/*

TiMidity -- Experimental MIDI to WAVE converter

Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>

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

it under the terms of the GNU General Public License as published by

the Free Software Foundation; either version 2 of the License, or

(at your option) any later version.

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

but WITHOUT ANY WARRANTY; without even the implied warranty of

MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the

GNU General Public License for more details.

You should have received a copy of the GNU General Public License

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

Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

playmidi.c -- random stuff in need of rearrangement

*/

#include <stdio.h>

#include <stdlib.h>

#include <string.h>

#include <SDL_rwops.h>

#include "config.h"

#include "common.h"

#include "instrum.h"

#include "playmidi.h"

#include "readmidi.h"

#include "output.h"

#include "mix.h"

#include "timidity.h"

#include "tables.h"

static int opt_expression_curve = 2;

static int opt_volume_curve = 2;

static int opt_stereo_surround = 0;

Channel channel[MAXCHAN];

signed char drumvolume[MAXCHAN][MAXNOTE];

signed char drumpanpot[MAXCHAN][MAXNOTE];

signed char drumreverberation[MAXCHAN][MAXNOTE];

signed char drumchorusdepth[MAXCHAN][MAXNOTE];

int

voices=DEFAULT_VOICES;

int32

control_ratio=0,

amplification=DEFAULT_AMPLIFICATION;

master_volume;

int32 drumchannels=DEFAULT_DRUMCHANNELS;

int adjust_panning_immediately=0;

struct _MidiSong {

int32 samples;

MidiEvent *events;

};

static int midi_playing = 0;

static int32 lost_notes, cut_notes;

static int32 *buffer_pointer;

static int32 buffered_count;

extern int32 *common_buffer;

static MidiEvent *event_list, *current_event;

static int32 sample_count, current_sample;

int GM_System_On=0;

int XG_System_On=0;

int GS_System_On=0;

int XG_System_reverb_type;

int XG_System_chorus_type;

int XG_System_variation_type;

static void adjust_amplification(void)

{

master_volume = (FLOAT_T)(amplification) / (FLOAT_T)100.0;

master_volume /= 2;

}

static void adjust_master_volume(int32 vol)

{

master_volume = (double)(vol*amplification) / 1638400.0L;

master_volume /= 2;

}

static void reset_voices(void)

{

int i;

for (i=0; i<MAX_VOICES; i++)

voice[i].status=VOICE_FREE;

}

/* Process the Reset All Controllers event */

static void reset_controllers(int c)

{

channel[c].volume=90; /* Some standard says, although the SCC docs say 0. */

channel[c].expression=127; /* SCC-1 does this. */

channel[c].sustain=0;

channel[c].pitchbend=0x2000;

channel[c].pitchfactor=0; /* to be computed */

channel[c].reverberation = 0;

channel[c].chorusdepth = 0;

}

static void redraw_controllers(int c)

{

ctl->volume(c, channel[c].volume);

ctl->expression(c, channel[c].expression);

ctl->sustain(c, channel[c].sustain);

ctl->pitch_bend(c, channel[c].pitchbend);

}

static void reset_midi(void)

{

int i;

{

reset_controllers(i);

/* The rest of these are unaffected by the Reset All Controllers event */

channel[i].program=default_program;

channel[i].panning=NO_PANNING;

channel[i].pitchsens=2;

channel[i].bank=0; /* tone bank or drum set */

channel[i].harmoniccontent=64,

channel[i].releasetime=64,

channel[i].attacktime=64,

channel[i].brightness=64,

channel[i].sfx=0;

}

reset_voices();

}

static void select_sample(int v, Instrument *ip)

{

int s,i;

Sample *sp, *closest;

s=ip->samples;

sp=ip->sample;

if (s==1)

{

voice[v].sample=sp;

return;

}

f=voice[v].orig_frequency;

/*

No suitable sample found! We'll select the sample whose root

frequency is closest to the one we want. (Actually we should

probably convert the low, high, and root frequencies to MIDI note

values and compare those.) */

cdiff=0x7FFFFFFF;

closest=sp=ip->sample;

for(i=0; i<s; i++)

{

diff=sp->root_freq - f;

/* But the root freq. can perfectly well lie outside the keyrange

* frequencies, so let's try:

*/

/* diff=midfreq - f; */

if (diff<0) diff=-diff;

if (diff<cdiff)

{

cdiff=diff;

closest=sp;

}

sp++;

}

voice[v].sample=closest;

return;

}

static void select_stereo_samples(int v, InstrumentLayer *lp)

{

Instrument *ip;

InstrumentLayer *nlp, *bestvel;

int diffvel, midvel, mindiff;

/* select closest velocity */

bestvel = lp;

mindiff = 500;

for (nlp = lp; nlp; nlp = nlp->next) {

midvel = (nlp->hi + nlp->lo)/2;

if (!midvel) diffvel = 127;

else if (voice[v].velocity < nlp->lo || voice[v].velocity > nlp->hi)

diffvel = 200;

else diffvel = voice[v].velocity - midvel;

if (diffvel < 0) diffvel = -diffvel;

if (diffvel < mindiff) {

mindiff = diffvel;

bestvel = nlp;

}

}

ip = bestvel->instrument;

if (ip->right_sample) {

ip->sample = ip->right_sample;

ip->samples = ip->right_samples;

select_sample(v, ip);

voice[v].right_sample = voice[v].sample;

}

else voice[v].right_sample = 0;

ip->sample = ip->left_sample;

ip->samples = ip->left_samples;

select_sample(v, ip);

}

static void recompute_freq(int v)

{

int

sign=(voice[v].sample_increment < 0), /* for bidirectional loops */

pb=channel[voice[v].channel].pitchbend;

double a;

if (!voice[v].sample->sample_rate)

return;

if (voice[v].vibrato_control_ratio)

{

/* This instrument has vibrato. Invalidate any precomputed

sample_increments. */

int i=VIBRATO_SAMPLE_INCREMENTS;

while (i--)

voice[v].vibrato_sample_increment[i]=0;

}

if (pb==0x2000 || pb<0 || pb>0x3FFF)

voice[v].frequency=voice[v].orig_frequency;

else

{

pb-=0x2000;

if (!(channel[voice[v].channel].pitchfactor))

{

/* Damn. Somebody bent the pitch. */

int32 i=pb*channel[voice[v].channel].pitchsens;

if (pb<0)

i=-i;

channel[voice[v].channel].pitchfactor=

}

if (pb>0)

voice[v].frequency=

(int32)(channel[voice[v].channel].pitchfactor *

(double)(voice[v].orig_frequency));

else

voice[v].frequency=

(int32)((double)(voice[v].orig_frequency) /

channel[voice[v].channel].pitchfactor);

}

a = FSCALE(((double)(voice[v].sample->sample_rate) *

(double)(voice[v].frequency)) /

((double)(voice[v].sample->root_freq) *

(double)(play_mode->rate)),

FRACTION_BITS);

if (sign)

a = -a; /* need to preserve the loop direction */

voice[v].sample_increment = (int32)(a);

}

static int expr_curve[128] = {

7, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 10, 10, 10, 10, 11,

11, 11, 11, 12, 12, 12, 12, 13, 13, 13, 14, 14, 14, 14, 15, 15,

15, 16, 16, 17, 17, 17, 18, 18, 19, 19, 19, 20, 20, 21, 21, 22,

22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 28, 28, 29, 30, 30, 31,

32, 32, 33, 34, 35, 35, 36, 37, 38, 39, 39, 40, 41, 42, 43, 44,

45, 46, 47, 48, 49, 50, 51, 53, 54, 55, 56, 57, 59, 60, 61, 63,

64, 65, 67, 68, 70, 71, 73, 75, 76, 78, 80, 82, 83, 85, 87, 89,

91, 93, 95, 97, 99, 102, 104, 106, 109, 111, 113, 116, 118, 121,

124, 127

};

static int panf(int pan, int speaker, int separation)

{

int val;

val = abs(pan - speaker);

val = (val * 127) / separation;

val = 127 - val;

if (val < 0) val = 0;

if (val > 127) val = 127;

return expr_curve[val];

}

static int vcurve[128] = {

0,0,18,29,36,42,47,51,55,58,

60,63,65,67,69,71,73,74,76,77,

79,80,81,82,83,84,85,86,87,88,

89,90,91,92,92,93,94,95,95,96,

97,97,98,99,99,100,100,101,101,102,

103,103,104,104,105,105,106,106,106,107,

107,108,108,109,109,109,110,110,111,111,

111,112,112,112,113,113,114,114,114,115,

115,115,116,116,116,116,117,117,117,118,

118,118,119,119,119,119,120,120,120,120,

121,121,121,122,122,122,122,123,123,123,

123,123,124,124,124,124,125,125,125,125,

126,126,126,126,126,127,127,127

};

static void recompute_amp(int v)

{

int32 tempamp;

int chan = voice[v].channel;

int panning = voice[v].panning;

int vol = channel[chan].volume;

int expr = channel[chan].expression;

int vel = vcurve[voice[v].velocity];

FLOAT_T curved_expression, curved_volume;

if (channel[chan].kit)

{

int note = voice[v].sample->note_to_use;

if (note>0 && drumvolume[chan][note]>=0) vol = drumvolume[chan][note];

if (note>0 && drumpanpot[chan][note]>=0) panning = drumvolume[chan][note];

}

if (opt_expression_curve == 2) curved_expression = 127.0 * vol_table[expr];

else if (opt_expression_curve == 1) curved_expression = 127.0 * expr_table[expr];

else curved_expression = (FLOAT_T)expr;

if (opt_volume_curve == 2) curved_volume = 127.0 * vol_table[vol];

else if (opt_volume_curve == 1) curved_volume = 127.0 * expr_table[vol];

else curved_volume = (FLOAT_T)vol;

tempamp= (int32)((FLOAT_T)vel * curved_volume * curved_expression); /* 21 bits */

/* TODO: use fscale */

{

voice[v].panned=PANNED_CENTER;

if (num_ochannels == 6) voice[v].left_amp =

FSCALENEG((double) (tempamp) * voice[v].sample->volume *

master_volume, 20);

else voice[v].left_amp=

FSCALENEG((double)(tempamp) * voice[v].sample->volume *

master_volume, 21);

{

voice[v].panned = PANNED_LEFT;

voice[v].left_amp=

FSCALENEG((double)(tempamp) * voice[v].sample->volume * master_volume,

20);

}

{

voice[v].panned = PANNED_RIGHT;

voice[v].left_amp= /* left_amp will be used */

FSCALENEG((double)(tempamp) * voice[v].sample->volume * master_volume,

20);

}

else

{

FLOAT_T refv = (double)(tempamp) * voice[v].sample->volume * master_volume;

int wide_panning = 64;

if (num_ochannels == 4) wide_panning = 95;

voice[v].lfe_amp = FSCALENEG(refv * 64, 27);

switch (num_ochannels)

{

case 2:

voice[v].lr_amp = 0;

voice[v].left_amp = FSCALENEG(refv * (128-panning), 27);

voice[v].ce_amp = 0;

voice[v].right_amp = FSCALENEG(refv * panning, 27);

voice[v].rr_amp = 0;

break;

case 4:

voice[v].lr_amp = FSCALENEG(refv * panf(panning, 0, wide_panning), 27);

voice[v].left_amp = FSCALENEG(refv * panf(panning, 32, wide_panning), 27);

voice[v].ce_amp = 0;

voice[v].right_amp = FSCALENEG(refv * panf(panning, 95, wide_panning), 27);

voice[v].rr_amp = FSCALENEG(refv * panf(panning, 128, wide_panning), 27);

break;

case 6:

voice[v].lr_amp = FSCALENEG(refv * panf(panning, 0, wide_panning), 27);

voice[v].left_amp = FSCALENEG(refv * panf(panning, 32, wide_panning), 27);

voice[v].ce_amp = FSCALENEG(refv * panf(panning, 64, wide_panning), 27);

voice[v].right_amp = FSCALENEG(refv * panf(panning, 95, wide_panning), 27);

voice[v].rr_amp = FSCALENEG(refv * panf(panning, 128, wide_panning), 27);

break;

}

}

}

else

{

voice[v].panned=PANNED_CENTER;

voice[v].left_amp=

FSCALENEG((double)(tempamp) * voice[v].sample->volume * master_volume,

21);

}

}

#define NOT_CLONE 0

#define STEREO_CLONE 1

#define REVERB_CLONE 2

#define CHORUS_CLONE 3

/* just a variant of note_on() */

static int vc_alloc(int j)

{

int i=voices;

while (i--)

{

if (i == j) continue;

if (voice[i].status & VOICE_FREE) {

return i;

}

}

return -1;

}

static void kill_note(int i);

static void kill_others(int i)

{

int j=voices;

if (!voice[i].sample->exclusiveClass) return;

while (j--)

{

if (voice[j].status & (VOICE_FREE|VOICE_OFF|VOICE_DIE)) continue;

if (i == j) continue;

if (voice[i].channel != voice[j].channel) continue;

if (voice[j].sample->note_to_use)

{

if (voice[j].sample->exclusiveClass != voice[i].sample->exclusiveClass) continue;

kill_note(j);

}

}

}

static void clone_voice(Instrument *ip, int v, MidiEvent *e, int clone_type, int variationbank)

{

int chan = voice[v].channel;

if (clone_type == STEREO_CLONE) {

if (!voice[v].right_sample && variationbank != 3) return;

if (variationbank == 6) return;

}

if (channel[chan].kit) {

reverb = drumreverberation[chan][voice[v].note];

chorus = drumchorusdepth[chan][voice[v].note];

}

else {

reverb = channel[chan].reverberation;

chorus = channel[chan].chorusdepth;

}

if (clone_type == REVERB_CLONE) chorus = 0;

else if (clone_type == CHORUS_CLONE) reverb = 0;

else if (clone_type == STEREO_CLONE) reverb = chorus = 0;

if (reverb > 127) reverb = 127;

if (chorus > 127) chorus = 127;

if (clone_type == CHORUS_CLONE) {

if (variationbank == 32) chorus = 30;

else if (variationbank == 33) chorus = 60;

else if (variationbank == 34) chorus = 90;

}

chorus /= 2; /* This is an ad hoc adjustment. */

if (!reverb && !chorus && clone_type != STEREO_CLONE) return;

if ( (w = vc_alloc(v)) < 0 ) return;

voice[w] = voice[v];

if (clone_type==STEREO_CLONE) voice[v].clone_voice = w;

voice[w].clone_voice = v;

voice[w].clone_type = clone_type;

voice[w].sample = voice[v].right_sample;

voice[w].velocity= e->b;

milli = play_mode->rate/1000;

if (clone_type == STEREO_CLONE) {

int left, right, leftpan, rightpan;

int panrequest = voice[v].panning;

if (variationbank == 3) {

voice[v].panning = 0;

voice[w].panning = 127;

}

else {

if (voice[v].sample->panning > voice[w].sample->panning) {

left = w;

right = v;

}

else {

left = v;

right = w;

}

#define INSTRUMENT_SEPARATION 12

leftpan = panrequest - INSTRUMENT_SEPARATION / 2;

rightpan = leftpan + INSTRUMENT_SEPARATION;

if (leftpan < 0) {

leftpan = 0;

rightpan = leftpan + INSTRUMENT_SEPARATION;

}

if (rightpan > 127) {

rightpan = 127;

leftpan = rightpan - INSTRUMENT_SEPARATION;

}

voice[left].panning = leftpan;

voice[right].panning = rightpan;

voice[right].echo_delay = 20 * milli;

}

}

voice[w].volume = voice[w].sample->volume;

if (reverb) {

if (opt_stereo_surround) {

if (voice[w].panning > 64) voice[w].panning = 127;

else voice[w].panning = 0;

}

else {

if (voice[v].panning < 64) voice[w].panning = 64 + reverb/2;

else voice[w].panning = 64 - reverb/2;

}

/* try 98->99 for melodic instruments ? (bit much for percussion) */

voice[w].volume *= vol_table[(127-reverb)/8 + 98];

voice[w].echo_delay += reverb * milli;

voice[w].envelope_rate[DECAY] *= 2;

voice[w].envelope_rate[RELEASE] /= 2;

if (XG_System_reverb_type >= 0) {

int subtype = XG_System_reverb_type & 0x07;

int rtype = XG_System_reverb_type >>3;

switch (rtype) {

case 0: /* no effect */

break;

case 1: /* hall */

if (subtype) voice[w].echo_delay += 100 * milli;

break;

case 2: /* room */

voice[w].echo_delay /= 2;

break;

case 3: /* stage */

voice[w].velocity = voice[v].velocity;

break;

case 4: /* plate */

voice[w].panning = voice[v].panning;

break;

case 16: /* white room */

voice[w].echo_delay = 0;

break;

case 17: /* tunnel */

voice[w].echo_delay *= 2;

voice[w].velocity /= 2;

break;

case 18: /* canyon */

voice[w].echo_delay *= 2;

break;

case 19: /* basement */

voice[w].velocity /= 2;

break;

default: break;

}

}

}

played_note = voice[w].sample->note_to_use;

if (!played_note) {

played_note = e->a & 0x7f;

if (variationbank == 35) played_note += 12;

else if (variationbank == 36) played_note -= 12;

else if (variationbank == 37) played_note += 7;

else if (variationbank == 36) played_note -= 7;

}

#if 0

played_note = ( (played_note - voice[w].sample->freq_center) * voice[w].sample->freq_scale ) / 1024 +

voice[w].sample->freq_center;

#endif

voice[w].note = played_note;

voice[w].orig_frequency = freq_table[played_note];

if (chorus) {

if (opt_stereo_surround) {

if (voice[v].panning < 64) voice[w].panning = voice[v].panning + 32;

else voice[w].panning = voice[v].panning - 32;

}

if (!voice[w].vibrato_control_ratio) {

voice[w].vibrato_control_ratio = 100;

voice[w].vibrato_depth = 6;

voice[w].vibrato_sweep = 74;

}

voice[w].volume *= 0.40;

voice[v].volume = voice[w].volume;

recompute_amp(v);

apply_envelope_to_amp(v);

voice[w].vibrato_sweep = chorus/2;

voice[w].vibrato_depth /= 2;

if (!voice[w].vibrato_depth) voice[w].vibrato_depth = 2;

voice[w].vibrato_control_ratio /= 2;

voice[w].echo_delay += 30 * milli;

if (XG_System_chorus_type >= 0) {

int subtype = XG_System_chorus_type & 0x07;

int chtype = 0x0f & (XG_System_chorus_type >> 3);

switch (chtype) {

case 0: /* no effect */

break;

case 1: /* chorus */

chorus /= 3;

if(channel[ voice[w].channel ].pitchbend + chorus < 0x2000)

voice[w].orig_frequency =

(uint32)( (FLOAT_T)voice[w].orig_frequency * bend_fine[chorus] );

else voice[w].orig_frequency =

(uint32)( (FLOAT_T)voice[w].orig_frequency / bend_fine[chorus] );

if (subtype) voice[w].vibrato_depth *= 2;

break;

case 2: /* celeste */

voice[w].orig_frequency += (voice[w].orig_frequency/128) * chorus;

break;

case 3: /* flanger */

voice[w].vibrato_control_ratio = 10;

voice[w].vibrato_depth = 100;

voice[w].vibrato_sweep = 8;

voice[w].echo_delay += 200 * milli;

break;

case 4: /* symphonic : cf Children of the Night /128 bad, /1024 ok */

voice[w].orig_frequency += (voice[w].orig_frequency/512) * chorus;

voice[v].orig_frequency -= (voice[v].orig_frequency/512) * chorus;

recompute_freq(v);

break;

case 8: /* phaser */

break;

default:

break;

}

}

else {

chorus /= 3;

if(channel[ voice[w].channel ].pitchbend + chorus < 0x2000)

voice[w].orig_frequency =

(uint32)( (FLOAT_T)voice[w].orig_frequency * bend_fine[chorus] );

else voice[w].orig_frequency =

(uint32)( (FLOAT_T)voice[w].orig_frequency / bend_fine[chorus] );

}

}

#if 0

voice[w].loop_start = voice[w].sample->loop_start;

voice[w].loop_end = voice[w].sample->loop_end;

#endif

voice[w].echo_delay_count = voice[w].echo_delay;

if (reverb) voice[w].echo_delay *= 2;

recompute_freq(w);

recompute_amp(w);

if (voice[w].sample->modes & MODES_ENVELOPE)

{

/* Ramp up from 0 */

voice[w].envelope_stage=ATTACK;

voice[w].modulation_stage=ATTACK;

voice[w].envelope_volume=0;

voice[w].modulation_volume=0;

voice[w].control_counter=0;

voice[w].modulation_counter=0;

recompute_envelope(w);

/*recompute_modulation(w);*/

}

else

{

voice[w].envelope_increment=0;

voice[w].modulation_increment=0;

}

apply_envelope_to_amp(w);

}

static void xremap(int *banknumpt, int *this_notept, int this_kit) {

int i, newmap;

int banknum = *banknumpt;

int this_note = *this_notept;

int newbank, newnote;

if (!this_kit) {

if (banknum == SFXBANK && tonebank[SFXBANK]) return;

if (banknum == SFXBANK && tonebank[120]) *banknumpt = 120;

return;

}

if (this_kit != 127 && this_kit != 126) return;

for (i = 0; i < XMAPMAX; i++) {

newmap = xmap[i][0];

if (!newmap) return;

if (this_kit == 127 && newmap != XGDRUM) continue;

if (this_kit == 126 && newmap != SFXDRUM1) continue;

if (xmap[i][1] != banknum) continue;

if (xmap[i][3] != this_note) continue;

newbank = xmap[i][2];

newnote = xmap[i][4];

if (newbank == banknum && newnote == this_note) return;

if (!drumset[newbank]) return;

if (!drumset[newbank]->tone[newnote].layer) return;

if (drumset[newbank]->tone[newnote].layer == MAGIC_LOAD_INSTRUMENT) return;

*banknumpt = newbank;

*this_notept = newnote;

return;

}

}

static void start_note(MidiEvent *e, int i)

{

int j, banknum, ch=e->channel;

int played_note, drumpan=NO_PANNING;

int32 rt;

int attacktime, releasetime, decaytime, variationbank;

int brightness = channel[ch].brightness;

int harmoniccontent = channel[ch].harmoniccontent;

int this_note = e->a;

int this_velocity = e->b;

int drumsflag = channel[ch].kit;

int this_prog = channel[ch].program;

if (channel[ch].sfx) banknum=channel[ch].sfx;

else banknum=channel[ch].bank;

voice[i].velocity=this_velocity;

if (XG_System_On) xremap(&banknum, &this_note, drumsflag);

/* if (current_config_pc42b) pcmap(&banknum, &this_note, &this_prog, &drumsflag); */

if (drumsflag)

return; /* No instrument? Then we can't play. */

}

ip = lp->instrument;

if (ip->type == INST_GUS && ip->samples != 1)

{

ctl->cmsg(CMSG_WARNING, VERB_VERBOSE,

"Strange: percussion instrument with %d samples!", ip->samples);

}

if (ip->sample->note_to_use) /* Do we have a fixed pitch? */

{

voice[i].orig_frequency=freq_table[(int)(ip->sample->note_to_use)];

drumpan=drumpanpot[ch][(int)ip->sample->note_to_use];

}

voice[i].orig_frequency=freq_table[this_note & 0x7F];

}

else

{

if (channel[ch].program==SPECIAL_PROGRAM)

lp=default_instrument;

else if (!(lp=tonebank[channel[ch].bank]->

tone[channel[ch].program].layer))

return; /* No instrument? Then we can't play. */

}

if (ip->sample->note_to_use) /* Fixed-pitch instrument? */

voice[i].orig_frequency=freq_table[(int)(ip->sample->note_to_use)];

else

}

select_stereo_samples(i, lp);

voice[i].starttime = e->time;

played_note = voice[i].sample->note_to_use;

if (!played_note || !drumsflag) played_note = this_note & 0x7f;

#if 0

played_note = ( (played_note - voice[i].sample->freq_center) * voice[i].sample->freq_scale ) / 1024 +

voice[i].sample->freq_center;

#endif

voice[i].channel=ch;

voice[i].note=played_note;

voice[i].velocity=this_velocity;

voice[i].sample_offset=0;

voice[i].sample_increment=0; /* make sure it isn't negative */

voice[i].tremolo_phase=0;

voice[i].tremolo_phase_increment=voice[i].sample->tremolo_phase_increment;

voice[i].tremolo_sweep=voice[i].sample->tremolo_sweep_increment;

voice[i].tremolo_sweep_position=0;

voice[i].vibrato_sweep=voice[i].sample->vibrato_sweep_increment;

voice[i].vibrato_sweep_position=0;

voice[i].vibrato_control_ratio=voice[i].sample->vibrato_control_ratio;

voice[i].vibrato_control_counter=voice[i].vibrato_phase=0;

voice[i].vibrato_delay = voice[i].sample->vibrato_delay;

kill_others(i);

for (j=0; j<VIBRATO_SAMPLE_INCREMENTS; j++)

voice[i].vibrato_sample_increment[j]=0;

attacktime = channel[ch].attacktime;

releasetime = channel[ch].releasetime;

decaytime = 64;

variationbank = channel[ch].variationbank;

switch (variationbank) {

case 8:

attacktime = 64+32;

break;

case 12:

decaytime = 64-32;

break;

case 16:

brightness = 64+16;

break;

case 17:

brightness = 64+32;

break;

case 18:

brightness = 64-16;

break;

case 19:

brightness = 64-32;

break;

case 20:

harmoniccontent = 64+16;

break;

#if 0

case 24:

voice[i].modEnvToFilterFc=2.0;

voice[i].sample->cutoff_freq = 800;

break;

case 25:

voice[i].modEnvToFilterFc=-2.0;

voice[i].sample->cutoff_freq = 800;

break;

case 27:

voice[i].modLfoToFilterFc=2.0;

voice[i].lfo_phase_increment=109;

voice[i].lfo_sweep=122;

voice[i].sample->cutoff_freq = 800;

break;

case 28:

voice[i].modLfoToFilterFc=-2.0;

voice[i].lfo_phase_increment=109;

voice[i].lfo_sweep=122;

voice[i].sample->cutoff_freq = 800;

break;

#endif

default:

break;

}

for (j=ATTACK; j<MAXPOINT; j++)

{

voice[i].envelope_rate[j]=voice[i].sample->envelope_rate[j];

voice[i].envelope_offset[j]=voice[i].sample->envelope_offset[j];

}

voice[i].echo_delay=voice[i].envelope_rate[DELAY];

voice[i].echo_delay_count = voice[i].echo_delay;

if (attacktime!=64)

{

rt = voice[i].envelope_rate[ATTACK];

rt = rt + ( (64-attacktime)*rt ) / 100;

if (rt > 1000) voice[i].envelope_rate[ATTACK] = rt;

}

if (releasetime!=64)

{

rt = voice[i].envelope_rate[RELEASE];

rt = rt + ( (64-releasetime)*rt ) / 100;

if (rt > 1000) voice[i].envelope_rate[RELEASE] = rt;

}

if (decaytime!=64)

{

rt = voice[i].envelope_rate[DECAY];

rt = rt + ( (64-decaytime)*rt ) / 100;

if (rt > 1000) voice[i].envelope_rate[DECAY] = rt;

}

if (channel[ch].panning != NO_PANNING)

voice[i].panning=channel[ch].panning;

else

voice[i].panning=voice[i].sample->panning;

if (drumpan != NO_PANNING)

voice[i].panning=drumpan;

if (variationbank == 1) {

int pan = voice[i].panning;

int disturb = 0;

/* If they're close up (no reverb) and you are behind the pianist,

* high notes come from the right, so we'll spread piano etc. notes

* out horizontally according to their pitches.

*/

if (this_prog < 21) {

int n = voice[i].velocity - 32;

if (n < 0) n = 0;

if (n > 64) n = 64;

pan = pan/2 + n;

}

/* For other types of instruments, the music sounds more alive if

* notes come from slightly different directions. However, instruments

* do drift around in a sometimes disconcerting way, so the following

* might not be such a good idea.

*/

else disturb = (voice[i].velocity/32 % 8) +

(voice[i].note % 8); /* /16? */

if (pan < 64) pan += disturb;

else pan -= disturb;

if (pan < 0) pan = 0;

else if (pan > 127) pan = 127;

voice[i].panning = pan;

}

recompute_freq(i);

recompute_amp(i);

if (voice[i].sample->modes & MODES_ENVELOPE)

{

/* Ramp up from 0 */

voice[i].envelope_volume=0;

voice[i].control_counter=0;

recompute_envelope(i);

}

else

{

voice[i].envelope_increment=0;

}

apply_envelope_to_amp(i);

voice[i].clone_voice = -1;

voice[i].clone_type = NOT_CLONE;

clone_voice(ip, i, e, STEREO_CLONE, variationbank);

clone_voice(ip, i, e, CHORUS_CLONE, variationbank);

clone_voice(ip, i, e, REVERB_CLONE, variationbank);

ctl->note(i);

}

static void kill_note(int i)

{