falkTX
/
Carla
mirror of https://github.com/falkTX/Carla
1
0
Fork 0
Code Releases 42 Activity
Audio plugin host https://kx.studio/carla
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
3007 Commits
16 Branches
275MB
Tree: 8deb54f77b
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from '8deb54f77b'
${ noResults }
Carla/source/native-plugins/zynaddsubfx/Synth/ADnote.cpp

1801 lines
67KB
Raw Blame History 

  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.     time   = 0.0f;

  53.     stereo = pars.GlobalPar.PStereo;

  54. 

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

  56.                                      pars.GlobalPar.PCoarseDetune,

  57.                                      pars.GlobalPar.PDetune);

  58.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  59. 

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

  61.         NoteGlobalPar.Panning = RND;

  62.     else

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

  64. 

  65. 

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

  67.                                       + pars.GlobalPar.PFilterVelocityScale

  68.                                       / 127.0f * 6.0f       //velocity sensing

  69.                                       * (VelF(velocity,

  70.                                               pars.GlobalPar.

  71.                                               PFilterVelocityScaleFunction) - 1);

  72. 

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

  74.         NoteGlobalPar.Punch.Enabled = 1;

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

  76.         NoteGlobalPar.Punch.initialvalue =

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

  78.              * VelF(velocity,

  79.                     pars.GlobalPar.PPunchVelocitySensing));

  80.         float time =

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

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

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

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

  85.     }

  86.     else

  87.         NoteGlobalPar.Punch.Enabled = 0;

  88. 

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

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

  91.         NoteVoicePar[nvoice].OscilSmp = NULL;

  92.         NoteVoicePar[nvoice].FMSmp    = NULL;

  93.         NoteVoicePar[nvoice].VoiceOut = NULL;

  94. 

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

  96.         unison_size[nvoice] = 1;

  97. 

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

  99.             NoteVoicePar[nvoice].Enabled = OFF;

  100.             continue; //the voice is disabled

  101.         }

  102. 

  103.         unison_stereo_spread[nvoice] =

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

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

  106.         if(unison < 1)

  107.             unison = 1;

  108. 

  109.         //compute unison

  110.         unison_size[nvoice] = unison;

  111. 

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

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

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

  115.         float unison_spread =

  116.             pars.getUnisonFrequencySpreadCents(nvoice);

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

  118.         float unison_vibratto_a  =

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

  120. 

  121. 

  122.         switch(unison) {

  123.             case 1:

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

  125.                 break;

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

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

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

  129.             };

  130.                 break;

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

  132.                 float unison_values[unison];

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

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

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

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

  137.                     unison_values[k] = val;

  138.                     if (min > val) {

  139.                         min = val;

  140.                     }

  141.                     if (max < val) {

  142.                         max = val;

  143.                     }

  144.                 }

  145.                 float diff = max - min;

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

  147.                     unison_values[k] =

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

  149.                     unison_base_freq_rap[nvoice][k] =

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

  151.                 }

  152.             };

  153.         }

  154. 

  155.         //unison vibrattos

  156.         if(unison > 1)

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

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

  159.                                                   + (unison_base_freq_rap[

  160.                                                          nvoice][k] - 1.0f)

  161.                                                   * (1.0f - unison_vibratto_a);

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

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

  164.         unison_vibratto[nvoice].amplitude =

  165.             (unison_real_spread - 1.0f) * unison_vibratto_a;

  166. 

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

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

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

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

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

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

  173.             float vibratto_period = vibratto_base_period

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

  175. 

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

  177.             if(RND < 0.5f)

  178.                 m = -m;

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

  180.         }

  181. 

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

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

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

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

  186.         }

  187. 

  188.         //phase invert for unison

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

  190.         if(unison != 1) {

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

  192.             switch(inv) {

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

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

  195.                     break;

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

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

  198.                     break;

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

  200.                         unison_invert_phase[nvoice][k] =

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

  202.                     break;

  203.             }

  204.         }

  205. 

  206. 

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

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

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

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

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

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

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

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

  215. 

  216.         NoteVoicePar[nvoice].Enabled     = ON;

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

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

  219. 

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

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

  222.             NoteVoicePar[nvoice].Detune = getdetune(

  223.                 pars.VoicePar[nvoice].PDetuneType,

  224.                 pars.VoicePar[nvoice].

  225.                 PCoarseDetune,

  226.                 8192); //coarse detune

  227.             NoteVoicePar[nvoice].FineDetune = getdetune(

  228.                 pars.VoicePar[nvoice].PDetuneType,

  229.                 0,

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

  231.         }

  232.         else {

  233.             NoteVoicePar[nvoice].Detune = getdetune(

  234.                 pars.GlobalPar.PDetuneType,

  235.                 pars.VoicePar[nvoice].

  236.                 PCoarseDetune,

  237.                 8192); //coarse detune

  238.             NoteVoicePar[nvoice].FineDetune = getdetune(

  239.                 pars.GlobalPar.PDetuneType,

  240.                 0,

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

  242.         }

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

  244.             NoteVoicePar[nvoice].FMDetune = getdetune(

  245.                 pars.VoicePar[nvoice].PFMDetuneType,

  246.                 pars.VoicePar[nvoice].

  247.                 PFMCoarseDetune,

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

  249.         else

  250.             NoteVoicePar[nvoice].FMDetune = getdetune(

  251.                 pars.GlobalPar.PDetuneType,

  252.                 pars.VoicePar[nvoice].

  253.                 PFMCoarseDetune,

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

  255. 

  256. 

  257. 

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

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

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

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

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

  263.         }

  264. 

  265.         //the extra points contains the first point

  266.         NoteVoicePar[nvoice].OscilSmp =

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

  268. 

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

  270.         int vc = nvoice;

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

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

  273.         if(!pars.GlobalPar.Hrandgrouping)

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

  275.         int oscposhi_start =

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

  277.                                              getvoicebasefreq(nvoice),

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

  279. 

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

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

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

  283.                                           + i] =

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

  285. 

  286.         oscposhi_start +=

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

  288.                    - 64.0f) / 128.0f * synth.oscilsize + synth.oscilsize * 4);

  289.         oscposhi_start %= synth.oscilsize;

  290. 

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

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

  293.             //put random starting point for other subvoices

  294.             oscposhi_start      =

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

  296.                         127.0f * (synth.oscilsize - 1));

  297.         }

  298. 

  299.         NoteVoicePar[nvoice].FreqLfo      = NULL;

  300.         NoteVoicePar[nvoice].FreqEnvelope = NULL;

  301. 

  302.         NoteVoicePar[nvoice].AmpLfo      = NULL;

  303.         NoteVoicePar[nvoice].AmpEnvelope = NULL;

  304. 

  305.         NoteVoicePar[nvoice].VoiceFilterL   = NULL;

  306.         NoteVoicePar[nvoice].VoiceFilterR   = NULL;

  307.         NoteVoicePar[nvoice].FilterEnvelope = NULL;

  308.         NoteVoicePar[nvoice].FilterLfo      = NULL;

  309. 

  310.         NoteVoicePar[nvoice].FilterCenterPitch =

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

  312.         NoteVoicePar[nvoice].filterbypass =

  313.             pars.VoicePar[nvoice].Pfilterbypass;

  314. 

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

  316.             case 1:

  317.                 NoteVoicePar[nvoice].FMEnabled = MORPH;

  318.                 break;

  319.             case 2:

  320.                 NoteVoicePar[nvoice].FMEnabled = RING_MOD;

  321.                 break;

  322.             case 3:

  323.                 NoteVoicePar[nvoice].FMEnabled = PHASE_MOD;

  324.                 break;

  325.             case 4:

  326.                 NoteVoicePar[nvoice].FMEnabled = FREQ_MOD;

  327.                 break;

  328.             case 5:

  329.                 NoteVoicePar[nvoice].FMEnabled = PITCH_MOD;

  330.                 break;

  331.             default:

  332.                 NoteVoicePar[nvoice].FMEnabled = NONE;

  333.         }

  334. 

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

  336.         NoteVoicePar[nvoice].FMFreqEnvelope = NULL;

  337.         NoteVoicePar[nvoice].FMAmpEnvelope  = NULL;

  338. 

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

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

  341.                                    nvoice),

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

  343.                                - 1.0f);

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

  345.             case PHASE_MOD:

  346.                 fmvoldamp =

  347.                     powf(440.0f / getvoicebasefreq(

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

  349.                          / 64.0f);

  350.                 NoteVoicePar[nvoice].FMVolume =

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

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

  353.                 break;

  354.             case FREQ_MOD:

  355.                 NoteVoicePar[nvoice].FMVolume =

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

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

  358.                 break;

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

  360.             //	          break;

  361.             default:

  362.                 if(fmvoldamp > 1.0f)

  363.                     fmvoldamp = 1.0f;

  364.                 NoteVoicePar[nvoice].FMVolume =

  365.                     pars.VoicePar[nvoice].PFMVolume

  366.                     / 127.0f * fmvoldamp;

  367.         }

  368. 

  369.         //Voice's modulator velocity sensing

  370.         NoteVoicePar[nvoice].FMVolume *=

  371.             VelF(velocity,

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

  373. 

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

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

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

  377. 

  378.         firsttick[nvoice] = 1;

  379.         NoteVoicePar[nvoice].DelayTicks =

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

  381.                         * logf(50.0f))

  382.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  383.     }

  384. 

  385.     max_unison = 1;

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

  387.         if(unison_size[nvoice] > max_unison)

  388.             max_unison = unison_size[nvoice];

  389. 

  390. 

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

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

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

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

  395.     }

  396. 

  397.     initparameters();

  398. }

  399. 

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

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

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

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

  404. void ADnote::legatonote(LegatoParams lpars)

  405. {

  406.     //ADnoteParameters &pars = *partparams;

  407. 

  408.     // Manage legato stuff

  409.     if(legato.update(lpars))

  410.         return;

  411. 

  412.     portamento = lpars.portamento;

  413.     midinote   = lpars.midinote;

  414.     basefreq   = lpars.frequency;

  415. 

  416.     if(velocity > 1.0f)

  417.         velocity = 1.0f;

  418.     velocity = lpars.velocity;

  419. 

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

  421.                                      pars.GlobalPar.PCoarseDetune,

  422.                                      pars.GlobalPar.PDetune);

  423.     bandwidthDetuneMultiplier = pars.getBandwidthDetuneMultiplier();

  424. 

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

  426.         NoteGlobalPar.Panning = RND;

  427.     else

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

  429. 

  430.     //center freq

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

  432.                                       + pars.GlobalPar.PFilterVelocityScale

  433.                                       / 127.0f * 6.0f          //velocity sensing

  434.                                       * (VelF(velocity,

  435.                                               pars.GlobalPar.

  436.                                               PFilterVelocityScaleFunction) - 1);

  437. 

  438. 

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

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

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

  442. 

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

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

  445. 

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

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

  448.             NoteVoicePar[nvoice].Detune = getdetune(

  449.                 pars.VoicePar[nvoice].PDetuneType,

  450.                 pars.VoicePar[nvoice].PCoarseDetune,

  451.                 8192); //coarse detune

  452.             NoteVoicePar[nvoice].FineDetune = getdetune(

  453.                 pars.VoicePar[nvoice].PDetuneType,

  454.                 0,

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

  456.         }

  457.         else {

  458.             NoteVoicePar[nvoice].Detune = getdetune(

  459.                 pars.GlobalPar.PDetuneType,

  460.                 pars.VoicePar[nvoice].PCoarseDetune,

  461.                 8192); //coarse detune

  462.             NoteVoicePar[nvoice].FineDetune = getdetune(

  463.                 pars.GlobalPar.PDetuneType,

  464.                 0,

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

  466.         }

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

  468.             NoteVoicePar[nvoice].FMDetune = getdetune(

  469.                 pars.VoicePar[nvoice].PFMDetuneType,

  470.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  472.         else

  473.             NoteVoicePar[nvoice].FMDetune = getdetune(

  474.                 pars.GlobalPar.PDetuneType,

  475.                 pars.VoicePar[nvoice].PFMCoarseDetune,

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

  477. 

  478. 

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

  480.         int vc = nvoice;

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

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

  483.         if(!pars.GlobalPar.Hrandgrouping)

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

  485. 

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

  487.                                          getvoicebasefreq(nvoice),

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

  489. 

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

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

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

  493.                                           + i] =

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

  495. 

  496. 

  497.         NoteVoicePar[nvoice].FilterCenterPitch =

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

  499.         NoteVoicePar[nvoice].filterbypass =

  500.             pars.VoicePar[nvoice].Pfilterbypass;

  501. 

  502. 

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

  504. 

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

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

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

  508.                                - 1.0f);

  509. 

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

  511.             case PHASE_MOD:

  512.                 fmvoldamp =

  513.                     powf(440.0f / getvoicebasefreq(

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

  515.                          / 64.0f);

  516.                 NoteVoicePar[nvoice].FMVolume =

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

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

  519.                 break;

  520.             case FREQ_MOD:

  521.                 NoteVoicePar[nvoice].FMVolume =

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

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

  524.                 break;

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

  526.             //	          break;

  527.             default:

  528.                 if(fmvoldamp > 1.0f)

  529.                     fmvoldamp = 1.0f;

  530.                 NoteVoicePar[nvoice].FMVolume =

  531.                     pars.VoicePar[nvoice].PFMVolume

  532.                     / 127.0f * fmvoldamp;

  533.         }

  534. 

  535.         //Voice's modulator velocity sensing

  536.         NoteVoicePar[nvoice].FMVolume *=

  537.             VelF(velocity,

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

  539. 

  540.         NoteVoicePar[nvoice].DelayTicks =

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

  542.                         * logf(50.0f))

  543.                    - 1.0f) / synth.buffersize_f / 10.0f * synth.samplerate_f);

  544.     }

  545. 

  546.     ///    initparameters();

  547. 

  548.     ///////////////

  549.     // Altered content of initparameters():

  550. 

  551.     int tmp[NUM_VOICES];

  552. 

  553.     NoteGlobalPar.Volume = 4.0f

  554.                            * powf(0.1f, 3.0f

  555.                                   * (1.0f - pars.GlobalPar.PVolume

  556.                                      / 96.0f))                                      //-60 dB .. 0 dB

  557.                            * VelF(

  558.         velocity,

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

  560. 

  561.     globalnewamplitude = NoteGlobalPar.Volume

  562.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  564. 

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

  566.     NoteGlobalPar.FilterFreqTracking =

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

  568. 

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

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

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

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

  573. 

  574.     // Voice Parameter init

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

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

  577.             continue;

  578. 

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

  580.         /* Voice Amplitude Parameters Init */

  581.         NoteVoicePar[nvoice].Volume =

  582.             powf(0.1f, 3.0f

  583.                  * (1.0f - pars.VoicePar[nvoice].PVolume / 127.0f))             // -60 dB .. 0 dB

  584.             * VelF(velocity,

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

  586. 

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

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

  589. 

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

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

  592.         else

  593.             NoteVoicePar[nvoice].Panning =

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

  595. 

  596.         newamplitude[nvoice] = 1.0f;

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

  598.            && NoteVoicePar[nvoice].AmpEnvelope)

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

  600. 

  601. 

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

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

  604. 

  605. 

  606. 

  607.         NoteVoicePar[nvoice].FilterFreqTracking =

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

  609. 

  610.         /* Voice Modulation Parameters Init */

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

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

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

  614. 

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

  616. 

  617.             int vc = nvoice;

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

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

  620. 

  621.             if(!pars.GlobalPar.Hrandgrouping)

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

  623. 

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

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

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

  627.         }

  628. 

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

  630.                                  * ctl.fmamp.relamp;

  631. 

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

  633.            && NoteVoicePar[nvoice].FMAmpEnvelope)

  634.             FMnewamplitude[nvoice] *=

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

  636.     }

  637. 

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

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

  640.             tmp[i] = 0;

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

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

  643.                 tmp[i] = 1;

  644.     }

  645. }

  646. 

  647. 

  648. /*

  649.  * Kill a voice of ADnote

  650.  */

  651. void ADnote::KillVoice(int nvoice)

  652. {

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

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

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

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

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

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

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

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

  661. 

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

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

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

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

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

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

  668. 

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

  670. }

  671. 

  672. /*

  673.  * Kill the note

  674.  */

  675. void ADnote::KillNote()

  676. {

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

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

  679.             KillVoice(nvoice);

  680. 

  681.         if(NoteVoicePar[nvoice].VoiceOut)

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

  683.     }

  684. 

  685.     NoteGlobalPar.kill(memory);

  686. 

  687.     NoteEnabled = OFF;

  688. }

  689. 

  690. ADnote::~ADnote()

  691. {

  692.     if(NoteEnabled == ON)

  693.         KillNote();

  694.     memory.devalloc(tmpwavel);

  695.     memory.devalloc(tmpwaver);

  696.     memory.devalloc(bypassl);

  697.     memory.devalloc(bypassr);

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

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

  700.     memory.devalloc(tmpwave_unison);

  701. }

  702. 

  703. 

  704. /*

  705.  * Init the parameters

  706.  */

  707. void ADnote::initparameters()

  708. {

  709.     int tmp[NUM_VOICES];

  710.     //ADnoteParameters &pars = *partparams;

  711. 

  712.     // Global Parameters

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

  714.                                  memory, basefreq, velocity,

  715.                                  stereo);

  716. 

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

  718.     globalnewamplitude = NoteGlobalPar.Volume

  719.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  721. 

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

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

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

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

  726. 

  727.     // Voice Parameter init

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

  729.         Voice &vce = NoteVoicePar[nvoice];

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

  731. 

  732.         if(vce.Enabled == 0)

  733.             continue;

  734. 

  735.         vce.noisetype = param.Type;

  736.         /* Voice Amplitude Parameters Init */

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

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

  739. 

  740.         if(param.PVolumeminus)

  741.             vce.Volume = -vce.Volume;

  742. 

  743.         if(param.PPanning == 0)

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

  745.         else

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

  747. 

  748.         newamplitude[nvoice] = 1.0f;

  749.         if(param.PAmpEnvelopeEnabled) {

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

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

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

  753.         }

  754. 

  755.         if(param.PAmpLfoEnabled) {

  756.             vce.AmpLfo = memory.alloc<LFO>(*param.AmpLfo, basefreq, synth.dt());

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

  758.         }

  759. 

  760.         /* Voice Frequency Parameters Init */

  761.         if(param.PFreqEnvelopeEnabled)

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

  763. 

  764.         if(param.PFreqLfoEnabled)

  765.             vce.FreqLfo = memory.alloc<LFO>(*param.FreqLfo, basefreq, synth.dt());

  766. 

  767.         /* Voice Filter Parameters Init */

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

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

  770.                     synth.samplerate, synth.buffersize);

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

  772.                     synth.samplerate, synth.buffersize);

  773.         }

  774. 

  775.         if(param.PFilterEnvelopeEnabled)

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

  777. 

  778.         if(param.PFilterLfoEnabled)

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

  780. 

  781.         vce.FilterFreqTracking =

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

  783. 

  784.         /* Voice Modulation Parameters Init */

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

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

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

  788. 

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

  790. 

  791.             int vc = nvoice;

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

  793.                 vc = param.PextFMoscil;

  794. 

  795.             float tmp = 1.0f;

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

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

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

  799.                 tmp = getFMvoicebasefreq(nvoice);

  800. 

  801.             if(!pars.GlobalPar.Hrandgrouping)

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

  803. 

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

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

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

  807.                                              vce.FMSmp, tmp))

  808.                                         % synth.oscilsize;

  809. 

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

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

  812.             int oscposhiFM_add =

  813.                 (int)((param.PFMoscilphase

  814.                        - 64.0f) / 128.0f * synth.oscilsize

  815.                       + synth.oscilsize * 4);

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

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

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

  819.             }

  820.         }

  821. 

  822.         if(param.PFMFreqEnvelopeEnabled)

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

  824. 

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

  826. 

  827.         if(param.PFMAmpEnvelopeEnabled ) {

  828.             vce.FMAmpEnvelope =

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

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

  831.         }

  832.     }

  833. 

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

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

  836.             tmp[i] = 0;

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

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

  839.                 NoteVoicePar[nvoice].VoiceOut =

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

  841.                 tmp[i] = 1;

  842.             }

  843. 

  844.         if(NoteVoicePar[nvoice].VoiceOut)

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

  846.     }

  847. }

  848. 

  849. 

  850. /*

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

  852.  * This must be called before setfreq* functions

  853.  */

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

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

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

  857.         return;

  858.     }

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

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

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

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

  863.         pos += step;

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

  865.             pos  = -1.0f;

  866.             step = -step;

  867.         }

  868.         if(pos >= 1.0f) {

  869.             pos  = 1.0f;

  870.             step = -step;

  871.         }

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

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

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

  875.                                          - 1.0f) + vibratto_val

  876.                                         * unison_vibratto[nvoice].amplitude)

  877.                                      * relbw;

  878. 

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

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

  881.     }

  882. }

  883. 

  884. 

  885. /*

  886.  * Computes the frequency of an oscillator

  887.  */

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

  889. {

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

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

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

  893.         if(speed > synth.oscilsize_f)

  894.             speed = synth.oscilsize_f;

  895. 

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

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

  898.     }

  899. }

  900. 

  901. /*

  902.  * Computes the frequency of an modullator oscillator

  903.  */

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

  905. {

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

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

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

  909.         if(speed > synth.samplerate_f)

  910.             speed = synth.samplerate_f;

  911. 

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

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

  914.     }

  915. }

  916. 

  917. /*

  918.  * Get Voice base frequency

  919.  */

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

  921. {

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

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

  924.                    * ctl.bandwidth.relbw * bandwidthDetuneMultiplier

  925.                    + NoteGlobalPar.Detune / 100.0f;

  926. 

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

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

  929.     else { //the fixed freq is enabled

  930.         float fixedfreq   = 440.0f;

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

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

  933.             float tmp =

  934.                 (midinote

  935.                  - 69.0f) / 12.0f

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

  937.             if(fixedfreqET <= 64)

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

  939.             else

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

  941.         }

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

  943.     }

  944. }

  945. 

  946. /*

  947.  * Get Voice's Modullator base frequency

  948.  */

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

  950. {

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

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

  953. }

  954. 

  955. /*

  956.  * Computes all the parameters for each tick

  957.  */

  958. void ADnote::computecurrentparameters()

  959. {

  960.     int   nvoice;

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

  962.           FMrelativepitch, globalpitch, globalfilterpitch;

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

  964.                            + NoteGlobalPar.FreqLfo->lfoout()

  965.                            * ctl.modwheel.relmod);

  966.     globaloldamplitude = globalnewamplitude;

  967.     globalnewamplitude = NoteGlobalPar.Volume

  968.                          * NoteGlobalPar.AmpEnvelope->envout_dB()

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

  970. 

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

  972.                         + NoteGlobalPar.FilterLfo->lfoout()

  973.                         + NoteGlobalPar.FilterCenterPitch;

  974. 

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

  976.                           + NoteGlobalPar.FilterFreqTracking;

  977. 

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

  979. 

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

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

  982.     if(stereo != 0)

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

  984. 

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

  986.     float portamentofreqrap = 1.0f;

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

  988.         portamentofreqrap = ctl.portamento.freqrap;

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

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

  991.     }

  992. 

  993.     //compute parameters for all voices

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

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

  996.             continue;

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

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

  999.             continue;

  1000. 

  1001.         compute_unison_freq_rap(nvoice);

  1002. 

  1003.         /*******************/

  1004.         /* Voice Amplitude */

  1005.         /*******************/

  1006.         oldamplitude[nvoice] = newamplitude[nvoice];

  1007.         newamplitude[nvoice] = 1.0f;

  1008. 

  1009.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1010.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpEnvelope->envout_dB();

  1011. 

  1012.         if(NoteVoicePar[nvoice].AmpLfo)

  1013.             newamplitude[nvoice] *= NoteVoicePar[nvoice].AmpLfo->amplfoout();

  1014. 

  1015.         /****************/

  1016.         /* Voice Filter */

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

  1018.         if(NoteVoicePar[nvoice].VoiceFilterL) {

  1019.             filterpitch = NoteVoicePar[nvoice].FilterCenterPitch;

  1020. 

  1021.             if(NoteVoicePar[nvoice].FilterEnvelope)

  1022.                 filterpitch += NoteVoicePar[nvoice].FilterEnvelope->envout();

  1023. 

  1024.             if(NoteVoicePar[nvoice].FilterLfo)

  1025.                 filterpitch += NoteVoicePar[nvoice].FilterLfo->lfoout();

  1026. 

  1027.             filterfreq = filterpitch + NoteVoicePar[nvoice].FilterFreqTracking;

  1028.             filterfreq = Filter::getrealfreq(filterfreq);

  1029. 

  1030.             NoteVoicePar[nvoice].VoiceFilterL->setfreq(filterfreq);

  1031.             if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1032.                 NoteVoicePar[nvoice].VoiceFilterR->setfreq(filterfreq);

  1033.         }

  1034. 

  1035.         if(NoteVoicePar[nvoice].noisetype == 0) { //compute only if the voice isn't noise

  1036.             /*******************/

  1037.             /* Voice Frequency */

  1038.             /*******************/

  1039.             voicepitch = 0.0f;

  1040.             if(NoteVoicePar[nvoice].FreqLfo)

  1041.                 voicepitch += NoteVoicePar[nvoice].FreqLfo->lfoout() / 100.0f

  1042.                               * ctl.bandwidth.relbw;

  1043. 

  1044.             if(NoteVoicePar[nvoice].FreqEnvelope)

  1045.                 voicepitch += NoteVoicePar[nvoice].FreqEnvelope->envout()

  1046.                               / 100.0f;

  1047.             voicefreq = getvoicebasefreq(nvoice)

  1048.                         * powf(2, (voicepitch + globalpitch) / 12.0f);                //Hz frequency

  1049.             voicefreq *= ctl.pitchwheel.relfreq; //change the frequency by the controller

  1050.             setfreq(nvoice, voicefreq * portamentofreqrap);

  1051. 

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

  1053.             /*  Modulator */

  1054.             /***************/

  1055.             if(NoteVoicePar[nvoice].FMEnabled != NONE) {

  1056.                 FMrelativepitch = NoteVoicePar[nvoice].FMDetune / 100.0f;

  1057.                 if(NoteVoicePar[nvoice].FMFreqEnvelope)

  1058.                     FMrelativepitch +=

  1059.                         NoteVoicePar[nvoice].FMFreqEnvelope->envout() / 100;

  1060.                 FMfreq =

  1061.                     powf(2.0f, FMrelativepitch

  1062.                          / 12.0f) * voicefreq * portamentofreqrap;

  1063.                 setfreqFM(nvoice, FMfreq);

  1064. 

  1065.                 FMoldamplitude[nvoice] = FMnewamplitude[nvoice];

  1066.                 FMnewamplitude[nvoice] = NoteVoicePar[nvoice].FMVolume

  1067.                                          * ctl.fmamp.relamp;

  1068.                 if(NoteVoicePar[nvoice].FMAmpEnvelope)

  1069.                     FMnewamplitude[nvoice] *=

  1070.                         NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();

  1071.             }

  1072.         }

  1073.     }

  1074.     time += synth.buffersize_f / synth.samplerate_f;

  1075. }

  1076. 

  1077. 

  1078. /*

  1079.  * Fadein in a way that removes clicks but keep sound "punchy"

  1080.  */

  1081. inline void ADnote::fadein(float *smps) const

  1082. {

  1083.     int zerocrossings = 0;

  1084.     for(int i = 1; i < synth.buffersize; ++i)

  1085.         if((smps[i - 1] < 0.0f) && (smps[i] > 0.0f))

  1086.             zerocrossings++;  //this is only the possitive crossings

  1087. 

  1088.     float tmp = (synth.buffersize_f - 1.0f) / (zerocrossings + 1) / 3.0f;

  1089.     if(tmp < 8.0f)

  1090.         tmp = 8.0f;

  1091. 

  1092.     int n;

  1093.     F2I(tmp, n); //how many samples is the fade-in

  1094.     if(n > synth.buffersize)

  1095.         n = synth.buffersize;

  1096.     for(int i = 0; i < n; ++i) { //fade-in

  1097.         float tmp = 0.5f - cosf((float)i / (float) n * PI) * 0.5f;

  1098.         smps[i] *= tmp;

  1099.     }

  1100. }

  1101. 

  1102. /*

  1103.  * Computes the Oscillator (Without Modulation) - LinearInterpolation

  1104.  */

  1105. 

  1106. /* As the code here is a bit odd due to optimization, here is what happens

  1107.  * First the current possition and frequency are retrieved from the running

  1108.  * state. These are broken up into high and low portions to indicate how many

  1109.  * samples are skipped in one step and how many fractional samples are skipped.

  1110.  * Outside of this method the fractional samples are just handled with floating

  1111.  * point code, but that's a bit slower than it needs to be. In this code the low

  1112.  * portions are known to exist between 0.0 and 1.0 and it is known that they are

  1113.  * stored in single precision floating point IEEE numbers. This implies that

  1114.  * a maximum of 24 bits are significant. The below code does your standard

  1115.  * linear interpolation that you'll see throughout this codebase, but by

  1116.  * sticking to integers for tracking the overflow of the low portion, around 15%

  1117.  * of the execution time was shaved off in the ADnote test.

  1118.  */

  1119. inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)

  1120. {

  1121.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1122.         int    poshi  = oscposhi[nvoice][k];

  1123.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1124.         int    freqhi = oscfreqhi[nvoice][k];

  1125.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1126.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1127.         float *tw     = tmpwave_unison[k];

  1128.         assert(oscfreqlo[nvoice][k] < 1.0f);

  1129.         for(int i = 0; i < synth.buffersize; ++i) {

  1130.             tw[i]  = (smps[poshi] * ((1<<24) - poslo) + smps[poshi + 1] * poslo)/(1.0f*(1<<24));

  1131.             poslo += freqlo;

  1132.             poshi += freqhi + (poslo>>24);

  1133.             poslo &= 0xffffff;

  1134.             poshi &= synth.oscilsize - 1;

  1135.         }

  1136.         oscposhi[nvoice][k] = poshi;

  1137.         oscposlo[nvoice][k] = poslo/(1.0f*(1<<24));

  1138.     }

  1139. }

  1140. 

  1141. 

  1142. 

  1143. /*

  1144.  * Computes the Oscillator (Without Modulation) - CubicInterpolation

  1145.  *

  1146.  The differences from the Linear are to little to deserve to be used. This is because I am using a large synth.oscilsize (>512)

  1147. inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){

  1148.     int i,poshi;

  1149.     float poslo;

  1150. 

  1151.     poshi=oscposhi[nvoice];

  1152.     poslo=oscposlo[nvoice];

  1153.     float *smps=NoteVoicePar[nvoice].OscilSmp;

  1154.     float xm1,x0,x1,x2,a,b,c;

  1155.     for (i=0;i<synth.buffersize;i++){

  1156.     xm1=smps[poshi];

  1157.     x0=smps[poshi+1];

  1158.     x1=smps[poshi+2];

  1159.     x2=smps[poshi+3];

  1160.     a=(3.0f * (x0-x1) - xm1 + x2) / 2.0f;

  1161.     b = 2.0f*x1 + xm1 - (5.0f*x0 + x2) / 2.0f;

  1162.     c = (x1 - xm1) / 2.0f;

  1163.     tmpwave[i]=(((a * poslo) + b) * poslo + c) * poslo + x0;

  1164.     printf("a\n");

  1165.     //tmpwave[i]=smps[poshi]*(1.0f-poslo)+smps[poshi+1]*poslo;

  1166.     poslo+=oscfreqlo[nvoice];

  1167.     if (poslo>=1.0f) {

  1168.             poslo-=1.0f;

  1169.         poshi++;

  1170.     };

  1171.         poshi+=oscfreqhi[nvoice];

  1172.         poshi&=synth.oscilsize-1;

  1173.     };

  1174.     oscposhi[nvoice]=poshi;

  1175.     oscposlo[nvoice]=poslo;

  1176. };

  1177. */

  1178. /*

  1179.  * Computes the Oscillator (Morphing)

  1180.  */

  1181. inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)

  1182. {

  1183.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1184.     if(FMnewamplitude[nvoice] > 1.0f)

  1185.         FMnewamplitude[nvoice] = 1.0f;

  1186.     if(FMoldamplitude[nvoice] > 1.0f)

  1187.         FMoldamplitude[nvoice] = 1.0f;

  1188. 

  1189.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1190.         //if I use VoiceOut[] as modullator

  1191.         int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1192.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1193.             float *tw = tmpwave_unison[k];

  1194.             for(int i = 0; i < synth.buffersize; ++i) {

  1195.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1196.                                             FMnewamplitude[nvoice],

  1197.                                             i,

  1198.                                             synth.buffersize);

  1199.                 tw[i] = tw[i]

  1200.                     * (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1201.             }

  1202.         }

  1203.     }

  1204.     else

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

  1206.             int    poshiFM  = oscposhiFM[nvoice][k];

  1207.             float  posloFM  = oscposloFM[nvoice][k];

  1208.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1209.             float  freqloFM = oscfreqloFM[nvoice][k];

  1210.             float *tw = tmpwave_unison[k];

  1211. 

  1212.             for(int i = 0; i < synth.buffersize; ++i) {

  1213.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1214.                                             FMnewamplitude[nvoice],

  1215.                                             i,

  1216.                                             synth.buffersize);

  1217.                 tw[i] = tw[i] * (1.0f - amp) + amp

  1218.                         * (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1 - posloFM)

  1219.                            + NoteVoicePar[nvoice].FMSmp[poshiFM + 1] * posloFM);

  1220.                 posloFM += freqloFM;

  1221.                 if(posloFM >= 1.0f) {

  1222.                     posloFM -= 1.0f;

  1223.                     poshiFM++;

  1224.                 }

  1225.                 poshiFM += freqhiFM;

  1226.                 poshiFM &= synth.oscilsize - 1;

  1227.             }

  1228.             oscposhiFM[nvoice][k] = poshiFM;

  1229.             oscposloFM[nvoice][k] = posloFM;

  1230.         }

  1231. }

  1232. 

  1233. /*

  1234.  * Computes the Oscillator (Ring Modulation)

  1235.  */

  1236. inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)

  1237. {

  1238.     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1239.     if(FMnewamplitude[nvoice] > 1.0f)

  1240.         FMnewamplitude[nvoice] = 1.0f;

  1241.     if(FMoldamplitude[nvoice] > 1.0f)

  1242.         FMoldamplitude[nvoice] = 1.0f;

  1243.     if(NoteVoicePar[nvoice].FMVoice >= 0)

  1244.         // if I use VoiceOut[] as modullator

  1245.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1246.             float *tw = tmpwave_unison[k];

  1247.             for(int i = 0; i < synth.buffersize; ++i) {

  1248.                 float amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1249.                                             FMnewamplitude[nvoice],

  1250.                                             i,

  1251.                                             synth.buffersize);

  1252.                 int FMVoice = NoteVoicePar[nvoice].FMVoice;

  1253.                 tw[i] *= (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];

  1254.             }

  1255.         }

  1256.     else

  1257.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1258.             int    poshiFM  = oscposhiFM[nvoice][k];

  1259.             float  posloFM  = oscposloFM[nvoice][k];

  1260.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1261.             float  freqloFM = oscfreqloFM[nvoice][k];

  1262.             float *tw = tmpwave_unison[k];

  1263. 

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

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

  1266.                                             FMnewamplitude[nvoice],

  1267.                                             i,

  1268.                                             synth.buffersize);

  1269.                 tw[i] *= (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1.0f - posloFM)

  1270.                           + NoteVoicePar[nvoice].FMSmp[poshiFM

  1271.                                                        + 1] * posloFM) * amp

  1272.                          + (1.0f - amp);

  1273.                 posloFM += freqloFM;

  1274.                 if(posloFM >= 1.0f) {

  1275.                     posloFM -= 1.0f;

  1276.                     poshiFM++;

  1277.                 }

  1278.                 poshiFM += freqhiFM;

  1279.                 poshiFM &= synth.oscilsize - 1;

  1280.             }

  1281.             oscposhiFM[nvoice][k] = poshiFM;

  1282.             oscposloFM[nvoice][k] = posloFM;

  1283.         }

  1284. }

  1285. 

  1286. /*

  1287.  * Computes the Oscillator (Phase Modulation or Frequency Modulation)

  1288.  */

  1289. inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,

  1290.                                                               int FMmode)

  1291. {

  1292.     if(NoteVoicePar[nvoice].FMVoice >= 0) {

  1293.         //if I use VoiceOut[] as modulator

  1294.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1295.             float *tw = tmpwave_unison[k];

  1296.             const float *smps = NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut;

  1297.             memcpy(tw, smps, synth.bufferbytes);

  1298.         }

  1299.     } else {

  1300.         //Compute the modulator and store it in tmpwave_unison[][]

  1301.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1302.             int    poshiFM  = oscposhiFM[nvoice][k];

  1303.             int    posloFM  = oscposloFM[nvoice][k]  * (1<<24);

  1304.             int    freqhiFM = oscfreqhiFM[nvoice][k];

  1305.             int    freqloFM = oscfreqloFM[nvoice][k] * (1<<24);

  1306.             float *tw = tmpwave_unison[k];

  1307.             const float *smps = NoteVoicePar[nvoice].FMSmp;

  1308. 

  1309.             for(int i = 0; i < synth.buffersize; ++i) {

  1310.                 tw[i] = (smps[poshiFM] * ((1<<24) - posloFM)

  1311.                      + smps[poshiFM + 1] * posloFM) / (1.0f*(1<<24));

  1312.                 posloFM += freqloFM;

  1313.                 if(posloFM >= (1<<24)) {

  1314.                     posloFM &= 0xffffff;//fmod(posloFM, 1.0f);

  1315.                     poshiFM++;

  1316.                 }

  1317.                 poshiFM += freqhiFM;

  1318.                 poshiFM &= synth.oscilsize - 1;

  1319.             }

  1320.             oscposhiFM[nvoice][k] = poshiFM;

  1321.             oscposloFM[nvoice][k] = posloFM/((1<<24)*1.0f);

  1322.         }

  1323.     }

  1324.     // Amplitude interpolation

  1325.     if(ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],

  1326.                                  FMnewamplitude[nvoice])) {

  1327.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1328.             float *tw = tmpwave_unison[k];

  1329.             for(int i = 0; i < synth.buffersize; ++i)

  1330.                 tw[i] *= INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],

  1331.                                                FMnewamplitude[nvoice],

  1332.                                                i,

  1333.                                                synth.buffersize);

  1334.         }

  1335.     } else {

  1336.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1337.             float *tw = tmpwave_unison[k];

  1338.             for(int i = 0; i < synth.buffersize; ++i)

  1339.                 tw[i] *= FMnewamplitude[nvoice];

  1340.         }

  1341.     }

  1342. 

  1343. 

  1344.     //normalize: makes all sample-rates, oscil_sizes to produce same sound

  1345.     if(FMmode != 0) { //Frequency modulation

  1346.         const float normalize = synth.oscilsize_f / 262144.0f * 44100.0f

  1347.                           / synth.samplerate_f;

  1348.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1349.             float *tw    = tmpwave_unison[k];

  1350.             float  fmold = FMoldsmp[nvoice][k];

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

  1352.                 fmold = fmod(fmold + tw[i] * normalize, synth.oscilsize);

  1353.                 tw[i] = fmold;

  1354.             }

  1355.             FMoldsmp[nvoice][k] = fmold;

  1356.         }

  1357.     }

  1358.     else {  //Phase modulation

  1359.         const float normalize = synth.oscilsize_f / 262144.0f;

  1360.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1361.             float *tw = tmpwave_unison[k];

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

  1363.                 tw[i] *= normalize;

  1364.         }

  1365.     }

  1366. 

  1367.     //do the modulation

  1368.     for(int k = 0; k < unison_size[nvoice]; ++k) {

  1369.         float *smps   = NoteVoicePar[nvoice].OscilSmp;

  1370.         float *tw     = tmpwave_unison[k];

  1371.         int    poshi  = oscposhi[nvoice][k];

  1372.         int    poslo  = oscposlo[nvoice][k] * (1<<24);

  1373.         int    freqhi = oscfreqhi[nvoice][k];

  1374.         int    freqlo = oscfreqlo[nvoice][k] * (1<<24);

  1375. 

  1376.         for(int i = 0; i < synth.buffersize; ++i) {

  1377.             int FMmodfreqhi = 0;

  1378.             F2I(tw[i], FMmodfreqhi);

  1379.             float FMmodfreqlo = tw[i]-FMmodfreqhi;//fmod(tw[i] /*+ 0.0000000001f*/, 1.0f);

  1380.             if(FMmodfreqhi < 0)

  1381.                 FMmodfreqlo++;

  1382. 

  1383.             //carrier

  1384.             int carposhi = poshi + FMmodfreqhi;

  1385.             int carposlo = poslo + FMmodfreqlo;

  1386. 

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

  1388.                 carposhi++;

  1389.                 carposlo &= 0xffffff;//fmod(carposlo, 1.0f);

  1390.             }

  1391.             carposhi &= (synth.oscilsize - 1);

  1392. 

  1393.             tw[i] = (smps[carposhi] * ((1<<24) - carposlo)

  1394.                     + smps[carposhi + 1] * carposlo)/(1.0f*(1<<24));

  1395. 

  1396.             poslo += freqlo;

  1397.             if(poslo >= (1<<24)) {

  1398.                 poslo &= 0xffffff;//fmod(poslo, 1.0f);

  1399.                 poshi++;

  1400.             }

  1401. 

  1402.             poshi += freqhi;

  1403.             poshi &= synth.oscilsize - 1;

  1404.         }

  1405.         oscposhi[nvoice][k] = poshi;

  1406.         oscposlo[nvoice][k] = (poslo)/((1<<24)*1.0f);

  1407.     }

  1408. }

  1409. 

  1410. 

  1411. /*Calculeaza Oscilatorul cu PITCH MODULATION*/

  1412. inline void ADnote::ComputeVoiceOscillatorPitchModulation(int /*nvoice*/)

  1413. {

  1414. //TODO

  1415. }

  1416. 

  1417. /*

  1418.  * Computes the Noise

  1419.  */

  1420. inline void ADnote::ComputeVoiceNoise(int nvoice)

  1421. {

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

  1423.         float *tw = tmpwave_unison[k];

  1424.         for(int i = 0; i < synth.buffersize; ++i)

  1425.             tw[i] = RND * 2.0f - 1.0f;

  1426.     }

  1427. }

  1428. 

  1429. 

  1430. 

  1431. /*

  1432.  * Compute the ADnote samples

  1433.  * Returns 0 if the note is finished

  1434.  */

  1435. int ADnote::noteout(float *outl, float *outr)

  1436. {

  1437.     memcpy(outl, denormalkillbuf, synth.bufferbytes);

  1438.     memcpy(outr, denormalkillbuf, synth.bufferbytes);

  1439. 

  1440.     if(NoteEnabled == OFF)

  1441.         return 0;

  1442. 

  1443.     memset(bypassl, 0, synth.bufferbytes);

  1444.     memset(bypassr, 0, synth.bufferbytes);

  1445.     computecurrentparameters();

  1446. 

  1447.     for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {

  1448.         if((NoteVoicePar[nvoice].Enabled != ON)

  1449.            || (NoteVoicePar[nvoice].DelayTicks > 0))

  1450.             continue;

  1451.         if(NoteVoicePar[nvoice].noisetype == 0) //voice mode=sound

  1452.             switch(NoteVoicePar[nvoice].FMEnabled) {

  1453.                 case MORPH:

  1454.                     ComputeVoiceOscillatorMorph(nvoice);

  1455.                     break;

  1456.                 case RING_MOD:

  1457.                     ComputeVoiceOscillatorRingModulation(nvoice);

  1458.                     break;

  1459.                 case PHASE_MOD:

  1460.                     ComputeVoiceOscillatorFrequencyModulation(nvoice, 0);

  1461.                     break;

  1462.                 case FREQ_MOD:

  1463.                     ComputeVoiceOscillatorFrequencyModulation(nvoice, 1);

  1464.                     break;

  1465.                 //case PITCH_MOD:ComputeVoiceOscillatorPitchModulation(nvoice);break;

  1466.                 default:

  1467.                     ComputeVoiceOscillator_LinearInterpolation(nvoice);

  1468.                     //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice);

  1469.             }

  1470.         else

  1471.             ComputeVoiceNoise(nvoice);

  1472.         // Voice Processing

  1473. 

  1474. 

  1475.         //mix subvoices into voice

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

  1477.         if(stereo)

  1478.             memset(tmpwaver, 0, synth.bufferbytes);

  1479.         for(int k = 0; k < unison_size[nvoice]; ++k) {

  1480.             float *tw = tmpwave_unison[k];

  1481.             if(stereo) {

  1482.                 float stereo_pos = 0;

  1483.                 if(unison_size[nvoice] > 1)

  1484.                     stereo_pos = k

  1485.                                  / (float)(unison_size[nvoice]

  1486.                                            - 1) * 2.0f - 1.0f;

  1487.                 float stereo_spread = unison_stereo_spread[nvoice] * 2.0f; //between 0 and 2.0f

  1488.                 if(stereo_spread > 1.0f) {

  1489.                     float stereo_pos_1 = (stereo_pos >= 0.0f) ? 1.0f : -1.0f;

  1490.                     stereo_pos =

  1491.                         (2.0f

  1492.                          - stereo_spread) * stereo_pos

  1493.                         + (stereo_spread - 1.0f) * stereo_pos_1;

  1494.                 }

  1495.                 else

  1496.                     stereo_pos *= stereo_spread;

  1497. 

  1498.                 if(unison_size[nvoice] == 1)

  1499.                     stereo_pos = 0.0f;

  1500.                 float panning = (stereo_pos + 1.0f) * 0.5f;

  1501. 

  1502. 

  1503.                 float lvol = (1.0f - panning) * 2.0f;

  1504.                 if(lvol > 1.0f)

  1505.                     lvol = 1.0f;

  1506. 

  1507.                 float rvol = panning * 2.0f;

  1508.                 if(rvol > 1.0f)

  1509.                     rvol = 1.0f;

  1510. 

  1511.                 if(unison_invert_phase[nvoice][k]) {

  1512.                     lvol = -lvol;

  1513.                     rvol = -rvol;

  1514.                 }

  1515. 

  1516.                 for(int i = 0; i < synth.buffersize; ++i)

  1517.                     tmpwavel[i] += tw[i] * lvol;

  1518.                 for(int i = 0; i < synth.buffersize; ++i)

  1519.                     tmpwaver[i] += tw[i] * rvol;

  1520.             }

  1521.             else

  1522.                 for(int i = 0; i < synth.buffersize; ++i)

  1523.                     tmpwavel[i] += tw[i];

  1524.         }

  1525. 

  1526. 

  1527.         float unison_amplitude = 1.0f / sqrt(unison_size[nvoice]); //reduce the amplitude for large unison sizes

  1528.         // Amplitude

  1529.         float oldam = oldamplitude[nvoice] * unison_amplitude;

  1530.         float newam = newamplitude[nvoice] * unison_amplitude;

  1531. 

  1532.         if(ABOVE_AMPLITUDE_THRESHOLD(oldam, newam)) {

  1533.             int rest = synth.buffersize;

  1534.             //test if the amplitude if raising and the difference is high

  1535.             if((newam > oldam) && ((newam - oldam) > 0.25f)) {

  1536.                 rest = 10;

  1537.                 if(rest > synth.buffersize)

  1538.                     rest = synth.buffersize;

  1539.                 for(int i = 0; i < synth.buffersize - rest; ++i)

  1540.                     tmpwavel[i] *= oldam;

  1541.                 if(stereo)

  1542.                     for(int i = 0; i < synth.buffersize - rest; ++i)

  1543.                         tmpwaver[i] *= oldam;

  1544.             }

  1545.             // Amplitude interpolation

  1546.             for(int i = 0; i < rest; ++i) {

  1547.                 float amp = INTERPOLATE_AMPLITUDE(oldam, newam, i, rest);

  1548.                 tmpwavel[i + (synth.buffersize - rest)] *= amp;

  1549.                 if(stereo)

  1550.                     tmpwaver[i + (synth.buffersize - rest)] *= amp;

  1551.             }

  1552.         }

  1553.         else {

  1554.             for(int i = 0; i < synth.buffersize; ++i)

  1555.                 tmpwavel[i] *= newam;

  1556.             if(stereo)

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

  1558.                     tmpwaver[i] *= newam;

  1559.         }

  1560. 

  1561.         // Fade in

  1562.         if(firsttick[nvoice] != 0) {

  1563.             fadein(&tmpwavel[0]);

  1564.             if(stereo)

  1565.                 fadein(&tmpwaver[0]);

  1566.             firsttick[nvoice] = 0;

  1567.         }

  1568. 

  1569. 

  1570.         // Filter

  1571.         if(NoteVoicePar[nvoice].VoiceFilterL)

  1572.             NoteVoicePar[nvoice].VoiceFilterL->filterout(&tmpwavel[0]);

  1573.         if(stereo && NoteVoicePar[nvoice].VoiceFilterR)

  1574.             NoteVoicePar[nvoice].VoiceFilterR->filterout(&tmpwaver[0]);

  1575. 

  1576.         //check if the amplitude envelope is finished, if yes, the voice will be fadeout

  1577.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1578.             if(NoteVoicePar[nvoice].AmpEnvelope->finished()) {

  1579.                 for(int i = 0; i < synth.buffersize; ++i)

  1580.                     tmpwavel[i] *= 1.0f - (float)i / synth.buffersize_f;

  1581.                 if(stereo)

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

  1583.                         tmpwaver[i] *= 1.0f - (float)i / synth.buffersize_f;

  1584.             }

  1585.         //the voice is killed later

  1586. 

  1587. 

  1588.         // Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator)

  1589.         if(NoteVoicePar[nvoice].VoiceOut) {

  1590.             if(stereo)

  1591.                 for(int i = 0; i < synth.buffersize; ++i)

  1592.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i]

  1593.                                                        + tmpwaver[i];

  1594.             else   //mono

  1595.                 for(int i = 0; i < synth.buffersize; ++i)

  1596.                     NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i];

  1597.         }

  1598. 

  1599. 

  1600.         // Add the voice that do not bypass the filter to out

  1601.         if(NoteVoicePar[nvoice].filterbypass == 0) { //no bypass

  1602.             if(stereo)

  1603.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1604.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1605.                                * NoteVoicePar[nvoice].Panning * 2.0f;

  1606.                     outr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1607.                                * (1.0f - NoteVoicePar[nvoice].Panning) * 2.0f;

  1608.                 }

  1609.             else

  1610.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1611.                     outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1612.         }

  1613.         else {  //bypass the filter

  1614.             if(stereo)

  1615.                 for(int i = 0; i < synth.buffersize; ++i) { //stereo

  1616.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume

  1617.                                   * NoteVoicePar[nvoice].Panning * 2.0f;

  1618.                     bypassr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume

  1619.                                   * (1.0f

  1620.                                      - NoteVoicePar[nvoice].Panning) * 2.0f;

  1621.                 }

  1622.             else

  1623.                 for(int i = 0; i < synth.buffersize; ++i) //mono

  1624.                     bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;

  1625.         }

  1626.         // chech if there is necesary to proces the voice longer (if the Amplitude envelope isn't finished)

  1627.         if(NoteVoicePar[nvoice].AmpEnvelope)

  1628.             if(NoteVoicePar[nvoice].AmpEnvelope->finished())

  1629.                 KillVoice(nvoice);

  1630.     }

  1631. 

  1632. 

  1633.     //Processing Global parameters

  1634.     NoteGlobalPar.GlobalFilterL->filterout(&outl[0]);

  1635. 

  1636.     if(stereo == 0) { //set the right channel=left channel

  1637.         memcpy(outr, outl, synth.bufferbytes);

  1638.         memcpy(bypassr, bypassl, synth.bufferbytes);

  1639.     }

  1640.     else

  1641.         NoteGlobalPar.GlobalFilterR->filterout(&outr[0]);

  1642. 

  1643.     for(int i = 0; i < synth.buffersize; ++i) {

  1644.         outl[i] += bypassl[i];

  1645.         outr[i] += bypassr[i];

  1646.     }

  1647. 

  1648.     if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))

  1649.         // Amplitude Interpolation

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

  1651.             float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,

  1652.                                                  globalnewamplitude,

  1653.                                                  i,

  1654.                                                  synth.buffersize);

  1655.             outl[i] *= tmpvol * NoteGlobalPar.Panning;

  1656.             outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);

  1657.         }

  1658.     else

  1659.         for(int i = 0; i < synth.buffersize; ++i) {

  1660.             outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;

  1661.             outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);

  1662.         }

  1663. 

  1664.     //Apply the punch

  1665.     if(NoteGlobalPar.Punch.Enabled != 0)

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

  1667.             float punchamp = NoteGlobalPar.Punch.initialvalue

  1668.                              * NoteGlobalPar.Punch.t + 1.0f;

  1669.             outl[i] *= punchamp;

  1670.             outr[i] *= punchamp;

  1671.             NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;

  1672.             if(NoteGlobalPar.Punch.t < 0.0f) {

  1673.                 NoteGlobalPar.Punch.Enabled = 0;

  1674.                 break;

  1675.             }

  1676.         }

  1677. 

  1678. 

  1679.     // Apply legato-specific sound signal modifications

  1680.     legato.apply(*this, outl, outr);

  1681. 

  1682. 

  1683.     // Check if the global amplitude is finished.

  1684.     // If it does, disable the note

  1685.     if(NoteGlobalPar.AmpEnvelope->finished()) {

  1686.         for(int i = 0; i < synth.buffersize; ++i) { //fade-out

  1687.             float tmp = 1.0f - (float)i / synth.buffersize_f;

  1688.             outl[i] *= tmp;

  1689.             outr[i] *= tmp;

  1690.         }

  1691.         KillNote();

  1692.     }

  1693.     return 1;

  1694. }

  1695. 

  1696. 

  1697. /*

  1698.  * Release the key (NoteOff)

  1699.  */

  1700. void ADnote::releasekey()

  1701. {

  1702.     for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)

  1703.         NoteVoicePar[nvoice].releasekey();

  1704.     NoteGlobalPar.FreqEnvelope->releasekey();

  1705.     NoteGlobalPar.FilterEnvelope->releasekey();

  1706.     NoteGlobalPar.AmpEnvelope->releasekey();

  1707. }

  1708. 

  1709. /*

  1710.  * Check if the note is finished

  1711.  */

  1712. int ADnote::finished() const

  1713. {

  1714.     if(NoteEnabled == ON)

  1715.         return 0;

  1716.     else

  1717.         return 1;

  1718. }

  1719. 

  1720. void ADnote::Voice::releasekey()

  1721. {

  1722.     if(!Enabled)

  1723.         return;

  1724.     if(AmpEnvelope)

  1725.         AmpEnvelope->releasekey();

  1726.     if(FreqEnvelope)

  1727.         FreqEnvelope->releasekey();

  1728.     if(FilterEnvelope)

  1729.         FilterEnvelope->releasekey();

  1730.     if(FMFreqEnvelope)

  1731.         FMFreqEnvelope->releasekey();

  1732.     if(FMAmpEnvelope)

  1733.         FMAmpEnvelope->releasekey();

  1734. }

  1735. 

  1736. void ADnote::Voice::kill(Allocator &memory, const SYNTH_T &synth)

  1737. {

  1738.     memory.devalloc(OscilSmp);

  1739.     memory.dealloc(FreqEnvelope);

  1740.     memory.dealloc(FreqLfo);

  1741.     memory.dealloc(AmpEnvelope);

  1742.     memory.dealloc(AmpLfo);

  1743.     memory.dealloc(VoiceFilterL);

  1744.     memory.dealloc(VoiceFilterR);

  1745.     memory.dealloc(FilterEnvelope);

  1746.     memory.dealloc(FilterLfo);

  1747.     memory.dealloc(FMFreqEnvelope);

  1748.     memory.dealloc(FMAmpEnvelope);

  1749. 

  1750.     if((FMEnabled != NONE) && (FMVoice < 0))

  1751.         memory.devalloc(FMSmp);

  1752. 

  1753.     if(VoiceOut)

  1754.         memset(VoiceOut, 0, synth.bufferbytes);

  1755.     //the buffer can't be safely deleted as it may be

  1756.     //an input to another voice

  1757. 

  1758.     Enabled = OFF;

  1759. }

  1760. 

  1761. void ADnote::Global::kill(Allocator &memory)

  1762. {

  1763.     memory.dealloc(FreqEnvelope);

  1764.     memory.dealloc(FreqLfo);

  1765.     memory.dealloc(AmpEnvelope);

  1766.     memory.dealloc(AmpLfo);

  1767.     memory.dealloc(GlobalFilterL);

  1768.     memory.dealloc(GlobalFilterR);

  1769.     memory.dealloc(FilterEnvelope);

  1770.     memory.dealloc(FilterLfo);

  1771. }

  1772. 

  1773. void ADnote::Global::initparameters(const ADnoteGlobalParam &param,

  1774.                                     const SYNTH_T &synth,

  1775.                                     class Allocator &memory,

  1776.                                     float basefreq, float velocity,

  1777.                                     bool stereo)

  1778. {

  1779.     FreqEnvelope = memory.alloc<Envelope>(*param.FreqEnvelope, basefreq, synth.dt());

  1780.     FreqLfo      = memory.alloc<LFO>(*param.FreqLfo, basefreq, synth.dt());

  1781. 

  1782.     AmpEnvelope = memory.alloc<Envelope>(*param.AmpEnvelope, basefreq, synth.dt());

  1783.     AmpLfo      = memory.alloc<LFO>(*param.AmpLfo, basefreq, synth.dt());

  1784. 

  1785.     Volume = 4.0f * powf(0.1f, 3.0f * (1.0f - param.PVolume / 96.0f)) //-60 dB .. 0 dB

  1786.              * VelF(velocity, param.PAmpVelocityScaleFunction);     //sensing

  1787. 

  1788.     GlobalFilterL = Filter::generate(memory, param.GlobalFilter,

  1789.             synth.samplerate, synth.buffersize);

  1790.     if(stereo)

  1791.         GlobalFilterR = Filter::generate(memory, param.GlobalFilter,

  1792.                 synth.samplerate, synth.buffersize);

  1793.     else

  1794.         GlobalFilterR = NULL;

  1795. 

  1796.     FilterEnvelope = memory.alloc<Envelope>(*param.FilterEnvelope, basefreq, synth.dt());

  1797.     FilterLfo      = memory.alloc<LFO>(*param.FilterLfo, basefreq, synth.dt());

  1798.     FilterQ = param.GlobalFilter->getq();

  1799.     FilterFreqTracking = param.GlobalFilter->getfreqtracking(basefreq);

  1800. }