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playmidi.c
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/*
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>
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#include <SDL_rwops.h>
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#include "config.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "readmidi.h"
#include "output.h"
#include "mix.h"
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#include "ctrlmode.h"
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#include "timidity.h"
#include "tables.h"
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static int opt_expression_curve = 2;
static int opt_volume_curve = 2;
static int opt_stereo_surround = 0;
Channel channel[MAXCHAN];
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Voice voice[MAX_VOICES];
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signed char drumvolume[MAXCHAN][MAXNOTE];
signed char drumpanpot[MAXCHAN][MAXNOTE];
signed char drumreverberation[MAXCHAN][MAXNOTE];
signed char drumchorusdepth[MAXCHAN][MAXNOTE];
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int
voices=DEFAULT_VOICES;
int32
control_ratio=0,
amplification=DEFAULT_AMPLIFICATION;
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FLOAT_T
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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;
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extern resample_t *resample_buffer; /* to free it on Timidity_Close */
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static MidiEvent *event_list, *current_event;
static int32 sample_count, current_sample;
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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;
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static void adjust_amplification(void)
{
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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;
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}
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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 */
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channel[c].reverberation = 0;
channel[c].chorusdepth = 0;
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}
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;
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for (i=0; i<MAXCHAN; i++)
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{
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 */
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channel[i].harmoniccontent=64,
channel[i].releasetime=64,
channel[i].attacktime=64,
channel[i].brightness=64,
channel[i].sfx=0;
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}
reset_voices();
}
static void select_sample(int v, Instrument *ip)
{
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int32 f, cdiff, diff, midfreq;
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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;
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midfreq = (sp->low_freq + sp->high_freq) / 2;
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for(i=0; i<s; i++)
{
diff=sp->root_freq - f;
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/* But the root freq. can perfectly well lie outside the keyrange
* frequencies, so let's try:
*/
/* diff=midfreq - f; */
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if (diff<0) diff=-diff;
if (diff<cdiff)
{
cdiff=diff;
closest=sp;
}
sp++;
}
voice[v].sample=closest;
return;
}
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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);
}
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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=
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(FLOAT_T)(bend_fine[(i>>5) & 0xFF] * bend_coarse[i>>13]);
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}
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);
}
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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,
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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
};
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static void recompute_amp(int v)
{
int32 tempamp;
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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 */
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/* TODO: use fscale */
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if (num_ochannels > 1)
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{
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if (panning > 60 && panning < 68)
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{
voice[v].panned=PANNED_CENTER;
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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);
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}
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else if (panning<5)
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{
voice[v].panned = PANNED_LEFT;
voice[v].left_amp=
FSCALENEG((double)(tempamp) * voice[v].sample->volume * master_volume,
20);
}
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else if (panning>123)
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{
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
{
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FLOAT_T refv = (double)(tempamp) * voice[v].sample->volume * master_volume;
int wide_panning = 64;
if (num_ochannels == 4) wide_panning = 95;
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voice[v].panned = PANNED_MYSTERY;
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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;
}
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}
}
else
{
voice[v].panned=PANNED_CENTER;
voice[v].left_amp=
FSCALENEG((double)(tempamp) * voice[v].sample->volume * master_volume,
21);
}
}
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#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)
{
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int w, played_note, chorus=0, reverb=0, milli;
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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;
}
}
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759
static void start_note(MidiEvent *e, int i)
{
760
InstrumentLayer *lp;
761
Instrument *ip;
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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)
782
{
783
if (!(lp=drumset[banknum]->tone[this_note].layer))
784
{
785
if (!(lp=drumset[0]->tone[this_note].layer))
786
787
return; /* No instrument? Then we can't play. */
}
788
789
ip = lp->instrument;
if (ip->type == INST_GUS && ip->samples != 1)
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{
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? */
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{
voice[i].orig_frequency=freq_table[(int)(ip->sample->note_to_use)];
drumpan=drumpanpot[ch][(int)ip->sample->note_to_use];
}
800
else
801
802
voice[i].orig_frequency=freq_table[this_note & 0x7F];
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804
805
}
else
{
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807
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if (channel[ch].program==SPECIAL_PROGRAM)
lp=default_instrument;
else if (!(lp=tonebank[channel[ch].bank]->
tone[channel[ch].program].layer))
810
{
811
if (!(lp=tonebank[0]->tone[this_prog].layer))
812
813
return; /* No instrument? Then we can't play. */
}
814
ip = lp->instrument;
815
816
817
if (ip->sample->note_to_use) /* Fixed-pitch instrument? */
voice[i].orig_frequency=freq_table[(int)(ip->sample->note_to_use)];
else
818
voice[i].orig_frequency=freq_table[this_note & 0x7F];
819
820
}
821
822
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829
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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
831
voice[i].status=VOICE_ON;
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833
834
voice[i].channel=ch;
voice[i].note=played_note;
voice[i].velocity=this_velocity;
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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;
845
voice[i].vibrato_depth=voice[i].sample->vibrato_depth;
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847
voice[i].vibrato_control_ratio=voice[i].sample->vibrato_control_ratio;
voice[i].vibrato_control_counter=voice[i].vibrato_phase=0;
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849
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voice[i].vibrato_delay = voice[i].sample->vibrato_delay;
kill_others(i);
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for (j=0; j<VIBRATO_SAMPLE_INCREMENTS; j++)
voice[i].vibrato_sample_increment[j]=0;
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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;
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941
else
voice[i].panning=voice[i].sample->panning;
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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;
}
972
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974
975
976
977
recompute_freq(i);
recompute_amp(i);
if (voice[i].sample->modes & MODES_ENVELOPE)
{
/* Ramp up from 0 */
978
voice[i].envelope_stage=ATTACK;
979
980
981
982
983
984
985
986
voice[i].envelope_volume=0;
voice[i].control_counter=0;
recompute_envelope(i);
}
else
{
voice[i].envelope_increment=0;
}
987
988
989
990
991
992
993
994
995
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);
996
997
998
999
1000
ctl->note(i);
}
static void kill_note(int i)
{