvoicebasefreq(nvoice),

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

  502. 

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

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

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

  506.                                           + i] =

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

  508. 

  509. 

  510.         NoteVoicePar[nvoice].FilterCenterPitch =

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

  512.         NoteVoicePar[nvoice].filterbypass =

  513.             pars.VoicePar[nvoice].Pfilterbypass;

  514. 

  515. 

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

  517. 

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

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

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

  521.                                - 1.0f);

  522. 

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

  524.             case PHASE_MOD:

  525.                 fmvoldamp =

  526.                     powf(440.0f / getvoicebasefreq(

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

  528.                          / 64.0f);

  529.                 NoteVoicePar[nvoice].FMVolume =

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

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

  532.                 break;

  533.             case FREQ_MOD:

  534.                 NoteVoicePar[nvoice].FMVolume =

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

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

  537.                 break;

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

  539.             //	          break;

  540.             default:

  541.                 if(fmvoldamp > 1.0f)

  542.                     fmvoldamp = 1.0f;

  543.                 NoteVoicePar[nvoice].FMVolume =

  544.                     pars.VoicePar[nvoice].PFMVolume

  545.                     / 127.0f * fmvoldamp;

  546.         }

  547. 

  548.         //Voice's modulator velocity sensing

  549.         NoteVoicePar[nvoice].FMVolume *=

  550.             VelF(velocity,

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

  552. 

  553.         NoteVoicePar[nvoice].DelayTicks =

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

  555.                         * logf(50.0f))

  556.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  557.     }

  558. 

  559.     ///    initparameters();

  560. 

  561.     ///////////////

  562.     // Altered content of initparameters():

  563. 

  564.     int tmp[NUM_VOICES];

  565. 

  566.     NoteGlobalPar.Volume = 4.0f

  567.                            * powf(0.1f, 3.0f

  568.                                   * (1.0f - pars.GlobalPar.PVolume

  569.                                      / 96.0f))                                      //-60 dB .. 0 dB

  570.                            * VelF(

  571.         velocity,

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

  573. 

  574.     globalnewamplitude = NoteGlobalPar.Volume

  575.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  577. 

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

  579.     NoteGlobalPar.FilterFreqTracking =

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

  581. 

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

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

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

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

  586. 

  587.     // Voice Parameter init

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

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

  590.             continue;

  591. 

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

  593.         /* Voice Amplitude Parameters Init */

  594.         NoteVoicePar[nvoice].Volume =

  595.             powf(0.1f, 3.0f

  596.                  * (1.0f - pars.VoicePar[nvoice].PVolume / 127.0f))             // -60 dB .. 0 dB

  597.             * VelF(velocity,

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

  599. 

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

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

  602. 

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

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

  605.         else

  606.             NoteVoicePar[nvoice].Panning =

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

  608. 

  609.         newamplitude[nvoice] = 1.0f;

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

  611.            && NoteVoicePar[nvoice].AmpEnvelope)

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

  613. 

  614. 

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

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

  617. 

  618. 

  619. 

  620.         NoteVoicePar[nvoice].FilterFreqTracking =

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

  622. 

  623.         /* Voice Modulation Parameters Init */

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

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

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

  627. 

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

  629. 

  630.             int vc = nvoice;

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

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

  633. 

  634.             if(!pars.GlobalPar.Hrandgrouping)

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

  636. 

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

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

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

  640.         }

  641. 

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

  643.                                  * ctl.fmamp.relamp;

  644. 

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

  646.            && NoteVoicePar[nvoice].FMAmpEnvelope)

  647.             FMnewamplitude[nvoice] *=

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

  649.     }

  650. 

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

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

  653.             tmp[i] = 0;

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

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

  656.                 tmp[i] = 1;

  657.     }

  658. }

  659. 

  660. 

  661. /*

  662.  * Kill a voice of ADnote

  663.  */

  664. void ADnote::KillVoice(int nvoice)

  665. {

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

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

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

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

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

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

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

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

  674. 

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

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

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

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

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

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

  681. 

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

  683. }

  684. 

  685. /*

  686.  * Kill the note

  687.  */

  688. void ADnote::KillNote()

  689. {

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

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

  692.             KillVoice(nvoice);

  693. 

  694.         if(NoteVoicePar[nvoice].VoiceOut)

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

  696.     }

  697. 

  698.     NoteGlobalPar.kill(memory);

  699. 

  700.     NoteEnabled = OFF;

  701. }

  702. 

  703. ADnote::~ADnote()

  704. {

  705.     if(NoteEnabled == ON)

  706.         KillNote();

  707.     memory.devalloc(tmpwavel);

  708.     memory.devalloc(tmpwaver);

  709.     memory.devalloc(bypassl);

  710.     memory.devalloc(bypassr);

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

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

  713.     memory.devalloc(tmpwave_unison);

  714. }

  715. 

  716. 

  717. /*

  718.  * Init the parameters

  719.  */

  720. void ADnote::initparameters()

  721. {

  722.     int tmp[NUM_VOICES];

  723.     //ADnoteParameters &pars = *partparams;

  724. 

  725.     // Global Parameters

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

  727.                                  time,

  728.                                  memory, basefreq, velocity,

  729.                                  stereo);

  730. 

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

  732.     globalnewamplitude = NoteGlobalPar.Volume

  733.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  735. 

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

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

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

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

  740. 

  741.     // Voice Parameter init

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

  743.         Voice &vce = NoteVoicePar[nvoice];

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

  745. 

  746.         if(vce.Enabled == 0)

  747.             continue;

  748. 

  749.         vce.noisetype = param.Type;

  750.         /* Voice Amplitude Parameters Init */

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

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

  753. 

  754.         if(param.PVolumeminus)

  755.             vce.Volume = -vce.Volume;

  756. 

  757.         if(param.PPanning == 0)

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

  759.         else

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

  761. 

  762.         newamplitude[nvoice] = 1.0f;

  763.         if(param.PAmpEnvelopeEnabled) {

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

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

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

  767.         }

  768. 

  769.         if(param.PAmpLfoEnabled) {

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

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

  772.         }

  773. 

  774.         /* Voice Frequency Parameters Init */

  775.         if(param.PFreqEnvelopeEnabled)

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

  777. 

  778.         if(param.PFreqLfoEnabled)

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

  780. 

  781.         /* Voice Filter Parameters Init */

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

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

  784.                     synth.samplerate, synth.buffersize);

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

  786.                     synth.samplerate, synth.buffersize);

  787.         }

  788. 

  789.         if(param.PFilterEnvelopeEnabled)

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

  791. 

  792.         if(param.PFilterLfoEnabled)

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

  794. 

  795.         vce.FilterFreqTracking =

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

  797. 

  798.         /* Voice Modulation Parameters Init */

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

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

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

  802. 

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

  804. 

  805.             int vc = nvoice;

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

  807.                 vc = param.PextFMoscil;

  808. 

  809.             float tmp = 1.0f;

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

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

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

  813.                 tmp = getFMvoicebasefreq(nvoice);

  814. 

  815.             if(!pars.GlobalPar.Hrandgrouping)

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

  817. 

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

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

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

  821.                                              vce.FMSmp, tmp))

  822.                                         % synth.oscilsize;

  823. 

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

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

  826.             int oscposhiFM_add =

  827.                 (int)((param.PFMoscilphase

  828.                        - 64.0f) / 128.0f * synth.oscilsize

  829.                       + synth.oscilsize * 4);

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

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

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

  833.             }

  834.         }

  835. 

  836.         if(param.PFMFreqEnvelopeEnabled)

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

  838. 

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

  840. 

  841.         if(param.PFMAmpEnvelopeEnabled ) {

  842.             vce.FMAmpEnvelope =

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

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

  845.         }

  846.     }

  847. 

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

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

  850.             tmp[i] = 0;

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

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

  853.                 NoteVoicePar[nvoice].VoiceOut =

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

  855.                 tmp[i] = 1;

  856.             }

  857. 

  858.         if(NoteVoicePar[nvoice].VoiceOut)

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

  860.     }

  861. }

  862. 

  863. 

  864. /*

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

  866.  * This must be called before setfreq* functions

  867.  */

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

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

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

  871.         return;

  872.     }

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

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

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

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

  877.         pos += step;

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

  879.             pos  = -1.0f;

  880.             step = -step;

  881.         }

  882.         if(pos >= 1.0f) {

  883.             pos  = 1.0f;

  884.             step = -step;

  885.         }

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

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

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

  889.                                          - 1.0f) + vibratto_val

  890.                                         * unison_vibratto[nvoice].amplitude)

  891.                                      * relbw;

  892. 

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

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

  895.     }

  896. }

  897. 

  898. 

  899. /*

  900.  * Computes the frequency of an oscillator

  901.  */

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

  903. {

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

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

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

  907.         if(speed > synth.oscilsize_f)

  908.             speed = synth.oscilsize_f;

  909. 

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

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

  912.     }

  913. }

  914. 

  915. /*

  916.  * Computes the frequency of an modullator oscillator

  917.  */

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

  919. {

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

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

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

  923.         if(speed > synth.samplerate_f)

  924.             speed = synth.samplerate_f;

  925. 

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

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

  928.     }

  929. }

  930. 

  931. /*

  932.  * Get Voice base frequency

  933.  */

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

  935. {

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

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

  938.                    * ctl.bandwidth.relbw * bandwidthDetuneMultiplier

  939.                    + NoteGlobalPar.Detune / 100.0f;

  940. 

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

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

  943.     else { //the fixed freq is enabled

  944.         float fixedfreq   = 440.0f;

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

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

  947.             float tmp =

  948.                 (midinote

  949.                  - 69.0f) / 12.0f

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

  951.             if(fixedfreqET <= 64)

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

  953.             else

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

  955.         }

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

  957.     }

  958. }

  959. 

  960. /*

  961.  * Get Voice's Modullator base frequency

  962.  */

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

  964. {

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

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

  967. }

  968. 

  969. /*

  970.  * Computes all the parameters for each tick

  971.  */

  972. void ADnote::computecurrentparameters()

  973. {

  974.     int   nvoice;

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

  976.           FMrelativepitch, globalpitch, globalfilterpitch;

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

  978.                            + NoteGlobalPar.FreqLfo->lfoout()

  979.                            * ctl.modwheel.relmod);

  980.     globaloldamplitude = globalnewamplitude;

  981.     globalnewamplitude = NoteGlobalPar.Volume

  982.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  984. 

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

  986.                         + NoteGlobalPar.FilterLfo->lfoout()

  987.                         + NoteGlobalPar.FilterCenterPitch;

  988. 

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

  990.                           + NoteGlobalPar.FilterFreqTracking;

  991. 

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

  993. 

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

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

  996.     if(stereo != 0)

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

  998. 

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

  1000.     float portamentofreqrap = 1.0f;

  1001.     if(portamento != 0) { //this voice use portamento

  1002.         portamentofreqrap = ctl.portamento.freqrap;

  1003.         if(ctl.portamento.used == 0) //the portamento has finished

  1004.             portamento = 0;  //this note is no longer "portamented"

  1005.     }

  1006. 

  1007.     //compute parameters for all voices

  1008.     for(nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1009.         if(NoteVoicePar[nvoice].Enabled != ON)

  1010.             continue;

  1011.         NoteVoicePar[nvoice].DelayTicks -= 1;

  1012.         if(NoteVoicePar[nvoice].DelayTicks > 0)

  1013.             continue;

  1014. 

  1015.         compute_unison_freq_rap(nvoice);

  1016. 

  1017.         /*******************/

  1018.         /* Voice Amplitude */

  1019.         /*******************/

  1020.         oldamplitude[nvoice] = newamplitude[nvoice];

  1021.         newamplitude[nvoice] = 1.0f;

  1022. 

  1023.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1024.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpEnvelope->envout_dB();

  1025. 

  1026.         if(NoteVoicePar[nvoice].AmpLfo)

  1027.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpLfo->amplfoout();

  1028. 

  1029.         /****************/

  1030.         /* Voice Filter */

  1031.         /****************/

  1032.         if(NoteVoicePar[nvoice].VoiceFilterL) {

  1033.             filterpitch = NoteVoicePar[nvoice].FilterCenterPitch;

  1034. 

  1035.             if(NoteVoicePar[nvoice].FilterEnvelope)

  1036.                 filterpitch += NoteVoicePar[nvoice].FilterEnvelope->envout();

  1037. 

  1038.             if(NoteVoicePar[nvoice].FilterLfo)

  1039.                 filterpitch += NoteVoicePar[nvoice].FilterLfo->lfoout();

  1040. 

  1041.             filterfreq = filterpitch + NoteVoicePar[nvoice].FilterFreqTracking;

  1042.             filterfreq = Filter::getrealfreq(filterfreq);

  1043. 

  1044.             NoteVoicePar[nvoice].VoiceFilterL->setfreq(filterfreq);

  1045.             if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1046.                 NoteVoicePar[nvoice].VoiceFilterR->setfreq(filterfreq);

  1047.         }

  1048. 

  1049.         if(NoteVoicePar[nvoice].noisetype == 0) { //compute only if the voice isn't noise

  1050.             /*******************/

  1051.             /* Voice Frequency */

  1052.             /*******************/

  1053.             voicepitch = 0.0f;

  1054.             if(NoteVoicePar[nvoice].FreqLfo)

  1055.                 voicepitch += NoteVoicePar[nvoice].FreqLfo->lfoout() / 100.0f

  1056.                               * ctl.bandwidth.relbw;

  1057. 

  1058.             if(NoteVoicePar[nvoice].FreqEnvelope)

  1059.                 voicepitch += NoteVoicePar[nvoice].FreqEnvelope->envout()

  1060.                               / 100.0f;

  1061.             voicefreq = getvoicebasefreq(nvoice)

  1062.                         * powf(2, (voicepitch + globalpitch) / 12.0f);                //Hz frequency

  1063.             voicefreq *= ctl.pitchwheel.relfreq; //change the frequency by the controller

  1064.             setfreq(nvoice, voicefreq * portamentofreqrap);

  1065. 

  1066.             /***************/

  1067.             /*  Modulator */

  1068.             /***************/

  1069.             if(NoteVoicePar[nvoice].FMEnabled != NONE) {

  1070.                 FMrelativepitch = NoteVoicePar[nvoice].FMDetune / 100.0f;

  1071.                 if(NoteVoicePar[nvoice].FMFreqEnvelope)

  1072.                     FMrelativepitch +=

  1073.                         NoteVoicePar[nvoice].FMFreqEnvelope->envout() / 100;

  1074.                 FMfreq =

  1075.                     powf(2.0f, FMrelativepitch

  1076.                          / 12.0f) * voicefreq * portamentofreqrap;

  1077.                 setfreqFM(nvoice, FMfreq);

  1078. 

  1079.                 FMoldamplitude[nvoice] = FMnewamplitude[nvoice];

  1080.                 FMnewamplitude[nvoice] = NoteVoicePar[nvoice].FMVolume

  1081.                                          * ctl.fmamp.relamp;

  1082.                 if(NoteVoicePar[nvoice].FMAmpEnvelope)

  1083.                     FMnewamplitude[nvoice] *=

  1084.                         NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();

  1085.             }

  1086.         }

  1087.     }

  1088. }

  1089. 

  1090. 

  1091. /*

  1092.  * Fadein in a way that removes clicks but keep sound "punchy"

  1093.  */

  1094. inline void ADnote::fadein(float *smps) const

  1095. {

  1096.     int zerocrossings = 0;

  1097.     for(int i = 1; i < synth.buffersize; ++i)

  1098.         if((smps[i - 1] < 0.0f) && (smps[i] > 0.0f))

  1099.             zerocrossings++;  //this is only the possitive crossings

  1100. 

  1101.     float tmp = (synth.buffersize_f - 1.0f) / (zerocrossings + 1) / 3.0f;

  1102.     if(tmp < 8.0f)

  1103.         tmp = 8.0f;

  1104. 

  1105.     int n;

  1106.     F2I(tmp, n); //how many samples is the fade-in

  1107.     if(n > synth.buffersize)

  1108.         n = synth.buffersize;

  1109.     for(int i = 0; i < n; ++i) { //fade-in

  1110.         float tmp = 0.5f - cosf((float)i / (float) n * PI) * 0.5f;

  1111.         smps[i] *= tmp;

  1112.     }

  1113. }

  1114. 

  1115. /*

  1116.  * Computes the Oscillator (Without Modulation) - LinearInterpolation

  1117.  */

  1118. 

  1119. /* As the code here is a bit odd due to optimization, here is what happens

  1120.  * First the current possition and frequency are retrieved from the running

  1121.  * state. These are broken up into high and low portions to indicate how many

  1122.  * samples are skipped in one step and how many fractional samples are skipped.

  1123.  * Outside of this method the fractional samples are just handled with floating

  1124.  * point code, but that's a bit slower than it needs to be. In this code the low

  1125.  * portions are known to exist between 0.0 and 1.0 and it is known that they are

  1126.  * stored in single precision floating point IEEE numbers. This implies that

  1127.  * a maximum of 24 bits are significant. The below code does your standard

  1128.  * linear interpolation that you'll see throughout this codebase, but by

  1129.  * sticking to integers for tracking the overflow of the low portion, around 15%

  1130.  * of the execution time was shaved off in the ADnote test.

  1131.  */

  1132. inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)

  1133. {

  1134.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1135.         int    poshi  = oscposhi[nvoice][k];

  1136.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1137.         int    freqhi = oscfreqhi[nvoice][k];

  1138.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1139.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1140.         float *tw     = tmpwave_unison[k];

  1141.         assert(oscfreqlo[nvoice][k] < 1.0f);

  1142.         for(int i = 0; i < synth.buffersize; ++i) {

  1143.             tw[i]  = (smps[poshi] * ((1<<24) - poslo) + smps[poshi + 1] * poslo)/(1.0f*(1<<24));

  1144.             poslo += freqlo;

  1145.             poshi += freqhi + (poslo>>24);

  1146.             poslo &= 0xffffff;

  1147.             poshi &= synth.oscilsize - 1;

  1148.         }

  1149.         oscposhi[nvoice][k] = poshi;

  1150.         oscposlo[nvoice][k] = poslo/(1.0f*(1<<24));

  1151.     }

  1152. }

  1153. 

  1154. 

  1155. 

  1156. /*

  1157.  * Computes the Oscillator (Without Modulation) - CubicInterpolation

  1158.  *

  1159.  The differences from the Linear are to little to deserve to be used. This is because I am using a large synth.oscilsize (>512)

  1160. inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){

  1161.     int i,poshi;

  1162.     float poslo;

  1163. 

  1164.     poshi=oscposhi[nvoice];

  1165.     poslo=oscposlo[nvoice];

  1166.     float *smps=NoteVoicePar[nvoice].OscilSmp;

  1167.     float xm1,x0,x1,x2,a,b,c;

  1168.     for (i=0;i<synth.buffersize;i++){

  1169.     xm1=smps[poshi];

  1170.     x0=smps[poshi+1];

  1171.     x1=smps[poshi+2];

  1172.     x2=smps[poshi+3];

  1173.     a=(3.0f * (x0-x1) - xm1 + x2) / 2.0f;

  1174.     b = 2.0f*x1 + xm1 - (5.0f*x0 + x2) / 2.0f;

  1175.     c = (x1 - xm1) / 2.0f;

  1176.     tmpwave[i]=(((a * poslo) + b) * poslo + c) * poslo + x0;

  1177.     printf("a\n");

  1178.     //tmpwave[i]=smps[poshi]*(1.0f-poslo)+smps[poshi+1]*poslo;

  1179.     poslo+=oscfreqlo[nvoice];

  1180.     if (poslo>=1.0f) {

  1181.             poslo-=1.0f;

  1182.         poshi++;

  1183.     };

  1184.         poshi+=oscfreqhi[nvoice];

  1185.         poshi&=synth.oscilsize-1;

  1186.     };

  1187.     oscposhi[nvoice]=poshi;

  1188.     oscposlo[nvoice]=poslo;

  1189. };

  1190. */

  1191. /*

  1192.  * Computes the Oscillator (Morphing)

  1193.  */

  1194. inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)

  1195. {

  1196.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1197.     if(FMnewamplitude[nvoice] > 1.0f)

  1198.         FMnewamplitude[nvoice] = 1.0f;

  1199.     if(FMoldamplitude[nvoice] > 1.0f)

  1200.         FMoldamplitude[nvoice] = 1.0f;

  1201. 

  1202.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1203.         //if I use VoiceOut[] as modullator

  1204.         int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1205.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1206.             float *tw = tmpwave_unison[k];

  1207.             for(int i = 0; i < synth.buffersize; ++i) {

  1208.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1209.                                             FMnewamplitude[nvoice],

  1210.                                             i,

  1211.                                             synth.buffersize);

  1212.                 tw[i] = tw[i]

  1213.                     * (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1214.             }

  1215.         }

  1216.     }

  1217.     else

  1218.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1219.             int    poshiFM  = oscposhiFM[nvoice][k];

  1220.             float  posloFM  = oscposloFM[nvoice][k];

  1221.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1222.             float  freqloFM = oscfreqloFM[nvoice][k];

  1223.             float *tw = tmpwave_unison[k];

  1224. 

  1225.             for(int i = 0; i < synth.buffersize; ++i) {

  1226.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1227.                                             FMnewamplitude[nvoice],

  1228.                                             i,

  1229.                                             synth.buffersize);

  1230.                 tw[i] = tw[i] * (1.0f - amp) + amp

  1231.                         * (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1 - posloFM)

  1232.                            + NoteVoicePar[nvoice].FMSmp[poshiFM + 1] * posloFM);

  1233.                 posloFM += freqloFM;

  1234.                 if(posloFM >= 1.0f) {

  1235.                     posloFM -= 1.0f;

  1236.                     poshiFM++;

  1237.                 }

  1238.                 poshiFM += freqhiFM;

  1239.                 poshiFM &= synth.oscilsize - 1;

  1240.             }

  1241.             oscposhiFM[nvoice][k] = poshiFM;

  1242.             oscposloFM[nvoice][k] = posloFM;

  1243.         }

  1244. }

  1245. 

  1246. /*

  1247.  * Computes the Oscillator (Ring Modulation)

  1248.  */

  1249. inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)

  1250. {

  1251.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1252.     if(FMnewamplitude[nvoice] > 1.0f)

  1253.         FMnewamplitude[nvoice] = 1.0f;

  1254.     if(FMoldamplitude[nvoice] > 1.0f)

  1255.         FMoldamplitude[nvoice] = 1.0f;

  1256.     if(NoteVoicePar[nvoice].FMVoice >= 0)

  1257.         // if I use VoiceOut[] as modullator

  1258.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1259.             float *tw = tmpwave_unison[k];

  1260.             for(int i = 0; i < synth.buffersize; ++i) {

  1261.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1262.                                             FMnewamplitude[nvoice],

  1263.                                             i,

  1264.                                             synth.buffersize);

  1265.                 int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1266.                 tw[i] *= (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1267.             }

  1268.         }

  1269.     else

  1270.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1271.             int    poshiFM  = oscposhiFM[nvoice][k];

  1272.             float  posloFM  = oscposloFM[nvoice][k];

  1273.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1274.             float  freqloFM = oscfreqloFM[nvoice][k];

  1275.             float *tw = tmpwave_unison[k];

  1276. 

  1277.             for(int i = 0; i < synth.buffersize; ++i) {

  1278.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1279.                                             FMnewamplitude[nvoice],

  1280.                                             i,

  1281.                                             synth.buffersize);

  1282.                 tw[i] *= (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1.0f - posloFM)

  1283.                           + NoteVoicePar[nvoice].FMSmp[poshiFM

  1284.                                                        + 1] * posloFM) * amp

  1285.                          + (1.0f - amp);

  1286.                 posloFM += freqloFM;

  1287.                 if(posloFM >= 1.0f) {

  1288.                     posloFM -= 1.0f;

  1289.                     poshiFM++;

  1290.                 }

  1291.                 poshiFM += freqhiFM;

  1292.                 poshiFM &= synth.oscilsize - 1;

  1293.             }

  1294.             oscposhiFM[nvoice][k] = poshiFM;

  1295.             oscposloFM[nvoice][k] = posloFM;

  1296.         }

  1297. }

  1298. 

  1299. /*

  1300.  * Computes the Oscillator (Phase Modulation or Frequency Modulation)

  1301.  */

  1302. inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,

  1303.                                                               int FMmode)

  1304. {

  1305.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1306.         //if I use VoiceOut[] as modulator

  1307.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1308.             float *tw = tmpwave_unison[k];

  1309.             const float *smps = NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut;

  1310.             memcpy(tw, smps, synth.bufferbytes);

  1311.         }

  1312.     } else {

  1313.         //Compute the modulator and store it in tmpwave_unison[][]

  1314.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1315.             int    poshiFM  = oscposhiFM[nvoice][k];

  1316.             int    posloFM  = oscposloFM[nvoice][k]  * (1<<24);

  1317.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1318.             int    freqloFM = oscfreqloFM[nvoice][k] * (1<<24);

  1319.             float *tw = tmpwave_unison[k];

  1320.             const float *smps = NoteVoicePar[nvoice].FMSmp;

  1321. 

  1322.             for(int i = 0; i < synth.buffersize; ++i) {

  1323.                 tw[i] = (smps[poshiFM] * ((1<<24) - posloFM)

  1324.                      + smps[poshiFM + 1] * posloFM) / (1.0f*(1<<24));

  1325.                 posloFM += freqloFM;

  1326.                 if(posloFM >= (1<<24)) {

  1327.                     posloFM &= 0xffffff;//fmod(posloFM, 1.0f);

  1328.                     poshiFM++;

  1329.                 }

  1330.                 poshiFM += freqhiFM;

  1331.                 poshiFM &= synth.oscilsize - 1;

  1332.             }

  1333.             oscposhiFM[nvoice][k] = poshiFM;

  1334.             oscposloFM[nvoice][k] = posloFM/((1<<24)*1.0f);

  1335.         }

  1336.     }

  1337.     // Amplitude interpolation

  1338.     if(ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],

  1339.                                  FMnewamplitude[nvoice])) {

  1340.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1341.             float *tw = tmpwave_unison[k];

  1342.             for(int i = 0; i < synth.buffersize; ++i)

  1343.                 tw[i] *= INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1344.                                                FMnewamplitude[nvoice],

  1345.                                                i,

  1346.                                                synth.buffersize);

  1347.         }

  1348.     } else {

  1349.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1350.             float *tw = tmpwave_unison[k];

  1351.             for(int i = 0; i < synth.buffersize; ++i)

  1352.                 tw[i] *= FMnewamplitude[nvoice];

  1353.         }

  1354.     }

  1355. 

  1356. 

  1357.     //normalize: makes all sample-rates, oscil_sizes to produce same sound

  1358.     if(FMmode != 0) { //Frequency modulation

  1359.         const float normalize = synth.oscilsize_f / 262144.0f * 44100.0f

  1360.                           / synth.samplerate_f;

  1361.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1362.             float *tw    = tmpwave_unison[k];

  1363.             float  fmold = FMoldsmp[nvoice][k];

  1364.             for(int i = 0; i < synth.buffersize; ++i) {

  1365.                 fmold = fmod(fmold + tw[i] * normalize, synth.oscilsize);

  1366.                 tw[i] = fmold;

  1367.             }

  1368.             FMoldsmp[nvoice][k] = fmold;

  1369.         }

  1370.     }

  1371.     else {  //Phase modulation

  1372.         const float normalize = synth.oscilsize_f / 262144.0f;

  1373.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1374.             float *tw = tmpwave_unison[k];

  1375.             for(int i = 0; i < synth.buffersize; ++i)

  1376.                 tw[i] *= normalize;

  1377.         }

  1378.     }

  1379. 

  1380.     //do the modulation

  1381.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1382.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1383.         float *tw     = tmpwave_unison[k];

  1384.         int    poshi  = oscposhi[nvoice][k];

  1385.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1386.         int    freqhi = oscfreqhi[nvoice][k];

  1387.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1388. 

  1389.         for(int i = 0; i < synth.buffersize; ++i) {

  1390.             int FMmodfreqhi = 0;

  1391.             F2I(tw[i], FMmodfreqhi);

  1392.             float FMmodfreqlo = tw[i]-FMmodfreqhi;//fmod(tw[i] /*+ 0.0000000001f*/, 1.0f);

  1393.             if(FMmodfreqhi < 0)

  1394.                 FMmodfreqlo++;

  1395. 

  1396.             //carrier

  1397.             int carposhi = poshi + FMmodfreqhi;

  1398.             int carposlo = poslo + FMmodfreqlo;

  1399. 

  1400.             if(carposlo >= (1<<24)) {

  1401.                 carposhi++;

  1402.                 carposlo &= 0xffffff;//fmod(carposlo, 1.0f);

  1403.             }

  1404.             carposhi &= (synth.oscilsize - 1);

  1405. 

  1406.             tw[i] = (smps[carposhi] * ((1<<24) - carposlo)

  1407.                     + smps[carposhi + 1] * carposlo)/(1.0f*(1<<24));

  1408. 

  1409.             poslo += freqlo;

  1410.             if(poslo >= (1<<24)) {

  1411.                 poslo &= 0xffffff;//fmod(poslo, 1.0f);

  1412.                 poshi++;

  1413.             }

  1414. 

  1415.             poshi += freqhi;

  1416.             poshi &= synth.oscilsize - 1;

  1417.         }

  1418.         oscposhi[nvoice][k] = poshi;

  1419.         oscposlo[nvoice][k] = (poslo)/((1<<24)*1.0f);

  1420.     }

  1421. }

  1422. 

  1423. 

  1424. /*Calculeaza Oscilatorul cu PITCH MODULATION*/

  1425. inline void ADnote::ComputeVoiceOscillatorPitchModulation(int /*nvoice*/)

  1426. {

  1427. //TODO

  1428. }

  1429. 

  1430. /*

  1431.  * Computes the Noise

  1432.  */

  1433. inline void ADnote::ComputeVoiceWhiteNoise(int nvoice)

  1434. {

  1435.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1436.         float *tw = tmpwave_unison[k];

  1437.         for(int i = 0; i < synth.buffersize; ++i)

  1438.             tw[i] = RND * 2.0f - 1.0f;

  1439.     }

  1440. }

  1441. 

  1442. inline void ADnote::ComputeVoicePinkNoise(int nvoice)

  1443. {

  1444.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1445.         float *tw = tmpwave_unison[k];

  1446.         float *f = &pinking[nvoice][k > 0 ? 7 : 0];

  1447.         for(int i = 0; i < synth.buffersize; ++i) {

  1448. 	    float white = (RND-0.5)/4.0;

  1449. 	    f[0] = 0.99886*f[0]+white*0.0555179;

  1450. 	    f[1] = 0.99332*f[1]+white*0.0750759;

  1451. 	    f[2] = 0.96900*f[2]+white*0.1538520;

  1452. 	    f[3] = 0.86650*f[3]+white*0.3104856;

  1453. 	    f[4] = 0.55000*f[4]+white*0.5329522;

  1454. 	    f[5] = -0.7616*f[5]-white*0.0168980;

  1455. 	    tw[i] = f[0]+f[1]+f[2]+f[3]+f[4]+f[5]+f[6]+white*0.5362;

  1456. 	    f[6] = white*0.115926;

  1457.         }

  1458.     }

  1459. }

  1460. 

  1461. 

  1462. 

  1463. /*

  1464.  * Compute the ADnote samples

  1465.  * Returns 0 if the note is finished

  1466.  */

  1467. int ADnote::noteout(float *outl, float *outr)

  1468. {

  1469.     memcpy(outl, synth.denormalkillbuf, synth.bufferbytes);

  1470.     memcpy(outr, synth.denormalkillbuf, synth.bufferbytes);

  1471. 

  1472.     if(NoteEnabled == OFF)

  1473.         return 0;

  1474. 

  1475.     memset(bypassl, 0, synth.bufferbytes);

  1476.     memset(bypassr, 0, synth.bufferbytes);

  1477.     computecurrentparameters();

  1478. 

  1479.     for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1480.         if((NoteVoicePar[nvoice].Enabled != ON)

  1481.            || (NoteVoicePar[nvoice].DelayTicks > 0))

  1482.             continue;

  1483.         switch (NoteVoicePar[nvoice].noisetype) {

  1484.             case 0: //voice mode=sound

  1485.                 switch(NoteVoicePar[nvoice].FMEnabled) {

  1486.                     case MORPH:

  1487.                         ComputeVoiceOscillatorMorph(nvoice);

  1488.                         break;

  1489.                     case RING_MOD:

  1490.                         ComputeVoiceOscillatorRingModulation(nvoice);

  1491.                         break;

  1492.                     case PHASE_MOD:

  1493.                         ComputeVoiceOscillatorFrequencyModulation(nvoice, 0);

  1494.                         break;

  1495.                     case FREQ_MOD:

  1496.                         ComputeVoiceOscillatorFrequencyModulation(nvoice, 1);

  1497.                         break;

  1498.                     //case PITCH_MOD:ComputeVoiceOscillatorPitchModulation(nvoice);break;

  1499.                     default:

  1500.                         ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1501.                         //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice);

  1502.                 }

  1503.                 break;

  1504.             case 1:

  1505.                 ComputeVoiceWhiteNoise(nvoice);

  1506.                 break;

  1507.             default:

  1508.                 ComputeVoicePinkNoise(nvoice);

  1509.                 break;

  1510.         }

  1511.         // Voice Processing

  1512. 

  1513. 

  1514.         //mix subvoices into voice

  1515.         memset(tmpwavel, 0, synth.bufferbytes);

  1516.         if(stereo)

  1517.             memset(tmpwaver, 0, synth.bufferbytes);

  1518.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1519.             float *tw = tmpwave_unison[k];

  1520.             if(stereo) {

  1521.                 float stereo_pos = 0;

  1522.                 if(unison_size[nvoice] > 1)

  1523.                     stereo_pos = k

  1524.                                  / (float)(unison_size[nvoice]

  1525.                                            - 1) * 2.0f - 1.0f;

  1526.                 float stereo_spread = unison_stereo_spread[nvoice] * 2.0f; //between 0 and 2.0f

  1527.                 if(stereo_spread > 1.0f) {

  1528.                     float stereo_pos_1 = (stereo_pos >= 0.0f) ? 1.0f : -1.0f;

  1529.                     stereo_pos =

  1530.                         (2.0f

  1531.                          - stereo_spread) * stereo_pos

  1532.                         + (stereo_spread - 1.0f) * stereo_pos_1;

  1533.                 }

  1534.                 else

  1535.                     stereo_pos *= stereo_spread;

  1536. 

  1537.                 if(unison_size[nvoice] == 1)

  1538.                     stereo_pos = 0.0f;

  1539.                 float panning = (stereo_pos + 1.0f) * 0.5f;

  1540. 

  1541. 

  1542.                 float lvol = (1.0f - panning) * 2.0f;

  1543.                 if(lvol > 1.0f)

  1544.                     lvol = 1.0f;

  1545. 

  1546.                 float rvol = panning * 2.0f;

  1547.                 if(rvol > 1.0f)

  1548.                     rvol = 1.0f;

  1549. 

  1550.                 if(unison_invert_phase[nvoice][k]) {

  1551.                     lvol = -lvol;

  1552.                     rvol = -rvol;

  1553.                 }

  1554. 

  1555.                 for(int i = 0; i < synth.buffersize; ++i)

  1556.                     tmpwavel[i] += tw[i] * lvol;

  1557.                 for(int i = 0; i < synth.buffersize; ++i)

  1558.                     tmpwaver[i] += tw[i] * rvol;

  1559.             }

  1560.             else

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

  1562.                     tmpwavel[i] += tw[i];

  1563.         }

  1564. 

  1565. 

  1566.         float unison_amplitude = 1.0f / sqrt(unison_size[nvoice]); //reduce the amplitude for large unison sizes

  1567.         // Amplitude

  1568.         float oldam = oldamplitude[nvoice] * unison_amplitude;

  1569.         float newam = newamplitude[nvoice] * unison_amplitude;

  1570. 

  1571.         if(ABOVE_AMPLITUDE_THRESHOLD(oldam, newam)) {

  1572.             int rest = synth.buffersize;

  1573.             //test if the amplitude if raising and the difference is high

  1574.             if((newam > oldam) && ((newam - oldam) > 0.25f)) {

  1575.                 rest = 10;

  1576.                 if(rest > synth.buffersize)

  1577.                     rest = synth.buffersize;

  1578.                 for(int i = 0; i < synth.buffersize - rest; ++i)

  1579.                     tmpwavel[i] *= oldam;

  1580.                 if(stereo)

  1581.                     for(int i = 0; i < synth.buffersize - rest; ++i)

  1582.                         tmpwaver[i] *= oldam;

  1583.             }

  1584.             // Amplitude interpolation

  1585.             for(int i = 0; i < rest; ++i) {

  1586.                 float amp = INTERPOLATE_AMPLITUDE(oldam, newam, i, rest);

  1587.                 tmpwavel[i + (synth.buffersize - rest)] *= amp;

  1588.                 if(stereo)

  1589.                     tmpwaver[i + (synth.buffersize - rest)] *= amp;

  1590.             }

  1591.         }

  1592.         else {

  1593.             for(int i = 0; i < synth.buffersize; ++i)

  1594.                 tmpwavel[i] *= newam;

  1595.             if(stereo)

  1596.                 for(int i = 0; i < synth.buffersize; ++i)

  1597.                     tmpwaver[i] *= newam;

  1598.         }

  1599. 

  1600.         // Fade in

  1601.         if(firsttick[nvoice] != 0) {

  1602.             fadein(&tmpwavel[0]);

  1603.             if(stereo)

  1604.                 fadein(&tmpwaver[0]);

  1605.             firsttick[nvoice] = 0;

  1606.         }

  1607. 

  1608. 

  1609.         // Filter

  1610.         if(NoteVoicePar[nvoice].VoiceFilterL)

  1611.             NoteVoicePar[nvoice].VoiceFilterL->filterout(&tmpwavel[0]);

  1612.         if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1613.             NoteVoicePar[nvoice].VoiceFilterR->filterout(&tmpwaver[0]);

  1614. 

  1615.         //check if the amplitude envelope is finished, if yes, the voice will be fadeout

  1616.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1617.             if(NoteVoicePar[nvoice].AmpEnvelope->finished()) {

  1618.                 for(int i = 0; i < synth.buffersize; ++i)

  1619.                     tmpwavel[i] *= 1.0f - (float)i / synth.buffersize_f;

  1620.                 if(stereo)

  1621.                     for(int i = 0; i < synth.buffersize; ++i)

  1622.                         tmpwaver[i] *= 1.0f - (float)i / synth.buffersize_f;

  1623.             }

  1624.         //the voice is killed later

  1625. 

  1626. 

  1627.         // Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator)

  1628.         if(NoteVoicePar[nvoice].VoiceOut) {

  1629.             if(stereo)

  1630.                 for(int i = 0; i < synth.buffersize; ++i)

  1631.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i]

  1632.                                                        + tmpwaver[i];

  1633.             else   //mono

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

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

  1636.         }

  1637. 

  1638. 

  1639.         // Add the voice that do not bypass the filter to out

  1640.         if(NoteVoicePar[nvoice].filterbypass == 0) { //no bypass

  1641.             if(stereo)

  1642.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1643.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1644.                                * NoteVoicePar[nvoice].Panning * 2.0f;

  1645.                     outr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1646.                                * (1.0f - NoteVoicePar[nvoice].Panning) * 2.0f;

  1647.                 }

  1648.             else

  1649.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1650.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1651.         }

  1652.         else {  //bypass the filter

  1653.             if(stereo)

  1654.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1655.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1656.                                   * NoteVoicePar[nvoice].Panning * 2.0f;

  1657.                     bypassr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1658.                                   * (1.0f

  1659.                                      - NoteVoicePar[nvoice].Panning) * 2.0f;

  1660.                 }

  1661.             else

  1662.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1663.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1664.         }

  1665.         // chech if there is necesary to proces the voice longer (if the Amplitude envelope isn't finished)

  1666.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1667.             if(NoteVoicePar[nvoice].AmpEnvelope->finished())

  1668.                 KillVoice(nvoice);

  1669.     }

  1670. 

  1671. 

  1672.     //Processing Global parameters

  1673.     NoteGlobalPar.GlobalFilterL->filterout(&outl[0]);

  1674. 

  1675.     if(stereo == 0) { //set the right channel=left channel

  1676.         memcpy(outr, outl, synth.bufferbytes);

  1677.         memcpy(bypassr, bypassl, synth.bufferbytes);

  1678.     }

  1679.     else

  1680.         NoteGlobalPar.GlobalFilterR->filterout(&outr[0]);

  1681. 

  1682.     for(int i = 0; i < synth.buffersize; ++i) {

  1683.         outl[i] += bypassl[i];

  1684.         outr[i] += bypassr[i];

  1685.     }

  1686. 

  1687.     if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))

  1688.         // Amplitude Interpolation

  1689.         for(int i = 0; i < synth.buffersize; ++i) {

  1690.             float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,

  1691.                                                  globalnewamplitude,

  1692.                                                  i,

  1693.                                                  synth.buffersize);

  1694.             outl[i] *= tmpvol * NoteGlobalPar.Panning;

  1695.             outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);

  1696.         }

  1697.     else

  1698.         for(int i = 0; i < synth.buffersize; ++i) {

  1699.             outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;

  1700.             outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);

  1701.         }

  1702. 

  1703.     //Apply the punch

  1704.     if(NoteGlobalPar.Punch.Enabled != 0)

  1705.         for(int i = 0; i < synth.buffersize; ++i) {

  1706.             float punchamp = NoteGlobalPar.Punch.initialvalue

  1707.                              * NoteGlobalPar.Punch.t + 1.0f;

  1708.             outl[i] *= punchamp;

  1709.             outr[i] *= punchamp;

  1710.             NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;

  1711.             if(NoteGlobalPar.Punch.t < 0.0f) {

  1712.                 NoteGlobalPar.Punch.Enabled = 0;

  1713.                 break;

  1714.             }

  1715.         }

  1716. 

  1717. 

  1718.     // Apply legato-specific sound signal modifications

  1719.     legato.apply(*this, outl, outr);

  1720. 

  1721. 

  1722.     // Check if the global amplitude is finished.

  1723.     // If it does, disable the note

  1724.     if(NoteGlobalPar.AmpEnvelope->finished()) {

  1725.         for(int i = 0; i < synth.buffersize; ++i) { //fade-out

  1726.             float tmp = 1.0f - (float)i / synth.buffersize_f;

  1727.             outl[i] *= tmp;

  1728.             outr[i] *= tmp;

  1729.         }

  1730.         KillNote();

  1731.     }

  1732.     return 1;

  1733. }

  1734. 

  1735. 

  1736. /*

  1737.  * Release the key (NoteOff)

  1738.  */

  1739. void ADnote::releasekey()

  1740. {

  1741.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)

  1742.         NoteVoicePar[nvoice].releasekey();

  1743.     NoteGlobalPar.FreqEnvelope->releasekey();

  1744.     NoteGlobalPar.FilterEnvelope->releasekey();

  1745.     NoteGlobalPar.AmpEnvelope->releasekey();

  1746. }

  1747. 

  1748. /*

  1749.  * Check if the note is finished

  1750.  */

  1751. int ADnote::finished() const

  1752. {

  1753.     if(NoteEnabled == ON)

  1754.         return 0;

  1755.     else

  1756.         return 1;

  1757. }

  1758. 

  1759. void ADnote::Voice::releasekey()

  1760. {

  1761.     if(!Enabled)

  1762.         return;

  1763.     if(AmpEnvelope)

  1764.         AmpEnvelope->releasekey();

  1765.     if(FreqEnvelope)

  1766.         FreqEnvelope->releasekey();

  1767.     if(FilterEnvelope)

  1768.         FilterEnvelope->releasekey();

  1769.     if(FMFreqEnvelope)

  1770.         FMFreqEnvelope->releasekey();

  1771.     if(FMAmpEnvelope)

  1772.         FMAmpEnvelope->releasekey();

  1773. }

  1774. 

  1775. void ADnote::Voice::kill(Allocator &memory, const SYNTH_T &synth)

  1776. {

  1777.     memory.devalloc(OscilSmp);

  1778.     memory.dealloc(FreqEnvelope);

  1779.     memory.dealloc(FreqLfo);

  1780.     memory.dealloc(AmpEnvelope);

  1781.     memory.dealloc(AmpLfo);

  1782.     memory.dealloc(VoiceFilterL);

  1783.     memory.dealloc(VoiceFilterR);

  1784.     memory.dealloc(FilterEnvelope);

  1785.     memory.dealloc(FilterLfo);

  1786.     memory.dealloc(FMFreqEnvelope);

  1787.     memory.dealloc(FMAmpEnvelope);

  1788. 

  1789.     if((FMEnabled != NONE) && (FMVoice < 0))

  1790.         memory.devalloc(FMSmp);

  1791. 

  1792.     if(VoiceOut)

  1793.         memset(VoiceOut, 0, synth.bufferbytes);

  1794.     //the buffer can't be safely deleted as it may be

  1795.     //an input to another voice

  1796. 

  1797.     Enabled = OFF;

  1798. }

  1799. 

  1800. void ADnote::Global::kill(Allocator &memory)

  1801. {

  1802.     memory.dealloc(FreqEnvelope);

  1803.     memory.dealloc(FreqLfo);

  1804.     memory.dealloc(AmpEnvelope);

  1805.     memory.dealloc(AmpLfo);

  1806.     memory.dealloc(GlobalFilterL);

  1807.     memory.dealloc(GlobalFilterR);

  1808.     memory.dealloc(FilterEnvelope);

  1809.     memory.dealloc(FilterLfo);

  1810. }

  1811. 

  1812. void ADnote::Global::initparameters(const ADnoteGlobalParam &param,

  1813.                                     const SYNTH_T &synth,

  1814.                                     const AbsTime &time,

  1815.                                     class Allocator &memory,

  1816.                                     float basefreq, float velocity,

  1817.                                     bool stereo)

  1818. {

  1819.     FreqEnvelope = memory.alloc<Envelope>(*param.FreqEnvelope, basefreq, synth.dt());

  1820.     FreqLfo      = memory.alloc<LFO>(*param.FreqLfo, basefreq, time);

  1821. 

  1822.     AmpEnvelope = memory.alloc<Envelope>(*param.AmpEnvelope, basefreq, synth.dt());

  1823.     AmpLfo      = memory.alloc<LFO>(*param.AmpLfo, basefreq, time);

  1824. 

  1825.     Volume = 4.0f * powf(0.1f, 3.0f * (1.0f - param.PVolume / 96.0f)) //-60 dB .. 0 dB

  1826.              * VelF(velocity, param.PAmpVelocityScaleFunction);     //sensing

  1827. 

  1828.     GlobalFilterL = Filter::generate(memory, param.GlobalFilter,

  1829.             synth.samplerate, synth.buffersize);

  1830.     if(stereo)

  1831.         GlobalFilterR = Filter::generate(memory, param.GlobalFilter,

  1832.                 synth.samplerate, synth.buffersize);

  1833.     else

  1834.         GlobalFilterR = NULL;

  1835. 

  1836.     FilterEnvelope = memory.alloc<Envelope>(*param.FilterEnvelope, basefreq, synth.dt());

  1837.     FilterLfo      = memory.alloc<LFO>(*param.FilterLfo, basefreq, time);

  1838.     FilterQ = param.GlobalFilter->getq();

  1839.     FilterFreqTracking = param.GlobalFilter->getfreqtracking(basefreq);

  1840. }