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

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  1. /*

  2.   ZynAddSubFX - a software synthesizer

  3. 

  4.   ADnote.cpp - The "additive" synthesizer

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

  6.   Author: Nasca Octavian Paul

  7. 

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

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

  10.   as published by the Free Software Foundation.

  11. 

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

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

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

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

  16. 

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

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

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

  20. */

  21. 

  22. #include <cmath>

  23. #include <cstdlib>

  24. #include <cstdio>

  25. #include <cstring>

  26. #include <cassert>

  27. #include <stdint.h>

  28. 

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

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

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

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

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

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

  35. #include "OscilGen.h"

  36. #include "ADnote.h"

  37. 

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

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

  40. {

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

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

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

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

  45. 

  46.     ADnoteParameters &pars = *pars_;

  47.     portamento  = spars.portamento;

  48.     midinote    = spars.note;

  49.     NoteEnabled = ON;

  50.     basefreq    = spars.frequency;

  51.     velocity    = spars.velocity;

  52.     stereo = pars.GlobalPar.PStereo;

  53. 

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

  55.                                      pars.GlobalPar.PCoarseDetune,

  56.                                      pars.GlobalPar.PDetune);

  57.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  58. 

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

  60.         NoteGlobalPar.Panning = RND;

  61.     else

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

  63. 

  64. 

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

  66.                                       + pars.GlobalPar.PFilterVelocityScale

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

  68.                                       * (VelF(velocity,

  69.                                               pars.GlobalPar.

  70.                                               PFilterVelocityScaleFunction) - 1);

  71. 

  72.     NoteGlobalPar.Fadein_adjustment =

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

  74.     NoteGlobalPar.Fadein_adjustment *= NoteGlobalPar.Fadein_adjustment;

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

  76.         NoteGlobalPar.Punch.Enabled = 1;

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

  78.         NoteGlobalPar.Punch.initialvalue =

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

  80.              * VelF(velocity,

  81.                     pars.GlobalPar.PPunchVelocitySensing));

  82.         float time =

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

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

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

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

  87.     }

  88.     else

  89.         NoteGlobalPar.Punch.Enabled = 0;

  90. 

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

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

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

  94. 

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

  96.         NoteVoicePar[nvoice].OscilSmp = NULL;

  97.         NoteVoicePar[nvoice].FMSmp    = NULL;

  98.         NoteVoicePar[nvoice].VoiceOut = NULL;

  99. 

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

  101.         unison_size[nvoice] = 1;

  102. 

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

  104.             NoteVoicePar[nvoice].Enabled = OFF;

  105.             continue; //the voice is disabled

  106.         }

  107. 

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

  109.         if (BendAdj % 24 == 0)

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

  111.         else

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

  113. 

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

  115.         NoteVoicePar[nvoice].OffsetHz =

  116.             15.0f*(offset_val * sqrtf(fabsf(offset_val)));

  117. 

  118.         unison_stereo_spread[nvoice] =

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

  120. 

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

  122.         if(unison < 1)

  123.             unison = 1;

  124. 

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

  126. 

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

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

  129.             if (unison > 2)

  130.                 unison = 2;

  131.         } else if (is_pwm) {

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

  133.             unison *= 2;

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

  135.             if (unison > 64)

  136.                 unison = 64;

  137.         }

  138. 

  139.         //compute unison

  140.         unison_size[nvoice] = unison;

  141. 

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

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

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

  145.         float unison_spread =

  146.             pars.getUnisonFrequencySpreadCents(nvoice);

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

  148.         float unison_vibratto_a  =

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

  150. 

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

  152.         switch(true_unison) {

  153.             case 1:

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

  155.                 break;

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

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

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

  159.             };

  160.                 break;

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

  162.                 float unison_values[true_unison];

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

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

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

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

  167.                     unison_values[k] = val;

  168.                     if (min > val) {

  169.                         min = val;

  170.                     }

  171.                     if (max < val) {

  172.                         max = val;

  173.                     }

  174.                 }

  175.                 float diff = max - min;

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

  177.                     unison_values[k] =

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

  179.                     unison_base_freq_rap[nvoice][k] =

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

  181.                 }

  182.             };

  183.         }

  184.         if (is_pwm)

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

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

  187.                     unison_base_freq_rap[nvoice][i];

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

  189.                     unison_base_freq_rap[nvoice][i];

  190.             }

  191. 

  192.         //unison vibrattos

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

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

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

  196.                                                   + (unison_base_freq_rap[

  197.                                                          nvoice][k] - 1.0f)

  198.                                                   * (1.0f - unison_vibratto_a);

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

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

  201.         unison_vibratto[nvoice].amplitude =

  202.             (unison_real_spread - 1.0f) * unison_vibratto_a;

  203. 

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

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

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

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

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

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

  210.             float vibratto_period = vibratto_base_period

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

  212. 

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

  214.             if(RND < 0.5f)

  215.                 m = -m;

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

  217. 

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

  219.             if (is_pwm)

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

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

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

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

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

  225.                 }

  226.         }

  227. 

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

  229.             if (is_pwm) {

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

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

  232.             }

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

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

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

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

  237.             }

  238.         }

  239. 

  240.         //phase invert for unison

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

  242.         if(unison != 1) {

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

  244.             switch(inv) {

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

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

  247.                     break;

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

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

  250.                     break;

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

  252.                         unison_invert_phase[nvoice][k] =

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

  254.                     break;

  255.             }

  256.         }

  257. 

  258. 

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

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

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

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

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

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

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

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

  267. 

  268.         NoteVoicePar[nvoice].Enabled     = ON;

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

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

  271. 

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

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

  274.             NoteVoicePar[nvoice].Detune = getdetune(

  275.                 pars.VoicePar[nvoice].PDetuneType,

  276.                 pars.VoicePar[nvoice].

  277.                 PCoarseDetune,

  278.                 8192); //coarse detune

  279.             NoteVoicePar[nvoice].FineDetune = getdetune(

  280.                 pars.VoicePar[nvoice].PDetuneType,

  281.                 0,

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

  283.         }

  284.         else {

  285.             NoteVoicePar[nvoice].Detune = getdetune(

  286.                 pars.GlobalPar.PDetuneType,

  287.                 pars.VoicePar[nvoice].

  288.                 PCoarseDetune,

  289.                 8192); //coarse detune

  290.             NoteVoicePar[nvoice].FineDetune = getdetune(

  291.                 pars.GlobalPar.PDetuneType,

  292.                 0,

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

  294.         }

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

  296.             NoteVoicePar[nvoice].FMDetune = getdetune(

  297.                 pars.VoicePar[nvoice].PFMDetuneType,

  298.                 pars.VoicePar[nvoice].

  299.                 PFMCoarseDetune,

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

  301.         else

  302.             NoteVoicePar[nvoice].FMDetune = getdetune(

  303.                 pars.GlobalPar.PDetuneType,

  304.                 pars.VoicePar[nvoice].

  305.                 PFMCoarseDetune,

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

  307. 

  308. 

  309. 

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

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

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

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

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

  315.         }

  316. 

  317.         //the extra points contains the first point

  318.         NoteVoicePar[nvoice].OscilSmp =

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

  320. 

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

  322.         int vc = nvoice;

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

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

  325.         if(!pars.GlobalPar.Hrandgrouping)

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

  327.         int oscposhi_start =

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

  329.                                              getvoicebasefreq(nvoice),

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

  331. 

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

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

  334.         // machines run this stuff too.

  335.         //

  336.         // //Find range of generated wave

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

  338.         // float max = min;

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

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

  341.         //     if (*smpls > max)

  342.         //         max = *smpls;

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

  344.         //         min = *smpls;

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

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

  347. 

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

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

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

  351.                                           + i] =

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

  353. 

  354.         NoteVoicePar[nvoice].phase_offset =

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

  356.                    - 64.0f) / 128.0f * synth.oscilsize

  357.                   + synth.oscilsize * 4);

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

  359. 

  360.         int kth_start = oscposhi_start;

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

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

  363.             //put random starting point for other subvoices

  364.             kth_start      = oscposhi_start +

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

  366.                         127.0f * (synth.oscilsize - 1));

  367.         }

  368. 

  369.         NoteVoicePar[nvoice].FreqLfo      = NULL;

  370.         NoteVoicePar[nvoice].FreqEnvelope = NULL;

  371. 

  372.         NoteVoicePar[nvoice].AmpLfo      = NULL;

  373.         NoteVoicePar[nvoice].AmpEnvelope = NULL;

  374. 

  375.         NoteVoicePar[nvoice].VoiceFilterL   = NULL;

  376.         NoteVoicePar[nvoice].VoiceFilterR   = NULL;

  377.         NoteVoicePar[nvoice].FilterEnvelope = NULL;

  378.         NoteVoicePar[nvoice].FilterLfo      = NULL;

  379. 

  380.         NoteVoicePar[nvoice].FilterCenterPitch =

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

  382.             + pars.VoicePar[nvoice].PFilterVelocityScale

  383.             / 127.0f * 6.0f       //velocity sensing

  384.             * (VelF(velocity,

  385.                     pars.VoicePar[nvoice].PFilterVelocityScaleFunction) - 1);

  386.         NoteVoicePar[nvoice].filterbypass =

  387.             pars.VoicePar[nvoice].Pfilterbypass;

  388. 

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

  390.             NoteVoicePar[nvoice].FMEnabled = NONE;

  391.         else

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

  393.                 case 1:

  394.                     NoteVoicePar[nvoice].FMEnabled = MORPH;

  395.                     break;

  396.                 case 2:

  397.                     NoteVoicePar[nvoice].FMEnabled = RING_MOD;

  398.                     break;

  399.                 case 3:

  400.                     NoteVoicePar[nvoice].FMEnabled = PHASE_MOD;

  401.                     break;

  402.                 case 4:

  403.                     NoteVoicePar[nvoice].FMEnabled = FREQ_MOD;

  404.                     break;

  405.                 case 5:

  406.                     NoteVoicePar[nvoice].FMEnabled = PW_MOD;

  407.                     break;

  408.                 default:

  409.                     NoteVoicePar[nvoice].FMEnabled = NONE;

  410.             }

  411. 

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

  413.         NoteVoicePar[nvoice].FMFreqEnvelope = NULL;

  414.         NoteVoicePar[nvoice].FMAmpEnvelope  = NULL;

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

  416. 

  417. 

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

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

  420.                                    nvoice),

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

  422.                                - 1.0f);

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

  424.             case PHASE_MOD:

  425.             case PW_MOD:

  426.                 fmvoldamp =

  427.                     powf(440.0f / getvoicebasefreq(

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

  429.                          / 64.0f);

  430.                 NoteVoicePar[nvoice].FMVolume =

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

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

  433.                 break;

  434.             case FREQ_MOD:

  435.                 NoteVoicePar[nvoice].FMVolume =

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

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

  438.                 break;

  439.             default:

  440.                 if(fmvoldamp > 1.0f)

  441.                     fmvoldamp = 1.0f;

  442.                 NoteVoicePar[nvoice].FMVolume =

  443.                     pars.VoicePar[nvoice].PFMVolume

  444.                     / 127.0f * fmvoldamp;

  445.         }

  446. 

  447.         //Voice's modulator velocity sensing

  448.         NoteVoicePar[nvoice].FMVolume *=

  449.             VelF(velocity,

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

  451. 

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

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

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

  455. 

  456.         firsttick[nvoice] = 1;

  457.         NoteVoicePar[nvoice].DelayTicks =

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

  459.                         * logf(50.0f))

  460.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  461.     }

  462. 

  463.     max_unison = 1;

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

  465.         if(unison_size[nvoice] > max_unison)

  466.             max_unison = unison_size[nvoice];

  467. 

  468. 

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

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

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

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

  473.     }

  474. 

  475.     initparameters();

  476. }

  477. 

  478. SynthNote *ADnote::cloneLegato(void)

  479. {

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

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

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

  483. }

  484. 

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

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

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

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

  489. void ADnote::legatonote(LegatoParams lpars)

  490. {

  491.     //ADnoteParameters &pars = *partparams;

  492. 

  493.     // Manage legato stuff

  494.     if(legato.update(lpars))

  495.         return;

  496. 

  497.     portamento = lpars.portamento;

  498.     midinote   = lpars.midinote;

  499.     basefreq   = lpars.frequency;

  500. 

  501.     if(velocity > 1.0f)

  502.         velocity = 1.0f;

  503.     velocity = lpars.velocity;

  504. 

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

  506.                                      pars.GlobalPar.PCoarseDetune,

  507.                                      pars.GlobalPar.PDetune);

  508.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  509. 

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

  511.         NoteGlobalPar.Panning = RND;

  512.     else

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

  514. 

  515.     //center freq

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

  517.                                       + pars.GlobalPar.PFilterVelocityScale

  518.                                       / 127.0f * 6.0f          //velocity sensing

  519.                                       * (VelF(velocity,

  520.                                               pars.GlobalPar.

  521.                                               PFilterVelocityScaleFunction) - 1);

  522. 

  523. 

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

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

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

  527. 

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

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

  530. 

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

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

  533.             NoteVoicePar[nvoice].Detune = getdetune(

  534.                 pars.VoicePar[nvoice].PDetuneType,

  535.                 pars.VoicePar[nvoice].PCoarseDetune,

  536.                 8192); //coarse detune

  537.             NoteVoicePar[nvoice].FineDetune = getdetune(

  538.                 pars.VoicePar[nvoice].PDetuneType,

  539.                 0,

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

  541.         }

  542.         else {

  543.             NoteVoicePar[nvoice].Detune = getdetune(

  544.                 pars.GlobalPar.PDetuneType,

  545.                 pars.VoicePar[nvoice].PCoarseDetune,

  546.                 8192); //coarse detune

  547.             NoteVoicePar[nvoice].FineDetune = getdetune(

  548.                 pars.GlobalPar.PDetuneType,

  549.                 0,

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

  551.         }

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

  553.             NoteVoicePar[nvoice].FMDetune = getdetune(

  554.                 pars.VoicePar[nvoice].PFMDetuneType,

  555.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  557.         else

  558.             NoteVoicePar[nvoice].FMDetune = getdetune(

  559.                 pars.GlobalPar.PDetuneType,

  560.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  562. 

  563. 

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

  565.         int vc = nvoice;

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

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

  568.         if(!pars.GlobalPar.Hrandgrouping)

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

  570. 

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

  572.                                          getvoicebasefreq(nvoice),

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

  574. 

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

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

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

  578.                                           + i] =

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

  580. 

  581. 

  582.         NoteVoicePar[nvoice].FilterCenterPitch =

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

  584.             + pars.VoicePar[nvoice].PFilterVelocityScale

  585.             / 127.0f * 6.0f       //velocity sensing

  586.             * (VelF(velocity,

  587.                     pars.VoicePar[nvoice].PFilterVelocityScaleFunction) - 1);

  588. 

  589.         NoteVoicePar[nvoice].filterbypass =

  590.             pars.VoicePar[nvoice].Pfilterbypass;

  591. 

  592. 

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

  594. 

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

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

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

  598.                                - 1.0f);

  599. 

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

  601.             case PHASE_MOD:

  602.             case PW_MOD:

  603.                 fmvoldamp =

  604.                     powf(440.0f / getvoicebasefreq(

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

  606.                          / 64.0f);

  607.                 NoteVoicePar[nvoice].FMVolume =

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

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

  610.                 break;

  611.             case FREQ_MOD:

  612.                 NoteVoicePar[nvoice].FMVolume =

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

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

  615.                 break;

  616.             default:

  617.                 if(fmvoldamp > 1.0f)

  618.                     fmvoldamp = 1.0f;

  619.                 NoteVoicePar[nvoice].FMVolume =

  620.                     pars.VoicePar[nvoice].PFMVolume

  621.                     / 127.0f * fmvoldamp;

  622.         }

  623. 

  624.         //Voice's modulator velocity sensing

  625.         NoteVoicePar[nvoice].FMVolume *=

  626.             VelF(velocity,

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

  628. 

  629.         NoteVoicePar[nvoice].DelayTicks =

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

  631.                         * logf(50.0f))

  632.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  633.     }

  634. 

  635.     ///    initparameters();

  636. 

  637.     ///////////////

  638.     // Altered content of initparameters():

  639. 

  640.     int tmp[NUM_VOICES];

  641. 

  642.     NoteGlobalPar.Volume = 4.0f

  643.                            * powf(0.1f, 3.0f

  644.                                   * (1.0f - pars.GlobalPar.PVolume

  645.                                      / 96.0f))                                      //-60 dB .. 0 dB

  646.                            * VelF(

  647.         velocity,

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

  649. 

  650.     globalnewamplitude = NoteGlobalPar.Volume

  651.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  653. 

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

  655.     NoteGlobalPar.FilterFreqTracking =

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

  657. 

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

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

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

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

  662. 

  663.     // Voice Parameter init

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

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

  666.             continue;

  667. 

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

  669.         /* Voice Amplitude Parameters Init */

  670.         NoteVoicePar[nvoice].Volume =

  671.             powf(0.1f, 3.0f

  672.                  * (1.0f - pars.VoicePar[nvoice].PVolume / 127.0f))             // -60 dB .. 0 dB

  673.             * VelF(velocity,

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

  675. 

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

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

  678. 

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

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

  681.         else

  682.             NoteVoicePar[nvoice].Panning =

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

  684. 

  685.         newamplitude[nvoice] = 1.0f;

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

  687.            && NoteVoicePar[nvoice].AmpEnvelope)

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

  689. 

  690. 

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

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

  693. 

  694. 

  695. 

  696.         NoteVoicePar[nvoice].FilterFreqTracking =

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

  698. 

  699.         /* Voice Modulation Parameters Init */

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

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

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

  703. 

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

  705. 

  706.             int vc = nvoice;

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

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

  709. 

  710.             if(!pars.GlobalPar.Hrandgrouping)

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

  712. 

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

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

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

  716.         }

  717. 

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

  719.                                  * ctl.fmamp.relamp;

  720. 

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

  722.            && NoteVoicePar[nvoice].FMAmpEnvelope)

  723.             FMnewamplitude[nvoice] *=

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

  725.     }

  726. 

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

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

  729.             tmp[i] = 0;

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

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

  732.                 tmp[i] = 1;

  733.     }

  734. }

  735. 

  736. 

  737. /*

  738.  * Kill a voice of ADnote

  739.  */

  740. void ADnote::KillVoice(int nvoice)

  741. {

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

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

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

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

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

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

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

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

  750. 

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

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

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

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

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

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

  757. 

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

  759. }

  760. 

  761. /*

  762.  * Kill the note

  763.  */

  764. void ADnote::KillNote()

  765. {

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

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

  768.             KillVoice(nvoice);

  769. 

  770.         if(NoteVoicePar[nvoice].VoiceOut)

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

  772.     }

  773. 

  774.     NoteGlobalPar.kill(memory);

  775. 

  776.     NoteEnabled = OFF;

  777. }

  778. 

  779. ADnote::~ADnote()

  780. {

  781.     if(NoteEnabled == ON)

  782.         KillNote();

  783.     memory.devalloc(tmpwavel);

  784.     memory.devalloc(tmpwaver);

  785.     memory.devalloc(bypassl);

  786.     memory.devalloc(bypassr);

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

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

  789.     memory.devalloc(tmpwave_unison);

  790. }

  791. 

  792. 

  793. /*

  794.  * Init the parameters

  795.  */

  796. void ADnote::initparameters()

  797. {

  798.     int tmp[NUM_VOICES];

  799.     //ADnoteParameters &pars = *partparams;

  800. 

  801.     // Global Parameters

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

  803.                                  time,

  804.                                  memory, basefreq, velocity,

  805.                                  stereo);

  806. 

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

  808.     globalnewamplitude = NoteGlobalPar.Volume

  809.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  811. 

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

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

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

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

  816. 

  817.     // Voice Parameter init

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

  819.         Voice &vce = NoteVoicePar[nvoice];

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

  821. 

  822.         if(vce.Enabled == 0)

  823.             continue;

  824. 

  825.         vce.noisetype = param.Type;

  826.         /* Voice Amplitude Parameters Init */

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

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

  829. 

  830.         if(param.PVolumeminus)

  831.             vce.Volume = -vce.Volume;

  832. 

  833.         if(param.PPanning == 0)

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

  835.         else

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

  837. 

  838.         newamplitude[nvoice] = 1.0f;

  839.         if(param.PAmpEnvelopeEnabled) {

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

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

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

  843.         }

  844. 

  845.         if(param.PAmpLfoEnabled) {

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

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

  848.         }

  849. 

  850.         /* Voice Frequency Parameters Init */

  851.         if(param.PFreqEnvelopeEnabled)

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

  853. 

  854.         if(param.PFreqLfoEnabled)

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

  856. 

  857.         /* Voice Filter Parameters Init */

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

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

  860.                     synth.samplerate, synth.buffersize);

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

  862.                     synth.samplerate, synth.buffersize);

  863.         }

  864. 

  865.         if(param.PFilterEnvelopeEnabled)

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

  867. 

  868.         if(param.PFilterLfoEnabled)

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

  870. 

  871.         vce.FilterFreqTracking =

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

  873. 

  874.         /* Voice Modulation Parameters Init */

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

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

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

  878. 

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

  880. 

  881.             int vc = nvoice;

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

  883.                 vc = param.PextFMoscil;

  884. 

  885.             float tmp = 1.0f;

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

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

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

  889.                 tmp = getFMvoicebasefreq(nvoice);

  890. 

  891.             if(!pars.GlobalPar.Hrandgrouping)

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

  893. 

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

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

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

  897.                                              vce.FMSmp, tmp))

  898.                                         % synth.oscilsize;

  899. 

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

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

  902.             int oscposhiFM_add =

  903.                 (int)((param.PFMoscilphase

  904.                        - 64.0f) / 128.0f * synth.oscilsize

  905.                       + synth.oscilsize * 4);

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

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

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

  909.             }

  910.         }

  911. 

  912.         if(param.PFMFreqEnvelopeEnabled)

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

  914. 

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

  916. 

  917.         if(param.PFMAmpEnvelopeEnabled ) {

  918.             vce.FMAmpEnvelope =

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

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

  921.         }

  922.     }

  923. 

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

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

  926.             tmp[i] = 0;

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

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

  929.                 NoteVoicePar[nvoice].VoiceOut =

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

  931.                 tmp[i] = 1;

  932.             }

  933. 

  934.         if(NoteVoicePar[nvoice].VoiceOut)

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

  936.     }

  937. }

  938. 

  939. 

  940. /*

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

  942.  * This must be called before setfreq* functions

  943.  */

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

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

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

  947.         return;

  948.     }

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

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

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

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

  953.         pos += step;

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

  955.             pos  = -1.0f;

  956.             step = -step;

  957.         }

  958.         if(pos >= 1.0f) {

  959.             pos  = 1.0f;

  960.             step = -step;

  961.         }

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

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

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

  965.                                          - 1.0f) + vibratto_val

  966.                                         * unison_vibratto[nvoice].amplitude)

  967.                                      * relbw;

  968. 

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

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

  971.     }

  972. }

  973. 

  974. 

  975. /*

  976.  * Computes the frequency of an oscillator

  977.  */

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

  979. {

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

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

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

  983.         if(speed > synth.oscilsize_f)

  984.             speed = synth.oscilsize_f;

  985. 

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

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

  988.     }

  989. }

  990. 

  991. /*

  992.  * Computes the frequency of an modullator oscillator

  993.  */

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

  995. {

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

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

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

  999.         if(speed > synth.samplerate_f)

  1000.             speed = synth.samplerate_f;

  1001. 

  1002.         F2I(speed, oscfreqhiFM[nvoice][k]);

  1003.         oscfreqloFM[nvoice][k] = speed - floor(speed);

  1004.     }

  1005. }

  1006. 

  1007. /*

  1008.  * Get Voice base frequency

  1009.  */

  1010. float ADnote::getvoicebasefreq(int nvoice) const

  1011. {

  1012.     float detune = NoteVoicePar[nvoice].Detune / 100.0f

  1013.                    + NoteVoicePar[nvoice].FineDetune / 100.0f

  1014.                    * ctl.bandwidth.relbw * bandwidthDetuneMultiplier

  1015.                    + NoteGlobalPar.Detune / 100.0f;

  1016. 

  1017.     if(NoteVoicePar[nvoice].fixedfreq == 0)

  1018.         return this->basefreq * powf(2, detune / 12.0f);

  1019.     else { //the fixed freq is enabled

  1020.         float fixedfreq   = 440.0f;

  1021.         int   fixedfreqET = NoteVoicePar[nvoice].fixedfreqET;

  1022.         if(fixedfreqET != 0) { //if the frequency varies according the keyboard note

  1023.             float tmp =

  1024.                 (midinote

  1025.                  - 69.0f) / 12.0f

  1026.                 * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);

  1027.             if(fixedfreqET <= 64)

  1028.                 fixedfreq *= powf(2.0f, tmp);

  1029.             else

  1030.                 fixedfreq *= powf(3.0f, tmp);

  1031.         }

  1032.         return fixedfreq * powf(2.0f, detune / 12.0f);

  1033.     }

  1034. }

  1035. 

  1036. /*

  1037.  * Get Voice's Modullator base frequency

  1038.  */

  1039. float ADnote::getFMvoicebasefreq(int nvoice) const

  1040. {

  1041.     float detune = NoteVoicePar[nvoice].FMDetune / 100.0f;

  1042.     return getvoicebasefreq(nvoice) * powf(2, detune / 12.0f);

  1043. }

  1044. 

  1045. /*

  1046.  * Computes all the parameters for each tick

  1047.  */

  1048. void ADnote::computecurrentparameters()

  1049. {

  1050.     int   nvoice;

  1051.     float voicefreq, voicepitch, filterpitch, filterfreq, FMfreq,

  1052.           FMrelativepitch, globalpitch, globalfilterpitch;

  1053.     globalpitch = 0.01f * (NoteGlobalPar.FreqEnvelope->envout()

  1054.                            + NoteGlobalPar.FreqLfo->lfoout()

  1055.                            * ctl.modwheel.relmod);

  1056.     globaloldamplitude = globalnewamplitude;

  1057.     globalnewamplitude = NoteGlobalPar.Volume

  1058.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

  1059.                          * NoteGlobalPar.AmpLfo->amplfoout();

  1060. 

  1061.     globalfilterpitch = NoteGlobalPar.FilterEnvelope->envout()

  1062.                         + NoteGlobalPar.FilterLfo->lfoout()

  1063.                         + NoteGlobalPar.FilterCenterPitch;

  1064. 

  1065.     float tmpfilterfreq = globalfilterpitch + ctl.filtercutoff.relfreq

  1066.                           + NoteGlobalPar.FilterFreqTracking;

  1067. 

  1068.     tmpfilterfreq = Filter::getrealfreq(tmpfilterfreq);

  1069. 

  1070.     float globalfilterq = NoteGlobalPar.FilterQ * ctl.filterq.relq;

  1071.     NoteGlobalPar.GlobalFilterL->setfreq_and_q(tmpfilterfreq, globalfilterq);

  1072.     if(stereo != 0)

  1073.         NoteGlobalPar.GlobalFilterR->setfreq_and_q(tmpfilterfreq, globalfilterq);

  1074. 

  1075.     //compute the portamento, if it is used by this note

  1076.     float portamentofreqrap = 1.0f;

  1077.     if(portamento != 0) { //this voice use portamento

  1078.         portamentofreqrap = ctl.portamento.freqrap;

  1079.         if(ctl.portamento.used == 0) //the portamento has finished

  1080.             portamento = 0;  //this note is no longer "portamented"

  1081.     }

  1082. 

  1083.     //compute parameters for all voices

  1084.     for(nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1085.         if(NoteVoicePar[nvoice].Enabled != ON)

  1086.             continue;

  1087.         NoteVoicePar[nvoice].DelayTicks -= 1;

  1088.         if(NoteVoicePar[nvoice].DelayTicks > 0)

  1089.             continue;

  1090. 

  1091.         compute_unison_freq_rap(nvoice);

  1092. 

  1093.         /*******************/

  1094.         /* Voice Amplitude */

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

  1096.         oldamplitude[nvoice] = newamplitude[nvoice];

  1097.         newamplitude[nvoice] = 1.0f;

  1098. 

  1099.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1100.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpEnvelope->envout_dB();

  1101. 

  1102.         if(NoteVoicePar[nvoice].AmpLfo)

  1103.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpLfo->amplfoout();

  1104. 

  1105.         /****************/

  1106.         /* Voice Filter */

  1107.         /****************/

  1108.         if(NoteVoicePar[nvoice].VoiceFilterL) {

  1109.             filterpitch = NoteVoicePar[nvoice].FilterCenterPitch;

  1110. 

  1111.             if(NoteVoicePar[nvoice].FilterEnvelope)

  1112.                 filterpitch += NoteVoicePar[nvoice].FilterEnvelope->envout();

  1113. 

  1114.             if(NoteVoicePar[nvoice].FilterLfo)

  1115.                 filterpitch += NoteVoicePar[nvoice].FilterLfo->lfoout();

  1116. 

  1117.             filterfreq = filterpitch + NoteVoicePar[nvoice].FilterFreqTracking;

  1118.             filterfreq = Filter::getrealfreq(filterfreq);

  1119. 

  1120.             NoteVoicePar[nvoice].VoiceFilterL->setfreq(filterfreq);

  1121.             if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1122.                 NoteVoicePar[nvoice].VoiceFilterR->setfreq(filterfreq);

  1123.         }

  1124. 

  1125.         if(NoteVoicePar[nvoice].noisetype == 0) { //compute only if the voice isn't noise

  1126.             /*******************/

  1127.             /* Voice Frequency */

  1128.             /*******************/

  1129.             voicepitch = 0.0f;

  1130.             if(NoteVoicePar[nvoice].FreqLfo)

  1131.                 voicepitch += NoteVoicePar[nvoice].FreqLfo->lfoout() / 100.0f

  1132.                               * ctl.bandwidth.relbw;

  1133. 

  1134.             if(NoteVoicePar[nvoice].FreqEnvelope)

  1135.                 voicepitch += NoteVoicePar[nvoice].FreqEnvelope->envout()

  1136.                               / 100.0f;

  1137.             voicefreq = getvoicebasefreq(nvoice)

  1138.                         * powf(2, (voicepitch + globalpitch) / 12.0f);                //Hz frequency

  1139.             voicefreq *=

  1140.                 powf(ctl.pitchwheel.relfreq, NoteVoicePar[nvoice].BendAdjust); //change the frequency by the controller

  1141.             setfreq(nvoice, voicefreq * portamentofreqrap + NoteVoicePar[nvoice].OffsetHz);

  1142. 

  1143.             /***************/

  1144.             /*  Modulator */

  1145.             /***************/

  1146. 

  1147. 

  1148.             if(NoteVoicePar[nvoice].FMEnabled != NONE) {

  1149.                 FMrelativepitch = NoteVoicePar[nvoice].FMDetune / 100.0f;

  1150.                 if(NoteVoicePar[nvoice].FMFreqEnvelope)

  1151.                     FMrelativepitch +=

  1152.                         NoteVoicePar[nvoice].FMFreqEnvelope->envout() / 100;

  1153.                 if (NoteVoicePar[nvoice].FMFreqFixed)

  1154.                     FMfreq =

  1155.                         powf(2.0f, FMrelativepitch

  1156.                              / 12.0f) * 440.0f;

  1157.                 else

  1158.                     FMfreq =

  1159.                            powf(2.0f, FMrelativepitch

  1160.                                 / 12.0f) * voicefreq * portamentofreqrap;

  1161.                 setfreqFM(nvoice, FMfreq);

  1162. 

  1163.                 FMoldamplitude[nvoice] = FMnewamplitude[nvoice];

  1164.                 FMnewamplitude[nvoice] = NoteVoicePar[nvoice].FMVolume

  1165.                                          * ctl.fmamp.relamp;

  1166.                 if(NoteVoicePar[nvoice].FMAmpEnvelope)

  1167.                     FMnewamplitude[nvoice] *=

  1168.                         NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();

  1169.             }

  1170.         }

  1171.     }

  1172. }

  1173. 

  1174. 

  1175. /*

  1176.  * Fadein in a way that removes clicks but keep sound "punchy"

  1177.  */

  1178. inline void ADnote::fadein(float *smps) const

  1179. {

  1180.     int zerocrossings = 0;

  1181.     for(int i = 1; i < synth.buffersize; ++i)

  1182.         if((smps[i - 1] < 0.0f) && (smps[i] > 0.0f))

  1183.             zerocrossings++;  //this is only the possitive crossings

  1184. 

  1185.     float tmp = (synth.buffersize_f - 1.0f) / (zerocrossings + 1) / 3.0f;

  1186.     if(tmp < 8.0f)

  1187.         tmp = 8.0f;

  1188.     tmp *= NoteGlobalPar.Fadein_adjustment;

  1189. 

  1190.     int n;

  1191.     F2I(tmp, n); //how many samples is the fade-in

  1192.     if(n > synth.buffersize)

  1193.         n = synth.buffersize;

  1194.     for(int i = 0; i < n; ++i) { //fade-in

  1195.         float tmp = 0.5f - cosf((float)i / (float) n * PI) * 0.5f;

  1196.         smps[i] *= tmp;

  1197.     }

  1198. }

  1199. 

  1200. /*

  1201.  * Computes the Oscillator (Without Modulation) - LinearInterpolation

  1202.  */

  1203. 

  1204. /* As the code here is a bit odd due to optimization, here is what happens

  1205.  * First the current possition and frequency are retrieved from the running

  1206.  * state. These are broken up into high and low portions to indicate how many

  1207.  * samples are skipped in one step and how many fractional samples are skipped.

  1208.  * Outside of this method the fractional samples are just handled with floating

  1209.  * point code, but that's a bit slower than it needs to be. In this code the low

  1210.  * portions are known to exist between 0.0 and 1.0 and it is known that they are

  1211.  * stored in single precision floating point IEEE numbers. This implies that

  1212.  * a maximum of 24 bits are significant. The below code does your standard

  1213.  * linear interpolation that you'll see throughout this codebase, but by

  1214.  * sticking to integers for tracking the overflow of the low portion, around 15%

  1215.  * of the execution time was shaved off in the ADnote test.

  1216.  */

  1217. inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)

  1218. {

  1219.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1220.         int    poshi  = oscposhi[nvoice][k];

  1221.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1222.         int    freqhi = oscfreqhi[nvoice][k];

  1223.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1224.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1225.         float *tw     = tmpwave_unison[k];

  1226.         assert(oscfreqlo[nvoice][k] < 1.0f);

  1227.         for(int i = 0; i < synth.buffersize; ++i) {

  1228.             tw[i]  = (smps[poshi] * ((1<<24) - poslo) + smps[poshi + 1] * poslo)/(1.0f*(1<<24));

  1229.             poslo += freqlo;

  1230.             poshi += freqhi + (poslo>>24);

  1231.             poslo &= 0xffffff;

  1232.             poshi &= synth.oscilsize - 1;

  1233.         }

  1234.         oscposhi[nvoice][k] = poshi;

  1235.         oscposlo[nvoice][k] = poslo/(1.0f*(1<<24));

  1236.     }

  1237. }

  1238. 

  1239. 

  1240. 

  1241. /*

  1242.  * Computes the Oscillator (Without Modulation) - CubicInterpolation

  1243.  *

  1244.  The differences from the Linear are to little to deserve to be used. This is because I am using a large synth.oscilsize (>512)

  1245. inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){

  1246.     int i,poshi;

  1247.     float poslo;

  1248. 

  1249.     poshi=oscposhi[nvoice];

  1250.     poslo=oscposlo[nvoice];

  1251.     float *smps=NoteVoicePar[nvoice].OscilSmp;

  1252.     float xm1,x0,x1,x2,a,b,c;

  1253.     for (i=0;i<synth.buffersize;i++){

  1254.     xm1=smps[poshi];

  1255.     x0=smps[poshi+1];

  1256.     x1=smps[poshi+2];

  1257.     x2=smps[poshi+3];

  1258.     a=(3.0f * (x0-x1) - xm1 + x2) / 2.0f;

  1259.     b = 2.0f*x1 + xm1 - (5.0f*x0 + x2) / 2.0f;

  1260.     c = (x1 - xm1) / 2.0f;

  1261.     tmpwave[i]=(((a * poslo) + b) * poslo + c) * poslo + x0;

  1262.     printf("a\n");

  1263.     //tmpwave[i]=smps[poshi]*(1.0f-poslo)+smps[poshi+1]*poslo;

  1264.     poslo+=oscfreqlo[nvoice];

  1265.     if (poslo>=1.0f) {

  1266.             poslo-=1.0f;

  1267.         poshi++;

  1268.     };

  1269.         poshi+=oscfreqhi[nvoice];

  1270.         poshi&=synth.oscilsize-1;

  1271.     };

  1272.     oscposhi[nvoice]=poshi;

  1273.     oscposlo[nvoice]=poslo;

  1274. };

  1275. */

  1276. /*

  1277.  * Computes the Oscillator (Morphing)

  1278.  */

  1279. inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)

  1280. {

  1281.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1282.     if(FMnewamplitude[nvoice] > 1.0f)

  1283.         FMnewamplitude[nvoice] = 1.0f;

  1284.     if(FMoldamplitude[nvoice] > 1.0f)

  1285.         FMoldamplitude[nvoice] = 1.0f;

  1286. 

  1287.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1288.         //if I use VoiceOut[] as modullator

  1289.         int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1290.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1291.             float *tw = tmpwave_unison[k];

  1292.             for(int i = 0; i < synth.buffersize; ++i) {

  1293.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1294.                                             FMnewamplitude[nvoice],

  1295.                                             i,

  1296.                                             synth.buffersize);

  1297.                 tw[i] = tw[i]

  1298.                     * (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1299.             }

  1300.         }

  1301.     }

  1302.     else

  1303.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1304.             int    poshiFM  = oscposhiFM[nvoice][k];

  1305.             float  posloFM  = oscposloFM[nvoice][k];

  1306.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1307.             float  freqloFM = oscfreqloFM[nvoice][k];

  1308.             float *tw = tmpwave_unison[k];

  1309. 

  1310.             for(int i = 0; i < synth.buffersize; ++i) {

  1311.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1312.                                             FMnewamplitude[nvoice],

  1313.                                             i,

  1314.                                             synth.buffersize);

  1315.                 tw[i] = tw[i] * (1.0f - amp) + amp

  1316.                         * (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1 - posloFM)

  1317.                            + NoteVoicePar[nvoice].FMSmp[poshiFM + 1] * posloFM);

  1318.                 posloFM += freqloFM;

  1319.                 if(posloFM >= 1.0f) {

  1320.                     posloFM -= 1.0f;

  1321.                     poshiFM++;

  1322.                 }

  1323.                 poshiFM += freqhiFM;

  1324.                 poshiFM &= synth.oscilsize - 1;

  1325.             }

  1326.             oscposhiFM[nvoice][k] = poshiFM;

  1327.             oscposloFM[nvoice][k] = posloFM;

  1328.         }

  1329. }

  1330. 

  1331. /*

  1332.  * Computes the Oscillator (Ring Modulation)

  1333.  */

  1334. inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)

  1335. {

  1336.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1337.     if(FMnewamplitude[nvoice] > 1.0f)

  1338.         FMnewamplitude[nvoice] = 1.0f;

  1339.     if(FMoldamplitude[nvoice] > 1.0f)

  1340.         FMoldamplitude[nvoice] = 1.0f;

  1341.     if(NoteVoicePar[nvoice].FMVoice >= 0)

  1342.         // if I use VoiceOut[] as modullator

  1343.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1344.             float *tw = tmpwave_unison[k];

  1345.             for(int i = 0; i < synth.buffersize; ++i) {

  1346.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1347.                                             FMnewamplitude[nvoice],

  1348.                                             i,

  1349.                                             synth.buffersize);

  1350.                 int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1351.                 tw[i] *= (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1352.             }

  1353.         }

  1354.     else

  1355.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1356.             int    poshiFM  = oscposhiFM[nvoice][k];

  1357.             float  posloFM  = oscposloFM[nvoice][k];

  1358.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1359.             float  freqloFM = oscfreqloFM[nvoice][k];

  1360.             float *tw = tmpwave_unison[k];

  1361. 

  1362.             for(int i = 0; i < synth.buffersize; ++i) {

  1363.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1364.                                             FMnewamplitude[nvoice],

  1365.                                             i,

  1366.                                             synth.buffersize);

  1367.                 tw[i] *= (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1.0f - posloFM)

  1368.                           + NoteVoicePar[nvoice].FMSmp[poshiFM

  1369.                                                        + 1] * posloFM) * amp

  1370.                          + (1.0f - amp);

  1371.                 posloFM += freqloFM;

  1372.                 if(posloFM >= 1.0f) {

  1373.                     posloFM -= 1.0f;

  1374.                     poshiFM++;

  1375.                 }

  1376.                 poshiFM += freqhiFM;

  1377.                 poshiFM &= synth.oscilsize - 1;

  1378.             }

  1379.             oscposhiFM[nvoice][k] = poshiFM;

  1380.             oscposloFM[nvoice][k] = posloFM;

  1381.         }

  1382. }

  1383. 

  1384. /*

  1385.  * Computes the Oscillator (Phase Modulation or Frequency Modulation)

  1386.  */

  1387. inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,

  1388.                                                               int FMmode)

  1389. {

  1390.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1391.         //if I use VoiceOut[] as modulator

  1392.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1393.             float *tw = tmpwave_unison[k];

  1394.             const float *smps = NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut;

  1395.             if (FMmode == PW_MOD && (k & 1))

  1396.                 for (int i = 0; i < synth.buffersize; ++i)

  1397.                     tw[i] = -smps[i];

  1398.             else

  1399.                 memcpy(tw, smps, synth.bufferbytes);

  1400.         }

  1401.     } else {

  1402.         //Compute the modulator and store it in tmpwave_unison[][]

  1403.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1404.             int    poshiFM  = oscposhiFM[nvoice][k];

  1405.             int    posloFM  = oscposloFM[nvoice][k]  * (1<<24);

  1406.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1407.             int    freqloFM = oscfreqloFM[nvoice][k] * (1<<24);

  1408.             float *tw = tmpwave_unison[k];

  1409.             const float *smps = NoteVoicePar[nvoice].FMSmp;

  1410. 

  1411.             for(int i = 0; i < synth.buffersize; ++i) {

  1412.                 tw[i] = (smps[poshiFM] * ((1<<24) - posloFM)

  1413.                          + smps[poshiFM + 1] * posloFM) / (1.0f*(1<<24));

  1414.                 if (FMmode == PW_MOD && (k & 1))

  1415.                     tw[i] = -tw[i];

  1416. 

  1417.                 posloFM += freqloFM;

  1418.                 if(posloFM >= (1<<24)) {

  1419.                     posloFM &= 0xffffff;//fmod(posloFM, 1.0f);

  1420.                     poshiFM++;

  1421.                 }

  1422.                 poshiFM += freqhiFM;

  1423.                 poshiFM &= synth.oscilsize - 1;

  1424.             }

  1425.             oscposhiFM[nvoice][k] = poshiFM;

  1426.             oscposloFM[nvoice][k] = posloFM/((1<<24)*1.0f);

  1427.         }

  1428.     }

  1429.     // Amplitude interpolation

  1430.     if(ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],

  1431.                                  FMnewamplitude[nvoice])) {

  1432.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1433.             float *tw = tmpwave_unison[k];

  1434.             for(int i = 0; i < synth.buffersize; ++i)

  1435.                 tw[i] *= INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1436.                                                FMnewamplitude[nvoice],

  1437.                                                i,

  1438.                                                synth.buffersize);

  1439.         }

  1440.     } else {

  1441.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1442.             float *tw = tmpwave_unison[k];

  1443.             for(int i = 0; i < synth.buffersize; ++i)

  1444.                 tw[i] *= FMnewamplitude[nvoice];

  1445.         }

  1446.     }

  1447. 

  1448. 

  1449.     //normalize: makes all sample-rates, oscil_sizes to produce same sound

  1450.     if(FMmode == FREQ_MOD) { //Frequency modulation

  1451.         const float normalize = synth.oscilsize_f / 262144.0f * 44100.0f

  1452.                           / synth.samplerate_f;

  1453.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1454.             float *tw    = tmpwave_unison[k];

  1455.             float  fmold = FMoldsmp[nvoice][k];

  1456.             for(int i = 0; i < synth.buffersize; ++i) {

  1457.                 fmold = fmod(fmold + tw[i] * normalize, synth.oscilsize);

  1458.                 tw[i] = fmold;

  1459.             }

  1460.             FMoldsmp[nvoice][k] = fmold;

  1461.         }

  1462.     }

  1463.     else {  //Phase or PWM modulation

  1464.         const float normalize = synth.oscilsize_f / 262144.0f;

  1465.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1466.             float *tw = tmpwave_unison[k];

  1467.             for(int i = 0; i < synth.buffersize; ++i)

  1468.                 tw[i] *= normalize;

  1469.         }

  1470.     }

  1471. 

  1472.     //do the modulation

  1473.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1474.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1475.         float *tw     = tmpwave_unison[k];

  1476.         int    poshi  = oscposhi[nvoice][k];

  1477.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1478.         int    freqhi = oscfreqhi[nvoice][k];

  1479.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1480. 

  1481.         for(int i = 0; i < synth.buffersize; ++i) {

  1482.             int FMmodfreqhi = 0;

  1483.             F2I(tw[i], FMmodfreqhi);

  1484.             float FMmodfreqlo = tw[i]-FMmodfreqhi;//fmod(tw[i] /*+ 0.0000000001f*/, 1.0f);

  1485.             if(FMmodfreqhi < 0)

  1486.                 FMmodfreqlo++;

  1487. 

  1488.             //carrier

  1489.             int carposhi = poshi + FMmodfreqhi;

  1490.             int carposlo = poslo + FMmodfreqlo;

  1491.             if (FMmode == PW_MOD && (k & 1))

  1492.                 carposhi += NoteVoicePar[nvoice].phase_offset;

  1493. 

  1494.             if(carposlo >= (1<<24)) {

  1495.                 carposhi++;

  1496.                 carposlo &= 0xffffff;//fmod(carposlo, 1.0f);

  1497.             }

  1498.             carposhi &= (synth.oscilsize - 1);

  1499. 

  1500.             tw[i] = (smps[carposhi] * ((1<<24) - carposlo)

  1501.                     + smps[carposhi + 1] * carposlo)/(1.0f*(1<<24));

  1502. 

  1503.             poslo += freqlo;

  1504.             if(poslo >= (1<<24)) {

  1505.                 poslo &= 0xffffff;//fmod(poslo, 1.0f);

  1506.                 poshi++;

  1507.             }

  1508. 

  1509.             poshi += freqhi;

  1510.             poshi &= synth.oscilsize - 1;

  1511.         }

  1512.         oscposhi[nvoice][k] = poshi;

  1513.         oscposlo[nvoice][k] = (poslo)/((1<<24)*1.0f);

  1514.     }

  1515. }

  1516. 

  1517. 

  1518. /*

  1519.  * Computes the Noise

  1520.  */

  1521. inline void ADnote::ComputeVoiceWhiteNoise(int nvoice)

  1522. {

  1523.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1524.         float *tw = tmpwave_unison[k];

  1525.         for(int i = 0; i < synth.buffersize; ++i)

  1526.             tw[i] = RND * 2.0f - 1.0f;

  1527.     }

  1528. }

  1529. 

  1530. inline void ADnote::ComputeVoicePinkNoise(int nvoice)

  1531. {

  1532.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1533.         float *tw = tmpwave_unison[k];

  1534.         float *f = &pinking[nvoice][k > 0 ? 7 : 0];

  1535.         for(int i = 0; i < synth.buffersize; ++i) {

  1536. 	    float white = (RND-0.5)/4.0;

  1537. 	    f[0] = 0.99886*f[0]+white*0.0555179;

  1538. 	    f[1] = 0.99332*f[1]+white*0.0750759;

  1539. 	    f[2] = 0.96900*f[2]+white*0.1538520;

  1540. 	    f[3] = 0.86650*f[3]+white*0.3104856;

  1541. 	    f[4] = 0.55000*f[4]+white*0.5329522;

  1542. 	    f[5] = -0.7616*f[5]-white*0.0168980;

  1543. 	    tw[i] = f[0]+f[1]+f[2]+f[3]+f[4]+f[5]+f[6]+white*0.5362;

  1544. 	    f[6] = white*0.115926;

  1545.         }

  1546.     }

  1547. }

  1548. 

  1549. 

  1550. 

  1551. /*

  1552.  * Compute the ADnote samples

  1553.  * Returns 0 if the note is finished

  1554.  */

  1555. int ADnote::noteout(float *outl, float *outr)

  1556. {

  1557.     memcpy(outl, synth.denormalkillbuf, synth.bufferbytes);

  1558.     memcpy(outr, synth.denormalkillbuf, synth.bufferbytes);

  1559. 

  1560.     if(NoteEnabled == OFF)

  1561.         return 0;

  1562. 

  1563.     memset(bypassl, 0, synth.bufferbytes);

  1564.     memset(bypassr, 0, synth.bufferbytes);

  1565.     computecurrentparameters();

  1566. 

  1567.     for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1568.         if((NoteVoicePar[nvoice].Enabled != ON)

  1569.            || (NoteVoicePar[nvoice].DelayTicks > 0))

  1570.             continue;

  1571.         switch (NoteVoicePar[nvoice].noisetype) {

  1572.             case 0: //voice mode=sound

  1573.                 switch(NoteVoicePar[nvoice].FMEnabled) {

  1574.                     case MORPH:

  1575.                         ComputeVoiceOscillatorMorph(nvoice);

  1576.                         break;

  1577.                     case RING_MOD:

  1578.                         ComputeVoiceOscillatorRingModulation(nvoice);

  1579.                         break;

  1580.                     case FREQ_MOD:

  1581.                     case PHASE_MOD:

  1582.                     case PW_MOD:

  1583.                         ComputeVoiceOscillatorFrequencyModulation(nvoice,

  1584.                                                                   NoteVoicePar[nvoice].FMEnabled);

  1585.                         break;

  1586.                     default:

  1587.                         ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1588.                         //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice);

  1589.                 }

  1590.                 break;

  1591.             case 1:

  1592.                 ComputeVoiceWhiteNoise(nvoice);

  1593.                 break;

  1594.             default:

  1595.                 ComputeVoicePinkNoise(nvoice);

  1596.                 break;

  1597.         }

  1598.         // Voice Processing

  1599. 

  1600. 

  1601.         //mix subvoices into voice

  1602.         memset(tmpwavel, 0, synth.bufferbytes);

  1603.         if(stereo)

  1604.             memset(tmpwaver, 0, synth.bufferbytes);

  1605.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1606.             float *tw = tmpwave_unison[k];

  1607.             if(stereo) {

  1608.                 float stereo_pos = 0;

  1609.                 bool is_pwm = NoteVoicePar[nvoice].FMEnabled == PW_MOD;

  1610.                 if (is_pwm) {

  1611.                     if(unison_size[nvoice] > 2)

  1612.                         stereo_pos = k/2

  1613.                             / (float)(unison_size[nvoice]/2

  1614.                                       - 1) * 2.0f - 1.0f;

  1615.                 } else if(unison_size[nvoice] > 1) {

  1616.                     stereo_pos = k

  1617.                         / (float)(unison_size[nvoice]

  1618.                                   - 1) * 2.0f - 1.0f;

  1619.                 }

  1620.                 float stereo_spread = unison_stereo_spread[nvoice] * 2.0f; //between 0 and 2.0f

  1621.                 if(stereo_spread > 1.0f) {

  1622.                     float stereo_pos_1 = (stereo_pos >= 0.0f) ? 1.0f : -1.0f;

  1623.                     stereo_pos =

  1624.                         (2.0f

  1625.                          - stereo_spread) * stereo_pos

  1626.                         + (stereo_spread - 1.0f) * stereo_pos_1;

  1627.                 }

  1628.                 else

  1629.                     stereo_pos *= stereo_spread;

  1630. 

  1631.                 if(unison_size[nvoice] == 1 ||

  1632.                    (is_pwm && unison_size[nvoice] == 2))

  1633.                     stereo_pos = 0.0f;

  1634.                 float panning = (stereo_pos + 1.0f) * 0.5f;

  1635. 

  1636. 

  1637.                 float lvol = (1.0f - panning) * 2.0f;

  1638.                 if(lvol > 1.0f)

  1639.                     lvol = 1.0f;

  1640. 

  1641.                 float rvol = panning * 2.0f;

  1642.                 if(rvol > 1.0f)

  1643.                     rvol = 1.0f;

  1644. 

  1645.                 if(unison_invert_phase[nvoice][k]) {

  1646.                     lvol = -lvol;

  1647.                     rvol = -rvol;

  1648.                 }

  1649. 

  1650.                 for(int i = 0; i < synth.buffersize; ++i)

  1651.                     tmpwavel[i] += tw[i] * lvol;

  1652.                 for(int i = 0; i < synth.buffersize; ++i)

  1653.                     tmpwaver[i] += tw[i] * rvol;

  1654.             }

  1655.             else

  1656.                 for(int i = 0; i < synth.buffersize; ++i)

  1657.                     tmpwavel[i] += tw[i];

  1658.         }

  1659. 

  1660. 

  1661.         float unison_amplitude = 1.0f / sqrt(unison_size[nvoice]); //reduce the amplitude for large unison sizes

  1662.         // Amplitude

  1663.         float oldam = oldamplitude[nvoice] * unison_amplitude;

  1664.         float newam = newamplitude[nvoice] * unison_amplitude;

  1665. 

  1666.         if(ABOVE_AMPLITUDE_THRESHOLD(oldam, newam)) {

  1667.             int rest = synth.buffersize;

  1668.             //test if the amplitude if raising and the difference is high

  1669.             if((newam > oldam) && ((newam - oldam) > 0.25f)) {

  1670.                 rest = 10;

  1671.                 if(rest > synth.buffersize)

  1672.                     rest = synth.buffersize;

  1673.                 for(int i = 0; i < synth.buffersize - rest; ++i)

  1674.                     tmpwavel[i] *= oldam;

  1675.                 if(stereo)

  1676.                     for(int i = 0; i < synth.buffersize - rest; ++i)

  1677.                         tmpwaver[i] *= oldam;

  1678.             }

  1679.             // Amplitude interpolation

  1680.             for(int i = 0; i < rest; ++i) {

  1681.                 float amp = INTERPOLATE_AMPLITUDE(oldam, newam, i, rest);

  1682.                 tmpwavel[i + (synth.buffersize - rest)] *= amp;

  1683.                 if(stereo)

  1684.                     tmpwaver[i + (synth.buffersize - rest)] *= amp;

  1685.             }

  1686.         }

  1687.         else {

  1688.             for(int i = 0; i < synth.buffersize; ++i)

  1689.                 tmpwavel[i] *= newam;

  1690.             if(stereo)

  1691.                 for(int i = 0; i < synth.buffersize; ++i)

  1692.                     tmpwaver[i] *= newam;

  1693.         }

  1694. 

  1695.         // Fade in

  1696.         if(firsttick[nvoice] != 0) {

  1697.             fadein(&tmpwavel[0]);

  1698.             if(stereo)

  1699.                 fadein(&tmpwaver[0]);

  1700.             firsttick[nvoice] = 0;

  1701.         }

  1702. 

  1703. 

  1704.         // Filter

  1705.         if(NoteVoicePar[nvoice].VoiceFilterL)

  1706.             NoteVoicePar[nvoice].VoiceFilterL->filterout(&tmpwavel[0]);

  1707.         if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1708.             NoteVoicePar[nvoice].VoiceFilterR->filterout(&tmpwaver[0]);

  1709. 

  1710.         //check if the amplitude envelope is finished, if yes, the voice will be fadeout

  1711.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1712.             if(NoteVoicePar[nvoice].AmpEnvelope->finished()) {

  1713.                 for(int i = 0; i < synth.buffersize; ++i)

  1714.                     tmpwavel[i] *= 1.0f - (float)i / synth.buffersize_f;

  1715.                 if(stereo)

  1716.                     for(int i = 0; i < synth.buffersize; ++i)

  1717.                         tmpwaver[i] *= 1.0f - (float)i / synth.buffersize_f;

  1718.             }

  1719.         //the voice is killed later

  1720. 

  1721. 

  1722.         // Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator)

  1723.         if(NoteVoicePar[nvoice].VoiceOut) {

  1724.             if(stereo)

  1725.                 for(int i = 0; i < synth.buffersize; ++i)

  1726.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i]

  1727.                                                        + tmpwaver[i];

  1728.             else   //mono

  1729.                 for(int i = 0; i < synth.buffersize; ++i)

  1730.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i];

  1731.         }

  1732. 

  1733. 

  1734.         // Add the voice that do not bypass the filter to out

  1735.         if(NoteVoicePar[nvoice].filterbypass == 0) { //no bypass

  1736.             if(stereo)

  1737.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1738.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1739.                                * NoteVoicePar[nvoice].Panning * 2.0f;

  1740.                     outr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1741.                                * (1.0f - NoteVoicePar[nvoice].Panning) * 2.0f;

  1742.                 }

  1743.             else

  1744.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1745.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1746.         }

  1747.         else {  //bypass the filter

  1748.             if(stereo)

  1749.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1750.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1751.                                   * NoteVoicePar[nvoice].Panning * 2.0f;

  1752.                     bypassr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1753.                                   * (1.0f

  1754.                                      - NoteVoicePar[nvoice].Panning) * 2.0f;

  1755.                 }

  1756.             else

  1757.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1758.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1759.         }

  1760.         // chech if there is necesary to proces the voice longer (if the Amplitude envelope isn't finished)

  1761.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1762.             if(NoteVoicePar[nvoice].AmpEnvelope->finished())

  1763.                 KillVoice(nvoice);

  1764.     }

  1765. 

  1766. 

  1767.     //Processing Global parameters

  1768.     NoteGlobalPar.GlobalFilterL->filterout(&outl[0]);

  1769. 

  1770.     if(stereo == 0) { //set the right channel=left channel

  1771.         memcpy(outr, outl, synth.bufferbytes);

  1772.         memcpy(bypassr, bypassl, synth.bufferbytes);

  1773.     }

  1774.     else

  1775.         NoteGlobalPar.GlobalFilterR->filterout(&outr[0]);

  1776. 

  1777.     for(int i = 0; i < synth.buffersize; ++i) {

  1778.         outl[i] += bypassl[i];

  1779.         outr[i] += bypassr[i];

  1780.     }

  1781. 

  1782.     if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))

  1783.         // Amplitude Interpolation

  1784.         for(int i = 0; i < synth.buffersize; ++i) {

  1785.             float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,

  1786.                                                  globalnewamplitude,

  1787.                                                  i,

  1788.                                                  synth.buffersize);

  1789.             outl[i] *= tmpvol * NoteGlobalPar.Panning;

  1790.             outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);

  1791.         }

  1792.     else

  1793.         for(int i = 0; i < synth.buffersize; ++i) {

  1794.             outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;

  1795.             outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);

  1796.         }

  1797. 

  1798.     //Apply the punch

  1799.     if(NoteGlobalPar.Punch.Enabled != 0)

  1800.         for(int i = 0; i < synth.buffersize; ++i) {

  1801.             float punchamp = NoteGlobalPar.Punch.initialvalue

  1802.                              * NoteGlobalPar.Punch.t + 1.0f;

  1803.             outl[i] *= punchamp;

  1804.             outr[i] *= punchamp;

  1805.             NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;

  1806.             if(NoteGlobalPar.Punch.t < 0.0f) {

  1807.                 NoteGlobalPar.Punch.Enabled = 0;

  1808.                 break;

  1809.             }

  1810.         }

  1811. 

  1812. 

  1813.     // Apply legato-specific sound signal modifications

  1814.     legato.apply(*this, outl, outr);

  1815. 

  1816. 

  1817.     // Check if the global amplitude is finished.

  1818.     // If it does, disable the note

  1819.     if(NoteGlobalPar.AmpEnvelope->finished()) {

  1820.         for(int i = 0; i < synth.buffersize; ++i) { //fade-out

  1821.             float tmp = 1.0f - (float)i / synth.buffersize_f;

  1822.             outl[i] *= tmp;

  1823.             outr[i] *= tmp;

  1824.         }

  1825.         KillNote();

  1826.     }

  1827.     return 1;

  1828. }

  1829. 

  1830. 

  1831. /*

  1832.  * Release the key (NoteOff)

  1833.  */

  1834. void ADnote::releasekey()

  1835. {

  1836.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)

  1837.         NoteVoicePar[nvoice].releasekey();

  1838.     NoteGlobalPar.FreqEnvelope->releasekey();

  1839.     NoteGlobalPar.FilterEnvelope->releasekey();

  1840.     NoteGlobalPar.AmpEnvelope->releasekey();

  1841. }

  1842. 

  1843. /*

  1844.  * Check if the note is finished

  1845.  */

  1846. int ADnote::finished() const

  1847. {

  1848.     if(NoteEnabled == ON)

  1849.         return 0;

  1850.     else

  1851.         return 1;

  1852. }

  1853. 

  1854. void ADnote::Voice::releasekey()

  1855. {

  1856.     if(!Enabled)

  1857.         return;

  1858.     if(AmpEnvelope)

  1859.         AmpEnvelope->releasekey();

  1860.     if(FreqEnvelope)

  1861.         FreqEnvelope->releasekey();

  1862.     if(FilterEnvelope)

  1863.         FilterEnvelope->releasekey();

  1864.     if(FMFreqEnvelope)

  1865.         FMFreqEnvelope->releasekey();

  1866.     if(FMAmpEnvelope)

  1867.         FMAmpEnvelope->releasekey();

  1868. }

  1869. 

  1870. void ADnote::Voice::kill(Allocator &memory, const SYNTH_T &synth)

  1871. {

  1872.     memory.devalloc(OscilSmp);

  1873.     memory.dealloc(FreqEnvelope);

  1874.     memory.dealloc(FreqLfo);

  1875.     memory.dealloc(AmpEnvelope);

  1876.     memory.dealloc(AmpLfo);

  1877.     memory.dealloc(VoiceFilterL);

  1878.     memory.dealloc(VoiceFilterR);

  1879.     memory.dealloc(FilterEnvelope);

  1880.     memory.dealloc(FilterLfo);

  1881.     memory.dealloc(FMFreqEnvelope);

  1882.     memory.dealloc(FMAmpEnvelope);

  1883. 

  1884.     if((FMEnabled != NONE) && (FMVoice < 0))

  1885.         memory.devalloc(FMSmp);

  1886. 

  1887.     if(VoiceOut)

  1888.         memset(VoiceOut, 0, synth.bufferbytes);

  1889.     //the buffer can't be safely deleted as it may be

  1890.     //an input to another voice

  1891. 

  1892.     Enabled = OFF;

  1893. }

  1894. 

  1895. void ADnote::Global::kill(Allocator &memory)

  1896. {

  1897.     memory.dealloc(FreqEnvelope);

  1898.     memory.dealloc(FreqLfo);

  1899.     memory.dealloc(AmpEnvelope);

  1900.     memory.dealloc(AmpLfo);

  1901.     memory.dealloc(GlobalFilterL);

  1902.     memory.dealloc(GlobalFilterR);

  1903.     memory.dealloc(FilterEnvelope);

  1904.     memory.dealloc(FilterLfo);

  1905. }

  1906. 

  1907. void ADnote::Global::initparameters(const ADnoteGlobalParam &param,

  1908.                                     const SYNTH_T &synth,

  1909.                                     const AbsTime &time,

  1910.                                     class Allocator &memory,

  1911.                                     float basefreq, float velocity,

  1912.                                     bool stereo)

  1913. {

  1914.     FreqEnvelope = memory.alloc<Envelope>(*param.FreqEnvelope, basefreq, synth.dt());

  1915.     FreqLfo      = memory.alloc<LFO>(*param.FreqLfo, basefreq, time);

  1916. 

  1917.     AmpEnvelope = memory.alloc<Envelope>(*param.AmpEnvelope, basefreq, synth.dt());

  1918.     AmpLfo      = memory.alloc<LFO>(*param.AmpLfo, basefreq, time);

  1919. 

  1920.     Volume = 4.0f * powf(0.1f, 3.0f * (1.0f - param.PVolume / 96.0f)) //-60 dB .. 0 dB

  1921.              * VelF(velocity, param.PAmpVelocityScaleFunction);     //sensing

  1922. 

  1923.     GlobalFilterL = Filter::generate(memory, param.GlobalFilter,

  1924.             synth.samplerate, synth.buffersize);

  1925.     if(stereo)

  1926.         GlobalFilterR = Filter::generate(memory, param.GlobalFilter,

  1927.                 synth.samplerate, synth.buffersize);

  1928.     else

  1929.         GlobalFilterR = NULL;

  1930. 

  1931.     FilterEnvelope = memory.alloc<Envelope>(*param.FilterEnvelope, basefreq, synth.dt());

  1932.     FilterLfo      = memory.alloc<LFO>(*param.FilterLfo, basefreq, time);

  1933.     FilterQ = param.GlobalFilter->getq();

  1934.     FilterFreqTracking = param.GlobalFilter->getfreqtracking(basefreq);

  1935. }