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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 "../Params/ADnoteParameters.h"

  33. #include "ModFilter.h"

  34. #include "OscilGen.h"

  35. #include "ADnote.h"

  36. 

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

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

  39. {

  40.     memory.beginTransaction();

  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.Fadein_adjustment =

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

  67.     NoteGlobalPar.Fadein_adjustment *= NoteGlobalPar.Fadein_adjustment;

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

  69.         NoteGlobalPar.Punch.Enabled = 1;

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

  71.         NoteGlobalPar.Punch.initialvalue =

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

  73.              * VelF(velocity,

  74.                     pars.GlobalPar.PPunchVelocitySensing));

  75.         float time =

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

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

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

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

  80.     }

  81.     else

  82.         NoteGlobalPar.Punch.Enabled = 0;

  83. 

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

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

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

  87. 

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

  89.         NoteVoicePar[nvoice].OscilSmp = NULL;

  90.         NoteVoicePar[nvoice].FMSmp    = NULL;

  91.         NoteVoicePar[nvoice].VoiceOut = NULL;

  92. 

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

  94.         unison_size[nvoice] = 1;

  95. 

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

  97.             NoteVoicePar[nvoice].Enabled = OFF;

  98.             continue; //the voice is disabled

  99.         }

  100. 

  101.         int BendAdj = pars.VoicePar[nvoice].PBendAdjust - 64;

  102.         if (BendAdj % 24 == 0)

  103.             NoteVoicePar[nvoice].BendAdjust = BendAdj / 24;

  104.         else

  105.             NoteVoicePar[nvoice].BendAdjust = BendAdj / 24.0f;

  106. 

  107.         float offset_val = (pars.VoicePar[nvoice].POffsetHz - 64)/64.0f;

  108.         NoteVoicePar[nvoice].OffsetHz =

  109.             15.0f*(offset_val * sqrtf(fabsf(offset_val)));

  110. 

  111.         unison_stereo_spread[nvoice] =

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

  113. 

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

  115.         if(unison < 1)

  116.             unison = 1;

  117. 

  118.         bool is_pwm = pars.VoicePar[nvoice].PFMEnabled == PW_MOD;

  119. 

  120. 	if (pars.VoicePar[nvoice].Type != 0) {

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

  122.             if (unison > 2)

  123.                 unison = 2;

  124.         } else if (is_pwm) {

  125.             /* Pulse width mod uses pairs of subvoices. */

  126.             unison *= 2;

  127.             // This many is likely to sound like noise anyhow.

  128.             if (unison > 64)

  129.                 unison = 64;

  130.         }

  131. 

  132.         //compute unison

  133.         unison_size[nvoice] = unison;

  134. 

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

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

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

  138.         float unison_spread =

  139.             pars.getUnisonFrequencySpreadCents(nvoice);

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

  141.         float unison_vibratto_a  =

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

  143. 

  144.         int true_unison = unison / (is_pwm ? 2 : 1);

  145.         switch(true_unison) {

  146.             case 1:

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

  148.                 break;

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

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

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

  152.             };

  153.                 break;

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

  155.                 float unison_values[true_unison];

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

  157.                 for(int k = 0; k < true_unison; ++k) {

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

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

  160.                     unison_values[k] = val;

  161.                     if (min > val) {

  162.                         min = val;

  163.                     }

  164.                     if (max < val) {

  165.                         max = val;

  166.                     }

  167.                 }

  168.                 float diff = max - min;

  169.                 for(int k = 0; k < true_unison; ++k) {

  170.                     unison_values[k] =

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

  172.                     unison_base_freq_rap[nvoice][k] =

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

  174.                 }

  175.             };

  176.         }

  177.         if (is_pwm)

  178.             for (int i = true_unison - 1; i >= 0; i--) {

  179.                 unison_base_freq_rap[nvoice][2*i + 1] =

  180.                     unison_base_freq_rap[nvoice][i];

  181.                 unison_base_freq_rap[nvoice][2*i] =

  182.                     unison_base_freq_rap[nvoice][i];

  183.             }

  184. 

  185.         //unison vibrattos

  186.         if(unison > 2 || (!is_pwm && unison > 1))

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

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

  189.                                                   + (unison_base_freq_rap[

  190.                                                          nvoice][k] - 1.0f)

  191.                                                   * (1.0f - unison_vibratto_a);

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

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

  194.         unison_vibratto[nvoice].amplitude =

  195.             (unison_real_spread - 1.0f) * unison_vibratto_a;

  196. 

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

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

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

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

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

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

  203.             float vibratto_period = vibratto_base_period

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

  205. 

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

  207.             if(RND < 0.5f)

  208.                 m = -m;

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

  210. 

  211.             // Ugly, but the alternative is likely uglier.

  212.             if (is_pwm)

  213.                 for (int i = 0; i < unison; i += 2) {

  214.                     unison_vibratto[nvoice].step[i+1] =

  215.                         unison_vibratto[nvoice].step[i];

  216.                     unison_vibratto[nvoice].position[i+1] =

  217.                         unison_vibratto[nvoice].position[i];

  218.                 }

  219.         }

  220. 

  221.         if(unison <= 2) { //no vibratto for a single voice

  222.             if (is_pwm) {

  223.                 unison_vibratto[nvoice].step[1]     = 0.0f;

  224.                 unison_vibratto[nvoice].position[1] = 0.0f;

  225.             }

  226.             if (is_pwm || unison == 1) {

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

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

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

  230.             }

  231.         }

  232. 

  233.         //phase invert for unison

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

  235.         if(unison != 1) {

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

  237.             switch(inv) {

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

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

  240.                     break;

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

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

  243.                     break;

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

  245.                         unison_invert_phase[nvoice][k] =

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

  247.                     break;

  248.             }

  249.         }

  250. 

  251. 

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

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

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

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

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

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

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

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

  260. 

  261.         NoteVoicePar[nvoice].Enabled     = ON;

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

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

  264. 

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

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

  267.             NoteVoicePar[nvoice].Detune = getdetune(

  268.                 pars.VoicePar[nvoice].PDetuneType,

  269.                 pars.VoicePar[nvoice].

  270.                 PCoarseDetune,

  271.                 8192); //coarse detune

  272.             NoteVoicePar[nvoice].FineDetune = getdetune(

  273.                 pars.VoicePar[nvoice].PDetuneType,

  274.                 0,

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

  276.         }

  277.         else {

  278.             NoteVoicePar[nvoice].Detune = getdetune(

  279.                 pars.GlobalPar.PDetuneType,

  280.                 pars.VoicePar[nvoice].

  281.                 PCoarseDetune,

  282.                 8192); //coarse detune

  283.             NoteVoicePar[nvoice].FineDetune = getdetune(

  284.                 pars.GlobalPar.PDetuneType,

  285.                 0,

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

  287.         }

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

  289.             NoteVoicePar[nvoice].FMDetune = getdetune(

  290.                 pars.VoicePar[nvoice].PFMDetuneType,

  291.                 pars.VoicePar[nvoice].

  292.                 PFMCoarseDetune,

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

  294.         else

  295.             NoteVoicePar[nvoice].FMDetune = getdetune(

  296.                 pars.GlobalPar.PDetuneType,

  297.                 pars.VoicePar[nvoice].

  298.                 PFMCoarseDetune,

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

  300. 

  301. 

  302. 

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

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

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

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

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

  308.         }

  309. 

  310.         //the extra points contains the first point

  311.         NoteVoicePar[nvoice].OscilSmp =

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

  313. 

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

  315.         int vc = nvoice;

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

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

  318.         if(!pars.GlobalPar.Hrandgrouping)

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

  320.         int oscposhi_start =

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

  322.                                              getvoicebasefreq(nvoice),

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

  324. 

  325.         // This code was planned for biasing the carrier in MOD_RING

  326.         // but that's on hold for the moment.  Disabled 'cos small

  327.         // machines run this stuff too.

  328.         //

  329.         // //Find range of generated wave

  330.         // float min = NoteVoicePar[nvoice].OscilSmp[0];

  331.         // float max = min;

  332.         // float *smpls = &(NoteVoicePar[nvoice].OscilSmp[1]);

  333.         // for (int i = synth.oscilsize-1; i--; smpls++)

  334.         //     if (*smpls > max)

  335.         //         max = *smpls;

  336.         //     else if (*smpls < min)

  337.         //         min = *smpls;

  338.         // NoteVoicePar[nvoice].OscilSmpMin = min;

  339.         // NoteVoicePar[nvoice].OscilSmpMax = max;

  340. 

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

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

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

  344.                                           + i] =

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

  346. 

  347.         NoteVoicePar[nvoice].phase_offset =

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

  349.                    - 64.0f) / 128.0f * synth.oscilsize

  350.                   + synth.oscilsize * 4);

  351.         oscposhi_start += NoteVoicePar[nvoice].phase_offset;

  352. 

  353.         int kth_start = oscposhi_start;

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

  355.             oscposhi[nvoice][k] = kth_start % synth.oscilsize;

  356.             //put random starting point for other subvoices

  357.             kth_start      = oscposhi_start +

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

  359.                         127.0f * (synth.oscilsize - 1));

  360.         }

  361. 

  362.         NoteVoicePar[nvoice].FreqLfo      = NULL;

  363.         NoteVoicePar[nvoice].FreqEnvelope = NULL;

  364. 

  365.         NoteVoicePar[nvoice].AmpLfo      = NULL;

  366.         NoteVoicePar[nvoice].AmpEnvelope = NULL;

  367. 

  368.         NoteVoicePar[nvoice].Filter         = NULL;

  369.         NoteVoicePar[nvoice].FilterEnvelope = NULL;

  370.         NoteVoicePar[nvoice].FilterLfo      = NULL;

  371. 

  372.         NoteVoicePar[nvoice].filterbypass =

  373.             pars.VoicePar[nvoice].Pfilterbypass;

  374. 

  375.         if (pars.VoicePar[nvoice].Type != 0)

  376.             NoteVoicePar[nvoice].FMEnabled = NONE;

  377.         else

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

  379.                 case 1:

  380.                     NoteVoicePar[nvoice].FMEnabled = MORPH;

  381.                     break;

  382.                 case 2:

  383.                     NoteVoicePar[nvoice].FMEnabled = RING_MOD;

  384.                     break;

  385.                 case 3:

  386.                     NoteVoicePar[nvoice].FMEnabled = PHASE_MOD;

  387.                     break;

  388.                 case 4:

  389.                     NoteVoicePar[nvoice].FMEnabled = FREQ_MOD;

  390.                     break;

  391.                 case 5:

  392.                     NoteVoicePar[nvoice].FMEnabled = PW_MOD;

  393.                     break;

  394.                 default:

  395.                     NoteVoicePar[nvoice].FMEnabled = NONE;

  396.             }

  397. 

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

  399.         NoteVoicePar[nvoice].FMFreqEnvelope = NULL;

  400.         NoteVoicePar[nvoice].FMAmpEnvelope  = NULL;

  401.         NoteVoicePar[nvoice].FMFreqFixed  = pars.VoicePar[nvoice].PFMFixedFreq;

  402. 

  403. 

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

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

  406.                                    nvoice),

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

  408.                                - 1.0f);

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

  410.             case PHASE_MOD:

  411.             case PW_MOD:

  412.                 fmvoldamp =

  413.                     powf(440.0f / getvoicebasefreq(

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

  415.                          / 64.0f);

  416.                 NoteVoicePar[nvoice].FMVolume =

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

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

  419.                 break;

  420.             case FREQ_MOD:

  421.                 NoteVoicePar[nvoice].FMVolume =

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

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

  424.                 break;

  425.             default:

  426.                 if(fmvoldamp > 1.0f)

  427.                     fmvoldamp = 1.0f;

  428.                 NoteVoicePar[nvoice].FMVolume =

  429.                     pars.VoicePar[nvoice].PFMVolume

  430.                     / 127.0f * fmvoldamp;

  431.         }

  432. 

  433.         //Voice's modulator velocity sensing

  434.         NoteVoicePar[nvoice].FMVolume *=

  435.             VelF(velocity,

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

  437. 

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

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

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

  441. 

  442.         firsttick[nvoice] = 1;

  443.         NoteVoicePar[nvoice].DelayTicks =

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

  445.                         * logf(50.0f))

  446.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  447.     }

  448. 

  449.     max_unison = 1;

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

  451.         if(unison_size[nvoice] > max_unison)

  452.             max_unison = unison_size[nvoice];

  453. 

  454. 

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

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

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

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

  459.     }

  460. 

  461.     initparameters();

  462.     memory.endTransaction();

  463. }

  464. 

  465. SynthNote *ADnote::cloneLegato(void)

  466. {

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

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

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

  470. }

  471. 

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

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

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

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

  476. void ADnote::legatonote(LegatoParams lpars)

  477. {

  478.     //ADnoteParameters &pars = *partparams;

  479. 

  480.     // Manage legato stuff

  481.     if(legato.update(lpars))

  482.         return;

  483. 

  484.     portamento = lpars.portamento;

  485.     midinote   = lpars.midinote;

  486.     basefreq   = lpars.frequency;

  487. 

  488.     if(velocity > 1.0f)

  489.         velocity = 1.0f;

  490.     velocity = lpars.velocity;

  491. 

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

  493.                                      pars.GlobalPar.PCoarseDetune,

  494.                                      pars.GlobalPar.PDetune);

  495.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  496. 

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

  498.         NoteGlobalPar.Panning = RND;

  499.     else

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

  501. 

  502.     NoteGlobalPar.Filter->updateSense(velocity, 

  503.                 pars.GlobalPar.PFilterVelocityScale,

  504.                 pars.GlobalPar.PFilterVelocityScaleFunction);

  505. 

  506. 

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

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

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

  510. 

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

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

  513. 

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

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

  516.             NoteVoicePar[nvoice].Detune = getdetune(

  517.                 pars.VoicePar[nvoice].PDetuneType,

  518.                 pars.VoicePar[nvoice].PCoarseDetune,

  519.                 8192); //coarse detune

  520.             NoteVoicePar[nvoice].FineDetune = getdetune(

  521.                 pars.VoicePar[nvoice].PDetuneType,

  522.                 0,

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

  524.         }

  525.         else {

  526.             NoteVoicePar[nvoice].Detune = getdetune(

  527.                 pars.GlobalPar.PDetuneType,

  528.                 pars.VoicePar[nvoice].PCoarseDetune,

  529.                 8192); //coarse detune

  530.             NoteVoicePar[nvoice].FineDetune = getdetune(

  531.                 pars.GlobalPar.PDetuneType,

  532.                 0,

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

  534.         }

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

  536.             NoteVoicePar[nvoice].FMDetune = getdetune(

  537.                 pars.VoicePar[nvoice].PFMDetuneType,

  538.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  540.         else

  541.             NoteVoicePar[nvoice].FMDetune = getdetune(

  542.                 pars.GlobalPar.PDetuneType,

  543.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  545. 

  546. 

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

  548.         int vc = nvoice;

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

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

  551.         if(!pars.GlobalPar.Hrandgrouping)

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

  553. 

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

  555.                                          getvoicebasefreq(nvoice),

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

  557. 

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

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

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

  561.                                           + i] =

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

  563. 

  564.         auto &voiceFilter = NoteVoicePar[nvoice].Filter;

  565.         if(voiceFilter) {

  566.             const auto  &vce     = pars.VoicePar[nvoice];

  567.             voiceFilter->updateSense(velocity, vce.PFilterVelocityScale,

  568.                         vce.PFilterVelocityScaleFunction);

  569.         }

  570. 

  571.         NoteVoicePar[nvoice].filterbypass =

  572.             pars.VoicePar[nvoice].Pfilterbypass;

  573. 

  574. 

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

  576. 

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

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

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

  580.                                - 1.0f);

  581. 

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

  583.             case PHASE_MOD:

  584.             case PW_MOD:

  585.                 fmvoldamp =

  586.                     powf(440.0f / getvoicebasefreq(

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

  588.                          / 64.0f);

  589.                 NoteVoicePar[nvoice].FMVolume =

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

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

  592.                 break;

  593.             case FREQ_MOD:

  594.                 NoteVoicePar[nvoice].FMVolume =

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

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

  597.                 break;

  598.             default:

  599.                 if(fmvoldamp > 1.0f)

  600.                     fmvoldamp = 1.0f;

  601.                 NoteVoicePar[nvoice].FMVolume =

  602.                     pars.VoicePar[nvoice].PFMVolume

  603.                     / 127.0f * fmvoldamp;

  604.         }

  605. 

  606.         //Voice's modulator velocity sensing

  607.         NoteVoicePar[nvoice].FMVolume *=

  608.             VelF(velocity,

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

  610. 

  611.         NoteVoicePar[nvoice].DelayTicks =

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

  613.                         * logf(50.0f))

  614.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  615.     }

  616. 

  617.     ///    initparameters();

  618. 

  619.     ///////////////

  620.     // Altered content of initparameters():

  621. 

  622.     int tmp[NUM_VOICES];

  623. 

  624.     NoteGlobalPar.Volume = 4.0f

  625.                            * powf(0.1f, 3.0f

  626.                                   * (1.0f - pars.GlobalPar.PVolume

  627.                                      / 96.0f))                                      //-60 dB .. 0 dB

  628.                            * VelF(

  629.         velocity,

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

  631. 

  632.     globalnewamplitude = NoteGlobalPar.Volume

  633.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  635. 

  636.     {

  637.         auto        *filter  = NoteGlobalPar.Filter;

  638.         filter->updateSense(velocity, pars.GlobalPar.PFilterVelocityScale,

  639.                             pars.GlobalPar.PFilterVelocityScaleFunction);

  640.         filter->updateNoteFreq(basefreq);

  641.     }

  642. 

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

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

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

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

  647. 

  648.     // Voice Parameter init

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

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

  651.             continue;

  652. 

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

  654.         /* Voice Amplitude Parameters Init */

  655.         NoteVoicePar[nvoice].Volume =

  656.             powf(0.1f, 3.0f

  657.                  * (1.0f - pars.VoicePar[nvoice].PVolume / 127.0f))             // -60 dB .. 0 dB

  658.             * VelF(velocity,

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

  660. 

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

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

  663. 

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

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

  666.         else

  667.             NoteVoicePar[nvoice].Panning =

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

  669. 

  670.         newamplitude[nvoice] = 1.0f;

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

  672.            && NoteVoicePar[nvoice].AmpEnvelope)

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

  674. 

  675. 

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

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

  678. 

  679.         auto *voiceFilter = NoteVoicePar[nvoice].Filter;

  680.         if(voiceFilter) {

  681.             voiceFilter->updateSense(velocity, pars.VoicePar[nvoice].PFilterVelocityScale,

  682.                                      pars.VoicePar[nvoice].PFilterVelocityScaleFunction);

  683.             voiceFilter->updateNoteFreq(basefreq);

  684.         }

  685. 

  686.         /* Voice Modulation Parameters Init */

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

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

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

  690. 

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

  692. 

  693.             int vc = nvoice;

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

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

  696. 

  697.             if(!pars.GlobalPar.Hrandgrouping)

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

  699. 

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

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

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

  703.         }

  704. 

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

  706.                                  * ctl.fmamp.relamp;

  707. 

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

  709.            && NoteVoicePar[nvoice].FMAmpEnvelope)

  710.             FMnewamplitude[nvoice] *=

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

  712.     }

  713. 

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

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

  716.             tmp[i] = 0;

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

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

  719.                 tmp[i] = 1;

  720.     }

  721. }

  722. 

  723. 

  724. /*

  725.  * Kill a voice of ADnote

  726.  */

  727. void ADnote::KillVoice(int nvoice)

  728. {

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

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

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

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

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

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

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

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

  737. 

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

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

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

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

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

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

  744. 

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

  746. }

  747. 

  748. /*

  749.  * Kill the note

  750.  */

  751. void ADnote::KillNote()

  752. {

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

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

  755.             KillVoice(nvoice);

  756. 

  757.         if(NoteVoicePar[nvoice].VoiceOut)

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

  759.     }

  760. 

  761.     NoteGlobalPar.kill(memory);

  762. 

  763.     NoteEnabled = OFF;

  764. }

  765. 

  766. ADnote::~ADnote()

  767. {

  768.     if(NoteEnabled == ON)

  769.         KillNote();

  770.     memory.devalloc(tmpwavel);

  771.     memory.devalloc(tmpwaver);

  772.     memory.devalloc(bypassl);

  773.     memory.devalloc(bypassr);

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

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

  776.     memory.devalloc(tmpwave_unison);

  777. }

  778. 

  779. 

  780. /*

  781.  * Init the parameters

  782.  */

  783. void ADnote::initparameters()

  784. {

  785.     int tmp[NUM_VOICES];

  786.     //ADnoteParameters &pars = *partparams;

  787. 

  788.     // Global Parameters

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

  790.                                  time,

  791.                                  memory, basefreq, velocity,

  792.                                  stereo);

  793. 

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

  795.     globalnewamplitude = NoteGlobalPar.Volume

  796.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  798. 

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

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

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

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

  803. 

  804.     // Voice Parameter init

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

  806.         Voice &vce = NoteVoicePar[nvoice];

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

  808. 

  809.         if(vce.Enabled == 0)

  810.             continue;

  811. 

  812.         vce.noisetype = param.Type;

  813.         /* Voice Amplitude Parameters Init */

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

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

  816. 

  817.         if(param.PVolumeminus)

  818.             vce.Volume = -vce.Volume;

  819. 

  820.         if(param.PPanning == 0)

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

  822.         else

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

  824. 

  825.         newamplitude[nvoice] = 1.0f;

  826.         if(param.PAmpEnvelopeEnabled) {

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

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

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

  830.         }

  831. 

  832.         if(param.PAmpLfoEnabled) {

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

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

  835.         }

  836. 

  837.         /* Voice Frequency Parameters Init */

  838.         if(param.PFreqEnvelopeEnabled)

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

  840. 

  841.         if(param.PFreqLfoEnabled)

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

  843. 

  844.         /* Voice Filter Parameters Init */

  845.         if(param.PFilterEnabled) {

  846.             vce.Filter = memory.alloc<ModFilter>(*param.VoiceFilter, synth, time, memory, stereo,

  847.                       basefreq);

  848.             vce.Filter->updateSense(velocity, param.PFilterVelocityScale,

  849.                     param.PFilterVelocityScaleFunction);

  850. 

  851. 

  852.             if(param.PFilterEnvelopeEnabled) {

  853.                 vce.FilterEnvelope =

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

  855.                 vce.Filter->addMod(*vce.FilterEnvelope);

  856.             }

  857. 

  858.             if(param.PFilterLfoEnabled) {

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

  860.                 vce.Filter->addMod(*vce.FilterLfo);

  861.             }

  862.         }

  863. 

  864.         /* Voice Modulation Parameters Init */

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

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

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

  868. 

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

  870. 

  871.             int vc = nvoice;

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

  873.                 vc = param.PextFMoscil;

  874. 

  875.             float tmp = 1.0f;

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

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

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

  879.                 tmp = getFMvoicebasefreq(nvoice);

  880. 

  881.             if(!pars.GlobalPar.Hrandgrouping)

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

  883. 

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

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

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

  887.                                              vce.FMSmp, tmp))

  888.                                         % synth.oscilsize;

  889. 

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

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

  892.             int oscposhiFM_add =

  893.                 (int)((param.PFMoscilphase

  894.                        - 64.0f) / 128.0f * synth.oscilsize

  895.                       + synth.oscilsize * 4);

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

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

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

  899.             }

  900.         }

  901. 

  902.         if(param.PFMFreqEnvelopeEnabled)

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

  904. 

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

  906. 

  907.         if(param.PFMAmpEnvelopeEnabled ) {

  908.             vce.FMAmpEnvelope =

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

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

  911.         }

  912.     }

  913. 

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

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

  916.             tmp[i] = 0;

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

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

  919.                 NoteVoicePar[nvoice].VoiceOut =

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

  921.                 tmp[i] = 1;

  922.             }

  923. 

  924.         if(NoteVoicePar[nvoice].VoiceOut)

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

  926.     }

  927. }

  928. 

  929. 

  930. /*

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

  932.  * This must be called before setfreq* functions

  933.  */

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

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

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

  937.         return;

  938.     }

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

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

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

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

  943.         pos += step;

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

  945.             pos  = -1.0f;

  946.             step = -step;

  947.         }

  948.         if(pos >= 1.0f) {

  949.             pos  = 1.0f;

  950.             step = -step;

  951.         }

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

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

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

  955.                                          - 1.0f) + vibratto_val

  956.                                         * unison_vibratto[nvoice].amplitude)

  957.                                      * relbw;

  958. 

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

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

  961.     }

  962. }

  963. 

  964. 

  965. /*

  966.  * Computes the frequency of an oscillator

  967.  */

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

  969. {

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

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

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

  973.         if(speed > synth.oscilsize_f)

  974.             speed = synth.oscilsize_f;

  975. 

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

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

  978.     }

  979. }

  980. 

  981. /*

  982.  * Computes the frequency of an modullator oscillator

  983.  */

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

  985. {

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

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

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

  989.         if(speed > synth.samplerate_f)

  990.             speed = synth.samplerate_f;

  991. 

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

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

  994.     }

  995. }

  996. 

  997. /*

  998.  * Get Voice base frequency

  999.  */

  1000. float ADnote::getvoicebasefreq(int nvoice) const

  1001. {

  1002.     float detune = NoteVoicePar[nvoice].Detune / 100.0f

  1003.                    + NoteVoicePar[nvoice].FineDetune / 100.0f

  1004.                    * ctl.bandwidth.relbw * bandwidthDetuneMultiplier

  1005.                    + NoteGlobalPar.Detune / 100.0f;

  1006. 

  1007.     if(NoteVoicePar[nvoice].fixedfreq == 0)

  1008.         return this->basefreq * powf(2, detune / 12.0f);

  1009.     else { //the fixed freq is enabled

  1010.         float fixedfreq   = 440.0f;

  1011.         int   fixedfreqET = NoteVoicePar[nvoice].fixedfreqET;

  1012.         if(fixedfreqET != 0) { //if the frequency varies according the keyboard note

  1013.             float tmp =

  1014.                 (midinote

  1015.                  - 69.0f) / 12.0f

  1016.                 * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);

  1017.             if(fixedfreqET <= 64)

  1018.                 fixedfreq *= powf(2.0f, tmp);

  1019.             else

  1020.                 fixedfreq *= powf(3.0f, tmp);

  1021.         }

  1022.         return fixedfreq * powf(2.0f, detune / 12.0f);

  1023.     }

  1024. }

  1025. 

  1026. /*

  1027.  * Get Voice's Modullator base frequency

  1028.  */

  1029. float ADnote::getFMvoicebasefreq(int nvoice) const

  1030. {

  1031.     float detune = NoteVoicePar[nvoice].FMDetune / 100.0f;

  1032.     return getvoicebasefreq(nvoice) * powf(2, detune / 12.0f);

  1033. }

  1034. 

  1035. /*

  1036.  * Computes all the parameters for each tick

  1037.  */

  1038. void ADnote::computecurrentparameters()

  1039. {

  1040.     int   nvoice;

  1041.     float voicefreq, voicepitch, FMfreq,

  1042.           FMrelativepitch, globalpitch;

  1043.     globalpitch = 0.01f * (NoteGlobalPar.FreqEnvelope->envout()

  1044.                            + NoteGlobalPar.FreqLfo->lfoout()

  1045.                            * ctl.modwheel.relmod);

  1046.     globaloldamplitude = globalnewamplitude;

  1047.     globalnewamplitude = NoteGlobalPar.Volume

  1048.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

  1049.                          * NoteGlobalPar.AmpLfo->amplfoout();

  1050. 

  1051.     NoteGlobalPar.Filter->update(ctl.filtercutoff.relfreq,

  1052.                                  ctl.filterq.relq);

  1053. 

  1054. 

  1055.     //compute the portamento, if it is used by this note

  1056.     float portamentofreqrap = 1.0f;

  1057.     if(portamento != 0) { //this voice use portamento

  1058.         portamentofreqrap = ctl.portamento.freqrap;

  1059.         if(ctl.portamento.used == 0) //the portamento has finished

  1060.             portamento = 0;  //this note is no longer "portamented"

  1061.     }

  1062. 

  1063.     //compute parameters for all voices

  1064.     for(nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1065.         if(NoteVoicePar[nvoice].Enabled != ON)

  1066.             continue;

  1067.         NoteVoicePar[nvoice].DelayTicks -= 1;

  1068.         if(NoteVoicePar[nvoice].DelayTicks > 0)

  1069.             continue;

  1070. 

  1071.         compute_unison_freq_rap(nvoice);

  1072. 

  1073.         /*******************/

  1074.         /* Voice Amplitude */

  1075.         /*******************/

  1076.         oldamplitude[nvoice] = newamplitude[nvoice];

  1077.         newamplitude[nvoice] = 1.0f;

  1078. 

  1079.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1080.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpEnvelope->envout_dB();

  1081. 

  1082.         if(NoteVoicePar[nvoice].AmpLfo)

  1083.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpLfo->amplfoout();

  1084. 

  1085.         /****************/

  1086.         /* Voice Filter */

  1087.         /****************/

  1088.         auto *voiceFilter = NoteVoicePar[nvoice].Filter;

  1089.         if(voiceFilter) {

  1090.             voiceFilter->update(ctl.filtercutoff.relfreq,

  1091.                                 ctl.filterq.relq);

  1092.         }

  1093. 

  1094.         if(NoteVoicePar[nvoice].noisetype == 0) { //compute only if the voice isn't noise

  1095.             /*******************/

  1096.             /* Voice Frequency */

  1097.             /*******************/

  1098.             voicepitch = 0.0f;

  1099.             if(NoteVoicePar[nvoice].FreqLfo)

  1100.                 voicepitch += NoteVoicePar[nvoice].FreqLfo->lfoout() / 100.0f

  1101.                               * ctl.bandwidth.relbw;

  1102. 

  1103.             if(NoteVoicePar[nvoice].FreqEnvelope)

  1104.                 voicepitch += NoteVoicePar[nvoice].FreqEnvelope->envout()

  1105.                               / 100.0f;

  1106.             voicefreq = getvoicebasefreq(nvoice)

  1107.                         * powf(2, (voicepitch + globalpitch) / 12.0f);                //Hz frequency

  1108.             voicefreq *=

  1109.                 powf(ctl.pitchwheel.relfreq, NoteVoicePar[nvoice].BendAdjust); //change the frequency by the controller

  1110.             setfreq(nvoice, voicefreq * portamentofreqrap + NoteVoicePar[nvoice].OffsetHz);

  1111. 

  1112.             /***************/

  1113.             /*  Modulator */

  1114.             /***************/

  1115. 

  1116. 

  1117.             if(NoteVoicePar[nvoice].FMEnabled != NONE) {

  1118.                 FMrelativepitch = NoteVoicePar[nvoice].FMDetune / 100.0f;

  1119.                 if(NoteVoicePar[nvoice].FMFreqEnvelope)

  1120.                     FMrelativepitch +=

  1121.                         NoteVoicePar[nvoice].FMFreqEnvelope->envout() / 100;

  1122.                 if (NoteVoicePar[nvoice].FMFreqFixed)

  1123.                     FMfreq =

  1124.                         powf(2.0f, FMrelativepitch

  1125.                              / 12.0f) * 440.0f;

  1126.                 else

  1127.                     FMfreq =

  1128.                            powf(2.0f, FMrelativepitch

  1129.                                 / 12.0f) * voicefreq * portamentofreqrap;

  1130.                 setfreqFM(nvoice, FMfreq);

  1131. 

  1132.                 FMoldamplitude[nvoice] = FMnewamplitude[nvoice];

  1133.                 FMnewamplitude[nvoice] = NoteVoicePar[nvoice].FMVolume

  1134.                                          * ctl.fmamp.relamp;

  1135.                 if(NoteVoicePar[nvoice].FMAmpEnvelope)

  1136.                     FMnewamplitude[nvoice] *=

  1137.                         NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();

  1138.             }

  1139.         }

  1140.     }

  1141. }

  1142. 

  1143. 

  1144. /*

  1145.  * Fadein in a way that removes clicks but keep sound "punchy"

  1146.  */

  1147. inline void ADnote::fadein(float *smps) const

  1148. {

  1149.     int zerocrossings = 0;

  1150.     for(int i = 1; i < synth.buffersize; ++i)

  1151.         if((smps[i - 1] < 0.0f) && (smps[i] > 0.0f))

  1152.             zerocrossings++;  //this is only the possitive crossings

  1153. 

  1154.     float tmp = (synth.buffersize_f - 1.0f) / (zerocrossings + 1) / 3.0f;

  1155.     if(tmp < 8.0f)

  1156.         tmp = 8.0f;

  1157.     tmp *= NoteGlobalPar.Fadein_adjustment;

  1158. 

  1159.     int n;

  1160.     F2I(tmp, n); //how many samples is the fade-in

  1161.     if(n > synth.buffersize)

  1162.         n = synth.buffersize;

  1163.     for(int i = 0; i < n; ++i) { //fade-in

  1164.         float tmp = 0.5f - cosf((float)i / (float) n * PI) * 0.5f;

  1165.         smps[i] *= tmp;

  1166.     }

  1167. }

  1168. 

  1169. /*

  1170.  * Computes the Oscillator (Without Modulation) - LinearInterpolation

  1171.  */

  1172. 

  1173. /* As the code here is a bit odd due to optimization, here is what happens

  1174.  * First the current possition and frequency are retrieved from the running

  1175.  * state. These are broken up into high and low portions to indicate how many

  1176.  * samples are skipped in one step and how many fractional samples are skipped.

  1177.  * Outside of this method the fractional samples are just handled with floating

  1178.  * point code, but that's a bit slower than it needs to be. In this code the low

  1179.  * portions are known to exist between 0.0 and 1.0 and it is known that they are

  1180.  * stored in single precision floating point IEEE numbers. This implies that

  1181.  * a maximum of 24 bits are significant. The below code does your standard

  1182.  * linear interpolation that you'll see throughout this codebase, but by

  1183.  * sticking to integers for tracking the overflow of the low portion, around 15%

  1184.  * of the execution time was shaved off in the ADnote test.

  1185.  */

  1186. inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)

  1187. {

  1188.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1189.         int    poshi  = oscposhi[nvoice][k];

  1190.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1191.         int    freqhi = oscfreqhi[nvoice][k];

  1192.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1193.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1194.         float *tw     = tmpwave_unison[k];

  1195.         assert(oscfreqlo[nvoice][k] < 1.0f);

  1196.         for(int i = 0; i < synth.buffersize; ++i) {

  1197.             tw[i]  = (smps[poshi] * ((1<<24) - poslo) + smps[poshi + 1] * poslo)/(1.0f*(1<<24));

  1198.             poslo += freqlo;

  1199.             poshi += freqhi + (poslo>>24);

  1200.             poslo &= 0xffffff;

  1201.             poshi &= synth.oscilsize - 1;

  1202.         }

  1203.         oscposhi[nvoice][k] = poshi;

  1204.         oscposlo[nvoice][k] = poslo/(1.0f*(1<<24));

  1205.     }

  1206. }

  1207. 

  1208. 

  1209. 

  1210. /*

  1211.  * Computes the Oscillator (Without Modulation) - CubicInterpolation

  1212.  *

  1213.  The differences from the Linear are to little to deserve to be used. This is because I am using a large synth.oscilsize (>512)

  1214. inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){

  1215.     int i,poshi;

  1216.     float poslo;

  1217. 

  1218.     poshi=oscposhi[nvoice];

  1219.     poslo=oscposlo[nvoice];

  1220.     float *smps=NoteVoicePar[nvoice].OscilSmp;

  1221.     float xm1,x0,x1,x2,a,b,c;

  1222.     for (i=0;i<synth.buffersize;i++){

  1223.     xm1=smps[poshi];

  1224.     x0=smps[poshi+1];

  1225.     x1=smps[poshi+2];

  1226.     x2=smps[poshi+3];

  1227.     a=(3.0f * (x0-x1) - xm1 + x2) / 2.0f;

  1228.     b = 2.0f*x1 + xm1 - (5.0f*x0 + x2) / 2.0f;

  1229.     c = (x1 - xm1) / 2.0f;

  1230.     tmpwave[i]=(((a * poslo) + b) * poslo + c) * poslo + x0;

  1231.     printf("a\n");

  1232.     //tmpwave[i]=smps[poshi]*(1.0f-poslo)+smps[poshi+1]*poslo;

  1233.     poslo+=oscfreqlo[nvoice];

  1234.     if (poslo>=1.0f) {

  1235.             poslo-=1.0f;

  1236.         poshi++;

  1237.     };

  1238.         poshi+=oscfreqhi[nvoice];

  1239.         poshi&=synth.oscilsize-1;

  1240.     };

  1241.     oscposhi[nvoice]=poshi;

  1242.     oscposlo[nvoice]=poslo;

  1243. };

  1244. */

  1245. /*

  1246.  * Computes the Oscillator (Morphing)

  1247.  */

  1248. inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)

  1249. {

  1250.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1251.     if(FMnewamplitude[nvoice] > 1.0f)

  1252.         FMnewamplitude[nvoice] = 1.0f;

  1253.     if(FMoldamplitude[nvoice] > 1.0f)

  1254.         FMoldamplitude[nvoice] = 1.0f;

  1255. 

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

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

  1258.         int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1259.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1260.             float *tw = tmpwave_unison[k];

  1261.             for(int i = 0; i < synth.buffersize; ++i) {

  1262.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1263.                                             FMnewamplitude[nvoice],

  1264.                                             i,

  1265.                                             synth.buffersize);

  1266.                 tw[i] = tw[i]

  1267.                     * (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1268.             }

  1269.         }

  1270.     }

  1271.     else

  1272.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1273.             int    poshiFM  = oscposhiFM[nvoice][k];

  1274.             float  posloFM  = oscposloFM[nvoice][k];

  1275.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1276.             float  freqloFM = oscfreqloFM[nvoice][k];

  1277.             float *tw = tmpwave_unison[k];

  1278. 

  1279.             for(int i = 0; i < synth.buffersize; ++i) {

  1280.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1281.                                             FMnewamplitude[nvoice],

  1282.                                             i,

  1283.                                             synth.buffersize);

  1284.                 tw[i] = tw[i] * (1.0f - amp) + amp

  1285.                         * (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1 - posloFM)

  1286.                            + NoteVoicePar[nvoice].FMSmp[poshiFM + 1] * posloFM);

  1287.                 posloFM += freqloFM;

  1288.                 if(posloFM >= 1.0f) {

  1289.                     posloFM -= 1.0f;

  1290.                     poshiFM++;

  1291.                 }

  1292.                 poshiFM += freqhiFM;

  1293.                 poshiFM &= synth.oscilsize - 1;

  1294.             }

  1295.             oscposhiFM[nvoice][k] = poshiFM;

  1296.             oscposloFM[nvoice][k] = posloFM;

  1297.         }

  1298. }

  1299. 

  1300. /*

  1301.  * Computes the Oscillator (Ring Modulation)

  1302.  */

  1303. inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)

  1304. {

  1305.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1306.     if(FMnewamplitude[nvoice] > 1.0f)

  1307.         FMnewamplitude[nvoice] = 1.0f;

  1308.     if(FMoldamplitude[nvoice] > 1.0f)

  1309.         FMoldamplitude[nvoice] = 1.0f;

  1310.     if(NoteVoicePar[nvoice].FMVoice >= 0)

  1311.         // if I use VoiceOut[] as modullator

  1312.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1313.             float *tw = tmpwave_unison[k];

  1314.             for(int i = 0; i < synth.buffersize; ++i) {

  1315.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1316.                                             FMnewamplitude[nvoice],

  1317.                                             i,

  1318.                                             synth.buffersize);

  1319.                 int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1320.                 tw[i] *= (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1321.             }

  1322.         }

  1323.     else

  1324.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1325.             int    poshiFM  = oscposhiFM[nvoice][k];

  1326.             float  posloFM  = oscposloFM[nvoice][k];

  1327.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1328.             float  freqloFM = oscfreqloFM[nvoice][k];

  1329.             float *tw = tmpwave_unison[k];

  1330. 

  1331.             for(int i = 0; i < synth.buffersize; ++i) {

  1332.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1333.                                             FMnewamplitude[nvoice],

  1334.                                             i,

  1335.                                             synth.buffersize);

  1336.                 tw[i] *= (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1.0f - posloFM)

  1337.                           + NoteVoicePar[nvoice].FMSmp[poshiFM

  1338.                                                        + 1] * posloFM) * amp

  1339.                          + (1.0f - amp);

  1340.                 posloFM += freqloFM;

  1341.                 if(posloFM >= 1.0f) {

  1342.                     posloFM -= 1.0f;

  1343.                     poshiFM++;

  1344.                 }

  1345.                 poshiFM += freqhiFM;

  1346.                 poshiFM &= synth.oscilsize - 1;

  1347.             }

  1348.             oscposhiFM[nvoice][k] = poshiFM;

  1349.             oscposloFM[nvoice][k] = posloFM;

  1350.         }

  1351. }

  1352. 

  1353. /*

  1354.  * Computes the Oscillator (Phase Modulation or Frequency Modulation)

  1355.  */

  1356. inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,

  1357.                                                               int FMmode)

  1358. {

  1359.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1360.         //if I use VoiceOut[] as modulator

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

  1362.             float *tw = tmpwave_unison[k];

  1363.             const float *smps = NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut;

  1364.             if (FMmode == PW_MOD && (k & 1))

  1365.                 for (int i = 0; i < synth.buffersize; ++i)

  1366.                     tw[i] = -smps[i];

  1367.             else

  1368.                 memcpy(tw, smps, synth.bufferbytes);

  1369.         }

  1370.     } else {

  1371.         //Compute the modulator and store it in tmpwave_unison[][]

  1372.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1373.             int    poshiFM  = oscposhiFM[nvoice][k];

  1374.             int    posloFM  = oscposloFM[nvoice][k]  * (1<<24);

  1375.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1376.             int    freqloFM = oscfreqloFM[nvoice][k] * (1<<24);

  1377.             float *tw = tmpwave_unison[k];

  1378.             const float *smps = NoteVoicePar[nvoice].FMSmp;

  1379. 

  1380.             for(int i = 0; i < synth.buffersize; ++i) {

  1381.                 tw[i] = (smps[poshiFM] * ((1<<24) - posloFM)

  1382.                          + smps[poshiFM + 1] * posloFM) / (1.0f*(1<<24));

  1383.                 if (FMmode == PW_MOD && (k & 1))

  1384.                     tw[i] = -tw[i];

  1385. 

  1386.                 posloFM += freqloFM;

  1387.                 if(posloFM >= (1<<24)) {

  1388.                     posloFM &= 0xffffff;//fmod(posloFM, 1.0f);

  1389.                     poshiFM++;

  1390.                 }

  1391.                 poshiFM += freqhiFM;

  1392.                 poshiFM &= synth.oscilsize - 1;

  1393.             }

  1394.             oscposhiFM[nvoice][k] = poshiFM;

  1395.             oscposloFM[nvoice][k] = posloFM/((1<<24)*1.0f);

  1396.         }

  1397.     }

  1398.     // Amplitude interpolation

  1399.     if(ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],

  1400.                                  FMnewamplitude[nvoice])) {

  1401.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1402.             float *tw = tmpwave_unison[k];

  1403.             for(int i = 0; i < synth.buffersize; ++i)

  1404.                 tw[i] *= INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1405.                                                FMnewamplitude[nvoice],

  1406.                                                i,

  1407.                                                synth.buffersize);

  1408.         }

  1409.     } else {

  1410.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1411.             float *tw = tmpwave_unison[k];

  1412.             for(int i = 0; i < synth.buffersize; ++i)

  1413.                 tw[i] *= FMnewamplitude[nvoice];

  1414.         }

  1415.     }

  1416. 

  1417. 

  1418.     //normalize: makes all sample-rates, oscil_sizes to produce same sound

  1419.     if(FMmode == FREQ_MOD) { //Frequency modulation

  1420.         const float normalize = synth.oscilsize_f / 262144.0f * 44100.0f

  1421.                           / synth.samplerate_f;

  1422.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1423.             float *tw    = tmpwave_unison[k];

  1424.             float  fmold = FMoldsmp[nvoice][k];

  1425.             for(int i = 0; i < synth.buffersize; ++i) {

  1426.                 fmold = fmod(fmold + tw[i] * normalize, synth.oscilsize);

  1427.                 tw[i] = fmold;

  1428.             }

  1429.             FMoldsmp[nvoice][k] = fmold;

  1430.         }

  1431.     }

  1432.     else {  //Phase or PWM modulation

  1433.         const float normalize = synth.oscilsize_f / 262144.0f;

  1434.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1435.             float *tw = tmpwave_unison[k];

  1436.             for(int i = 0; i < synth.buffersize; ++i)

  1437.                 tw[i] *= normalize;

  1438.         }

  1439.     }

  1440. 

  1441.     //do the modulation

  1442.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1443.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1444.         float *tw     = tmpwave_unison[k];

  1445.         int    poshi  = oscposhi[nvoice][k];

  1446.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1447.         int    freqhi = oscfreqhi[nvoice][k];

  1448.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1449. 

  1450.         for(int i = 0; i < synth.buffersize; ++i) {

  1451.             int FMmodfreqhi = 0;

  1452.             F2I(tw[i], FMmodfreqhi);

  1453.             float FMmodfreqlo = tw[i]-FMmodfreqhi;//fmod(tw[i] /*+ 0.0000000001f*/, 1.0f);

  1454.             if(FMmodfreqhi < 0)

  1455.                 FMmodfreqlo++;

  1456. 

  1457.             //carrier

  1458.             int carposhi = poshi + FMmodfreqhi;

  1459.             int carposlo = poslo + FMmodfreqlo;

  1460.             if (FMmode == PW_MOD && (k & 1))

  1461.                 carposhi += NoteVoicePar[nvoice].phase_offset;

  1462. 

  1463.             if(carposlo >= (1<<24)) {

  1464.                 carposhi++;

  1465.                 carposlo &= 0xffffff;//fmod(carposlo, 1.0f);

  1466.             }

  1467.             carposhi &= (synth.oscilsize - 1);

  1468. 

  1469.             tw[i] = (smps[carposhi] * ((1<<24) - carposlo)

  1470.                     + smps[carposhi + 1] * carposlo)/(1.0f*(1<<24));

  1471. 

  1472.             poslo += freqlo;

  1473.             if(poslo >= (1<<24)) {

  1474.                 poslo &= 0xffffff;//fmod(poslo, 1.0f);

  1475.                 poshi++;

  1476.             }

  1477. 

  1478.             poshi += freqhi;

  1479.             poshi &= synth.oscilsize - 1;

  1480.         }

  1481.         oscposhi[nvoice][k] = poshi;

  1482.         oscposlo[nvoice][k] = (poslo)/((1<<24)*1.0f);

  1483.     }

  1484. }

  1485. 

  1486. 

  1487. /*

  1488.  * Computes the Noise

  1489.  */

  1490. inline void ADnote::ComputeVoiceWhiteNoise(int nvoice)

  1491. {

  1492.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1493.         float *tw = tmpwave_unison[k];

  1494.         for(int i = 0; i < synth.buffersize; ++i)

  1495.             tw[i] = RND * 2.0f - 1.0f;

  1496.     }

  1497. }

  1498. 

  1499. inline void ADnote::ComputeVoicePinkNoise(int nvoice)

  1500. {

  1501.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1502.         float *tw = tmpwave_unison[k];

  1503.         float *f = &pinking[nvoice][k > 0 ? 7 : 0];

  1504.         for(int i = 0; i < synth.buffersize; ++i) {

  1505. 	    float white = (RND-0.5)/4.0;

  1506. 	    f[0] = 0.99886*f[0]+white*0.0555179;

  1507. 	    f[1] = 0.99332*f[1]+white*0.0750759;

  1508. 	    f[2] = 0.96900*f[2]+white*0.1538520;

  1509. 	    f[3] = 0.86650*f[3]+white*0.3104856;

  1510. 	    f[4] = 0.55000*f[4]+white*0.5329522;

  1511. 	    f[5] = -0.7616*f[5]-white*0.0168980;

  1512. 	    tw[i] = f[0]+f[1]+f[2]+f[3]+f[4]+f[5]+f[6]+white*0.5362;

  1513. 	    f[6] = white*0.115926;

  1514.         }

  1515.     }

  1516. }

  1517. 

  1518. 

  1519. 

  1520. /*

  1521.  * Compute the ADnote samples

  1522.  * Returns 0 if the note is finished

  1523.  */

  1524. int ADnote::noteout(float *outl, float *outr)

  1525. {

  1526.     memcpy(outl, synth.denormalkillbuf, synth.bufferbytes);

  1527.     memcpy(outr, synth.denormalkillbuf, synth.bufferbytes);

  1528. 

  1529.     if(NoteEnabled == OFF)

  1530.         return 0;

  1531. 

  1532.     memset(bypassl, 0, synth.bufferbytes);

  1533.     memset(bypassr, 0, synth.bufferbytes);

  1534.     computecurrentparameters();

  1535. 

  1536.     for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1537.         if((NoteVoicePar[nvoice].Enabled != ON)

  1538.            || (NoteVoicePar[nvoice].DelayTicks > 0))

  1539.             continue;

  1540.         switch (NoteVoicePar[nvoice].noisetype) {

  1541.             case 0: //voice mode=sound

  1542.                 switch(NoteVoicePar[nvoice].FMEnabled) {

  1543.                     case MORPH:

  1544.                         ComputeVoiceOscillatorMorph(nvoice);

  1545.                         break;

  1546.                     case RING_MOD:

  1547.                         ComputeVoiceOscillatorRingModulation(nvoice);

  1548.                         break;

  1549.                     case FREQ_MOD:

  1550.                     case PHASE_MOD:

  1551.                     case PW_MOD:

  1552.                         ComputeVoiceOscillatorFrequencyModulation(nvoice,

  1553.                                                                   NoteVoicePar[nvoice].FMEnabled);

  1554.                         break;

  1555.                     default:

  1556.                         ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1557.                         //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice);

  1558.                 }

  1559.                 break;

  1560.             case 1:

  1561.                 ComputeVoiceWhiteNoise(nvoice);

  1562.                 break;

  1563.             default:

  1564.                 ComputeVoicePinkNoise(nvoice);

  1565.                 break;

  1566.         }

  1567.         // Voice Processing

  1568. 

  1569. 

  1570.         //mix subvoices into voice

  1571.         memset(tmpwavel, 0, synth.bufferbytes);

  1572.         if(stereo)

  1573.             memset(tmpwaver, 0, synth.bufferbytes);

  1574.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1575.             float *tw = tmpwave_unison[k];

  1576.             if(stereo) {

  1577.                 float stereo_pos = 0;

  1578.                 bool is_pwm = NoteVoicePar[nvoice].FMEnabled == PW_MOD;

  1579.                 if (is_pwm) {

  1580.                     if(unison_size[nvoice] > 2)

  1581.                         stereo_pos = k/2

  1582.                             / (float)(unison_size[nvoice]/2

  1583.                                       - 1) * 2.0f - 1.0f;

  1584.                 } else if(unison_size[nvoice] > 1) {

  1585.                     stereo_pos = k

  1586.                         / (float)(unison_size[nvoice]

  1587.                                   - 1) * 2.0f - 1.0f;

  1588.                 }

  1589.                 float stereo_spread = unison_stereo_spread[nvoice] * 2.0f; //between 0 and 2.0f

  1590.                 if(stereo_spread > 1.0f) {

  1591.                     float stereo_pos_1 = (stereo_pos >= 0.0f) ? 1.0f : -1.0f;

  1592.                     stereo_pos =

  1593.                         (2.0f

  1594.                          - stereo_spread) * stereo_pos

  1595.                         + (stereo_spread - 1.0f) * stereo_pos_1;

  1596.                 }

  1597.                 else

  1598.                     stereo_pos *= stereo_spread;

  1599. 

  1600.                 if(unison_size[nvoice] == 1 ||

  1601.                    (is_pwm && unison_size[nvoice] == 2))

  1602.                     stereo_pos = 0.0f;

  1603.                 float panning = (stereo_pos + 1.0f) * 0.5f;

  1604. 

  1605. 

  1606.                 float lvol = (1.0f - panning) * 2.0f;

  1607.                 if(lvol > 1.0f)

  1608.                     lvol = 1.0f;

  1609. 

  1610.                 float rvol = panning * 2.0f;

  1611.                 if(rvol > 1.0f)

  1612.                     rvol = 1.0f;

  1613. 

  1614.                 if(unison_invert_phase[nvoice][k]) {

  1615.                     lvol = -lvol;

  1616.                     rvol = -rvol;

  1617.                 }

  1618. 

  1619.                 for(int i = 0; i < synth.buffersize; ++i)

  1620.                     tmpwavel[i] += tw[i] * lvol;

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

  1622.                     tmpwaver[i] += tw[i] * rvol;

  1623.             }

  1624.             else

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

  1626.                     tmpwavel[i] += tw[i];

  1627.         }

  1628. 

  1629. 

  1630.         float unison_amplitude = 1.0f / sqrt(unison_size[nvoice]); //reduce the amplitude for large unison sizes

  1631.         // Amplitude

  1632.         float oldam = oldamplitude[nvoice] * unison_amplitude;

  1633.         float newam = newamplitude[nvoice] * unison_amplitude;

  1634. 

  1635.         if(ABOVE_AMPLITUDE_THRESHOLD(oldam, newam)) {

  1636.             int rest = synth.buffersize;

  1637.             //test if the amplitude if raising and the difference is high

  1638.             if((newam > oldam) && ((newam - oldam) > 0.25f)) {

  1639.                 rest = 10;

  1640.                 if(rest > synth.buffersize)

  1641.                     rest = synth.buffersize;

  1642.                 for(int i = 0; i < synth.buffersize - rest; ++i)

  1643.                     tmpwavel[i] *= oldam;

  1644.                 if(stereo)

  1645.                     for(int i = 0; i < synth.buffersize - rest; ++i)

  1646.                         tmpwaver[i] *= oldam;

  1647.             }

  1648.             // Amplitude interpolation

  1649.             for(int i = 0; i < rest; ++i) {

  1650.                 float amp = INTERPOLATE_AMPLITUDE(oldam, newam, i, rest);

  1651.                 tmpwavel[i + (synth.buffersize - rest)] *= amp;

  1652.                 if(stereo)

  1653.                     tmpwaver[i + (synth.buffersize - rest)] *= amp;

  1654.             }

  1655.         }

  1656.         else {

  1657.             for(int i = 0; i < synth.buffersize; ++i)

  1658.                 tmpwavel[i] *= newam;

  1659.             if(stereo)

  1660.                 for(int i = 0; i < synth.buffersize; ++i)

  1661.                     tmpwaver[i] *= newam;

  1662.         }

  1663. 

  1664.         // Fade in

  1665.         if(firsttick[nvoice] != 0) {

  1666.             fadein(&tmpwavel[0]);

  1667.             if(stereo)

  1668.                 fadein(&tmpwaver[0]);

  1669.             firsttick[nvoice] = 0;

  1670.         }

  1671. 

  1672.         // Filter

  1673.         if(NoteVoicePar[nvoice].Filter) {

  1674.             if(stereo)

  1675.                 NoteVoicePar[nvoice].Filter->filter(tmpwavel, tmpwaver);

  1676.             else

  1677.                 NoteVoicePar[nvoice].Filter->filter(tmpwavel, 0);

  1678.         }

  1679. 

  1680.         //check if the amplitude envelope is finished, if yes, the voice will be fadeout

  1681.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1682.             if(NoteVoicePar[nvoice].AmpEnvelope->finished()) {

  1683.                 for(int i = 0; i < synth.buffersize; ++i)

  1684.                     tmpwavel[i] *= 1.0f - (float)i / synth.buffersize_f;

  1685.                 if(stereo)

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

  1687.                         tmpwaver[i] *= 1.0f - (float)i / synth.buffersize_f;

  1688.             }

  1689.         //the voice is killed later

  1690. 

  1691. 

  1692.         // Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator)

  1693.         if(NoteVoicePar[nvoice].VoiceOut) {

  1694.             if(stereo)

  1695.                 for(int i = 0; i < synth.buffersize; ++i)

  1696.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i]

  1697.                                                        + tmpwaver[i];

  1698.             else   //mono

  1699.                 for(int i = 0; i < synth.buffersize; ++i)

  1700.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i];

  1701.         }

  1702. 

  1703. 

  1704.         // Add the voice that do not bypass the filter to out

  1705.         if(NoteVoicePar[nvoice].filterbypass == 0) { //no bypass

  1706.             if(stereo)

  1707.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1708.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1709.                                * NoteVoicePar[nvoice].Panning * 2.0f;

  1710.                     outr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1711.                                * (1.0f - NoteVoicePar[nvoice].Panning) * 2.0f;

  1712.                 }

  1713.             else

  1714.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1715.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1716.         }

  1717.         else {  //bypass the filter

  1718.             if(stereo)

  1719.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1720.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1721.                                   * NoteVoicePar[nvoice].Panning * 2.0f;

  1722.                     bypassr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1723.                                   * (1.0f

  1724.                                      - NoteVoicePar[nvoice].Panning) * 2.0f;

  1725.                 }

  1726.             else

  1727.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1728.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1729.         }

  1730.         // chech if there is necesary to proces the voice longer (if the Amplitude envelope isn't finished)

  1731.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1732.             if(NoteVoicePar[nvoice].AmpEnvelope->finished())

  1733.                 KillVoice(nvoice);

  1734.     }

  1735. 

  1736. 

  1737.     //Processing Global parameters

  1738.     if(stereo) {

  1739.         NoteGlobalPar.Filter->filter(outl, outr);

  1740.     } else { //set the right channel=left channel

  1741.         NoteGlobalPar.Filter->filter(outl, 0);

  1742.         memcpy(outr, outl, synth.bufferbytes);

  1743.         memcpy(bypassr, bypassl, synth.bufferbytes);

  1744.     }

  1745. 

  1746.     for(int i = 0; i < synth.buffersize; ++i) {

  1747.         outl[i] += bypassl[i];

  1748.         outr[i] += bypassr[i];

  1749.     }

  1750. 

  1751.     if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))

  1752.         // Amplitude Interpolation

  1753.         for(int i = 0; i < synth.buffersize; ++i) {

  1754.             float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,

  1755.                                                  globalnewamplitude,

  1756.                                                  i,

  1757.                                                  synth.buffersize);

  1758.             outl[i] *= tmpvol * NoteGlobalPar.Panning;

  1759.             outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);

  1760.         }

  1761.     else

  1762.         for(int i = 0; i < synth.buffersize; ++i) {

  1763.             outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;

  1764.             outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);

  1765.         }

  1766. 

  1767.     //Apply the punch

  1768.     if(NoteGlobalPar.Punch.Enabled != 0)

  1769.         for(int i = 0; i < synth.buffersize; ++i) {

  1770.             float punchamp = NoteGlobalPar.Punch.initialvalue

  1771.                              * NoteGlobalPar.Punch.t + 1.0f;

  1772.             outl[i] *= punchamp;

  1773.             outr[i] *= punchamp;

  1774.             NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;

  1775.             if(NoteGlobalPar.Punch.t < 0.0f) {

  1776.                 NoteGlobalPar.Punch.Enabled = 0;

  1777.                 break;

  1778.             }

  1779.         }

  1780. 

  1781. 

  1782.     // Apply legato-specific sound signal modifications

  1783.     legato.apply(*this, outl, outr);

  1784. 

  1785. 

  1786.     // Check if the global amplitude is finished.

  1787.     // If it does, disable the note

  1788.     if(NoteGlobalPar.AmpEnvelope->finished()) {

  1789.         for(int i = 0; i < synth.buffersize; ++i) { //fade-out

  1790.             float tmp = 1.0f - (float)i / synth.buffersize_f;

  1791.             outl[i] *= tmp;

  1792.             outr[i] *= tmp;

  1793.         }

  1794.         KillNote();

  1795.     }

  1796.     return 1;

  1797. }

  1798. 

  1799. 

  1800. /*

  1801.  * Release the key (NoteOff)

  1802.  */

  1803. void ADnote::releasekey()

  1804. {

  1805.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)

  1806.         NoteVoicePar[nvoice].releasekey();

  1807.     NoteGlobalPar.FreqEnvelope->releasekey();

  1808.     NoteGlobalPar.FilterEnvelope->releasekey();

  1809.     NoteGlobalPar.AmpEnvelope->releasekey();

  1810. }

  1811. 

  1812. /*

  1813.  * Check if the note is finished

  1814.  */

  1815. bool ADnote::finished() const

  1816. {

  1817.     if(NoteEnabled == ON)

  1818.         return 0;

  1819.     else

  1820.         return 1;

  1821. }

  1822. 

  1823. void ADnote::entomb(void)

  1824. {

  1825.     NoteGlobalPar.AmpEnvelope->forceFinish();

  1826. }

  1827. 

  1828. void ADnote::Voice::releasekey()

  1829. {

  1830.     if(!Enabled)

  1831.         return;

  1832.     if(AmpEnvelope)

  1833.         AmpEnvelope->releasekey();

  1834.     if(FreqEnvelope)

  1835.         FreqEnvelope->releasekey();

  1836.     if(FilterEnvelope)

  1837.         FilterEnvelope->releasekey();

  1838.     if(FMFreqEnvelope)

  1839.         FMFreqEnvelope->releasekey();

  1840.     if(FMAmpEnvelope)

  1841.         FMAmpEnvelope->releasekey();

  1842. }

  1843. 

  1844. void ADnote::Voice::kill(Allocator &memory, const SYNTH_T &synth)

  1845. {

  1846.     memory.devalloc(OscilSmp);

  1847.     memory.dealloc(FreqEnvelope);

  1848.     memory.dealloc(FreqLfo);

  1849.     memory.dealloc(AmpEnvelope);

  1850.     memory.dealloc(AmpLfo);

  1851.     memory.dealloc(Filter);

  1852.     memory.dealloc(FilterEnvelope);

  1853.     memory.dealloc(FilterLfo);

  1854.     memory.dealloc(FMFreqEnvelope);

  1855.     memory.dealloc(FMAmpEnvelope);

  1856. 

  1857.     if((FMEnabled != NONE) && (FMVoice < 0))

  1858.         memory.devalloc(FMSmp);

  1859. 

  1860.     if(VoiceOut)

  1861.         memset(VoiceOut, 0, synth.bufferbytes);

  1862.     //the buffer can't be safely deleted as it may be

  1863.     //an input to another voice

  1864. 

  1865.     Enabled = OFF;

  1866. }

  1867. 

  1868. void ADnote::Global::kill(Allocator &memory)

  1869. {

  1870.     memory.dealloc(FreqEnvelope);

  1871.     memory.dealloc(FreqLfo);

  1872.     memory.dealloc(AmpEnvelope);

  1873.     memory.dealloc(AmpLfo);

  1874.     memory.dealloc(Filter);

  1875.     memory.dealloc(FilterEnvelope);

  1876.     memory.dealloc(FilterLfo);

  1877. }

  1878. 

  1879. void ADnote::Global::initparameters(const ADnoteGlobalParam &param,

  1880.                                     const SYNTH_T &synth,

  1881.                                     const AbsTime &time,

  1882.                                     class Allocator &memory,

  1883.                                     float basefreq, float velocity,

  1884.                                     bool stereo)

  1885. {

  1886.     FreqEnvelope = memory.alloc<Envelope>(*param.FreqEnvelope, basefreq, synth.dt());

  1887.     FreqLfo      = memory.alloc<LFO>(*param.FreqLfo, basefreq, time);

  1888. 

  1889.     AmpEnvelope = memory.alloc<Envelope>(*param.AmpEnvelope, basefreq, synth.dt());

  1890.     AmpLfo      = memory.alloc<LFO>(*param.AmpLfo, basefreq, time);

  1891. 

  1892.     Volume = 4.0f * powf(0.1f, 3.0f * (1.0f - param.PVolume / 96.0f)) //-60 dB .. 0 dB

  1893.              * VelF(velocity, param.PAmpVelocityScaleFunction);     //sensing

  1894. 

  1895.     Filter = memory.alloc<ModFilter>(*param.GlobalFilter, synth, time, memory, 

  1896.             stereo, basefreq);

  1897. 

  1898.     FilterEnvelope = memory.alloc<Envelope>(*param.FilterEnvelope, basefreq, synth.dt());

  1899.     FilterLfo      = memory.alloc<LFO>(*param.FilterLfo, basefreq, time);

  1900. 

  1901.     Filter->addMod(*FilterEnvelope);

  1902.     Filter->addMod(*FilterLfo);

  1903.     

  1904.     {

  1905.         Filter->updateSense(velocity, param.PFilterVelocityScale,

  1906.                 param.PFilterVelocityScaleFunction);

  1907.     }

  1908. }