/*
TiMidity++ -- MIDI to WAVE converter and player
Copyright (C) 1999-2004 Masanao Izumo <iz@onicos.co.jp>
Copyright (C) 1995 Tuukka Toivonen <tt@cgs.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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
instrum.c
Code to load and unload GUS-compatible instrument patches.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif /* HAVE_CONFIG_H */
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#ifndef NO_STRING_H
#include <string.h>
#else
#include <strings.h>
#endif
#include "timidity.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "readmidi.h"
#include "output.h"
#include "controls.h"
#include "resample.h"
#include "tables.h"
#include "filter.h"
#include "quantity.h"
#include "freq.h"
#define INSTRUMENT_HASH_SIZE 128
struct InstrumentCache
{
char *name;
int panning, amp, note_to_use, strip_loop, strip_envelope, strip_tail;
Instrument *ip;
struct InstrumentCache *next;
};
static struct InstrumentCache *instrument_cache[INSTRUMENT_HASH_SIZE];
/* Some functions get aggravated if not even the standard banks are
available. */
static ToneBank standard_tonebank, standard_drumset;
ToneBank
*tonebank[128 + MAP_BANK_COUNT] = {&standard_tonebank},
*drumset[128 + MAP_BANK_COUNT] = {&standard_drumset};
/* bank mapping (mapped bank) */
struct bank_map_elem {
int16 used, mapid;
int bankno;
};
static struct bank_map_elem map_bank[MAP_BANK_COUNT], map_drumset[MAP_BANK_COUNT];
static int map_bank_counter;
/* This is a special instrument, used for all melodic programs */
Instrument *default_instrument=0;
SpecialPatch *special_patch[NSPECIAL_PATCH];
int progbase = 0;
struct inst_map_elem
{
int set, elem, mapped;
};
static struct inst_map_elem *inst_map_table[NUM_INST_MAP][128];
/* This is only used for tracks that don't specify a program */
int default_program[MAX_CHANNELS];
char *default_instrument_name = NULL;
int antialiasing_allowed=0;
#ifdef FAST_DECAY
int fast_decay=1;
#else
int fast_decay=0;
#endif
/*Pseudo Reverb*/
int32 modify_release;
/** below three functinos are imported from sndfont.c **/
/* convert from 8bit value to fractional offset (15.15) */
static int32 to_offset(int offset)
{
return (int32)offset << (7+15);
}
/* calculate ramp rate in fractional unit;
* diff = 8bit, time = msec
*/
static int32 calc_rate(int diff, double msec)
{
double rate;
if(msec < 6)
msec = 6;
if(diff == 0)
diff = 255;
diff <<= (7+15);
rate = ((double)diff / play_mode->rate) * control_ratio * 1000.0 / msec;
if(fast_decay)
rate *= 2;
return (int32)rate;
}
/*End of Pseudo Reverb*/
void free_instrument(Instrument *ip)
{
Sample *sp;
int i;
if (!ip) return;
for (i=0; i<ip->samples; i++)
{
sp=&(ip->sample[i]);
if(sp->data_alloced)
free(sp->data);
}
free(ip->sample);
free(ip);
}
void clear_magic_instruments(void)
{
int i, j;
for(j = 0; j < 128 + map_bank_counter; j++)
{
if(tonebank[j])
{
ToneBank *bank = tonebank[j];
for(i = 0; i < 128; i++)
if(IS_MAGIC_INSTRUMENT(bank->tone[i].instrument))
bank->tone[i].instrument = NULL;
}
if(drumset[j])
{
ToneBank *bank = drumset[j];
for(i = 0; i < 128; i++)
if(IS_MAGIC_INSTRUMENT(bank->tone[i].instrument))
bank->tone[i].instrument = NULL;
}
}
}
#define GUS_ENVRATE_MAX (int32)(0x3FFFFFFF >> 9)
static int32 convert_envelope_rate(uint8 rate)
{
int32 r;
r=3-((rate>>6) & 0x3);
r*=3;
r = (int32)(rate & 0x3f) << r; /* 6.9 fixed point */
/* 15.15 fixed point. */
r = r * 44100 / play_mode->rate * control_ratio * (1 << fast_decay);
if(r > GUS_ENVRATE_MAX) {r = GUS_ENVRATE_MAX;}
return (r << 9);
}
static int32 convert_envelope_offset(uint8 offset)
{
/* This is not too good... Can anyone tell me what these values mean?
Are they GUS-style "exponential" volumes? And what does that mean? */
/* 15.15 fixed point */
return offset << (7+15);
}
static int32 convert_tremolo_sweep(uint8 sweep)
{
if (!sweep)
return 0;
return
((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
(play_mode->rate * sweep);
}
static int32 convert_vibrato_sweep(uint8 sweep, int32 vib_control_ratio)
{
if (!sweep)
return 0;
return (int32)(TIM_FSCALE((double) (vib_control_ratio)
* SWEEP_TUNING, SWEEP_SHIFT)
/ (double)(play_mode->rate * sweep));
/* this was overflowing with seashore.pat
((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
(play_mode->rate * sweep); */
}
static int32 convert_tremolo_rate(uint8 rate)
{
return
((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) /
(TREMOLO_RATE_TUNING * play_mode->rate);
}
static int32 convert_vibrato_rate(uint8 rate)
{
/* Return a suitable vibrato_control_ratio value */
return
(VIBRATO_RATE_TUNING * play_mode->rate) /
(rate * 2 * VIBRATO_SAMPLE_INCREMENTS);
}
static void reverse_data(int16 *sp, int32 ls, int32 le)
{
int16 s, *ep=sp+le;
int32 i;
sp+=ls;
le-=ls;
le/=2;
for(i = 0; i < le; i++)
{
s=*sp;
*sp++=*ep;
*ep--=s;
}
}
static int name_hash(char *name)
{
unsigned int addr = 0;
while(*name)
addr += *name++;
return addr % INSTRUMENT_HASH_SIZE;
}
static Instrument *search_instrument_cache(char *name,
int panning, int amp, int note_to_use,
int strip_loop, int strip_envelope,
int strip_tail)
{
struct InstrumentCache *p;
for(p = instrument_cache[name_hash(name)]; p != NULL; p = p->next)
{
if(strcmp(p->name, name) != 0)
return NULL;
if(p->panning == panning &&
p->amp == amp &&
p->note_to_use == note_to_use &&
p->strip_loop == strip_loop &&
p->strip_envelope == strip_envelope &&
p->strip_tail == strip_tail)
return p->ip;
}
return NULL;
}
static void store_instrument_cache(Instrument *ip,
char *name,
int panning, int amp, int note_to_use,
int strip_loop, int strip_envelope,
int strip_tail)
{
struct InstrumentCache *p;
int addr;
addr = name_hash(name);
p = (struct InstrumentCache *)safe_malloc(sizeof(struct InstrumentCache));
p->next = instrument_cache[addr];
instrument_cache[addr] = p;
p->name = name;
p->panning = panning;
p->amp = amp;
p->note_to_use = note_to_use;
p->strip_loop = strip_loop;
p->strip_envelope = strip_envelope;
p->strip_tail = strip_tail;
p->ip = ip;
}
static int32 adjust_tune_freq(int32 val, float tune)
{
if (! tune)
return val;
return val / pow(2.0, tune / 12.0);
}
static int16 adjust_scale_tune(int16 val)
{
return 1024 * (double) val / 100 + 0.5;
}
static int16 adjust_fc(int16 val)
{
if (val < 0 || val > play_mode->rate / 2) {
return 0;
} else {
return val;
}
}
static int16 adjust_reso(int16 val)
{
if (val < 0 || val > 960) {
return 0;
} else {
return val;
}
}
static int32 to_rate(int rate)
{
return (rate) ? (int32) (0x200 * pow(2.0, rate / 17.0)
* 44100 / play_mode->rate * control_ratio) << fast_decay : 0;
}
#if 0
static int32 to_control(int control)
{
return (int32) (0x2000 / pow(2.0, control / 31.0));
}
#endif
static void apply_bank_parameter(Instrument *ip, ToneBankElement *tone)
{
int i, j;
Sample *sp;
if (tone->tunenum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->tunenum == 1) {
sp->low_freq = adjust_tune_freq(sp->low_freq, tone->tune[0]);
sp->high_freq = adjust_tune_freq(sp->high_freq, tone->tune[0]);
sp->root_freq = adjust_tune_freq(sp->root_freq, tone->tune[0]);
} else if (i < tone->tunenum) {
sp->low_freq = adjust_tune_freq(sp->low_freq, tone->tune[i]);
sp->high_freq = adjust_tune_freq(sp->high_freq, tone->tune[i]);
sp->root_freq = adjust_tune_freq(sp->root_freq, tone->tune[i]);
}
}
if (tone->envratenum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->envratenum == 1) {
for (j = 0; j < 6; j++)
if (tone->envrate[0][j] >= 0)
sp->envelope_rate[j] = to_rate(tone->envrate[0][j]);
} else if (i < tone->envratenum) {
for (j = 0; j < 6; j++)
if (tone->envrate[i][j] >= 0)
sp->envelope_rate[j] = to_rate(tone->envrate[i][j]);
}
}
if (tone->envofsnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->envofsnum == 1) {
for (j = 0; j < 6; j++)
if (tone->envofs[0][j] >= 0)
sp->envelope_offset[j] = to_offset(tone->envofs[0][j]);
} else if (i < tone->envofsnum) {
for (j = 0; j < 6; j++)
if (tone->envofs[i][j] >= 0)
sp->envelope_offset[j] = to_offset(tone->envofs[i][j]);
}
}
if (tone->tremnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->tremnum == 1) {
if (IS_QUANTITY_DEFINED(tone->trem[0][0]))
sp->tremolo_sweep_increment =
quantity_to_int(&tone->trem[0][0], 0);
if (IS_QUANTITY_DEFINED(tone->trem[0][1]))
sp->tremolo_phase_increment =
quantity_to_int(&tone->trem[0][1], 0);
if (IS_QUANTITY_DEFINED(tone->trem[0][2]))
sp->tremolo_depth =
quantity_to_int(&tone->trem[0][2], 0) << 1;
} else if (i < tone->tremnum) {
if (IS_QUANTITY_DEFINED(tone->trem[i][0]))
sp->tremolo_sweep_increment =
quantity_to_int(&tone->trem[i][0], 0);
if (IS_QUANTITY_DEFINED(tone->trem[i][1]))
sp->tremolo_phase_increment =
quantity_to_int(&tone->trem[i][1], 0);
if (IS_QUANTITY_DEFINED(tone->trem[i][2]))
sp->tremolo_depth =
quantity_to_int(&tone->trem[i][2], 0) << 1;
}
}
if (tone->vibnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->vibnum == 1) {
if (IS_QUANTITY_DEFINED(tone->vib[0][1]))
sp->vibrato_control_ratio =
quantity_to_int(&tone->vib[0][1], 0);
if (IS_QUANTITY_DEFINED(tone->vib[0][0]))
sp->vibrato_sweep_increment =
quantity_to_int(&tone->vib[0][0],
sp->vibrato_control_ratio);
if (IS_QUANTITY_DEFINED(tone->vib[0][2]))
sp->vibrato_depth = quantity_to_int(&tone->vib[0][2], 0);
} else if (i < tone->vibnum) {
if (IS_QUANTITY_DEFINED(tone->vib[i][1]))
sp->vibrato_control_ratio =
quantity_to_int(&tone->vib[i][1], 0);
if (IS_QUANTITY_DEFINED(tone->vib[i][0]))
sp->vibrato_sweep_increment =
quantity_to_int(&tone->vib[i][0],
sp->vibrato_control_ratio);
if (IS_QUANTITY_DEFINED(tone->vib[i][2]))
sp->vibrato_depth = quantity_to_int(&tone->vib[i][2], 0);
}
}
if (tone->sclnotenum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->sclnotenum == 1)
sp->scale_freq = tone->sclnote[0];
else if (i < tone->sclnotenum)
sp->scale_freq = tone->sclnote[i];
}
if (tone->scltunenum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->scltunenum == 1)
sp->scale_factor = adjust_scale_tune(tone->scltune[0]);
else if (i < tone->scltunenum)
sp->scale_factor = adjust_scale_tune(tone->scltune[i]);
}
if (tone->modenvratenum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->modenvratenum == 1) {
for (j = 0; j < 6; j++)
if (tone->modenvrate[0][j] >= 0)
sp->modenv_rate[j] = to_rate(tone->modenvrate[0][j]);
} else if (i < tone->modenvratenum) {
for (j = 0; j < 6; j++)
if (tone->modenvrate[i][j] >= 0)
sp->modenv_rate[j] = to_rate(tone->modenvrate[i][j]);
}
}
if (tone->modenvofsnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->modenvofsnum == 1) {
for (j = 0; j < 6; j++)
if (tone->modenvofs[0][j] >= 0)
sp->modenv_offset[j] =
to_offset(tone->modenvofs[0][j]);
} else if (i < tone->modenvofsnum) {
for (j = 0; j < 6; j++)
if (tone->modenvofs[i][j] >= 0)
sp->modenv_offset[j] =
to_offset(tone->modenvofs[i][j]);
}
}
if (tone->envkeyfnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->envkeyfnum == 1) {
for (j = 0; j < 6; j++)
if (tone->envkeyf[0][j] != -1)
sp->envelope_keyf[j] = tone->envkeyf[0][j];
} else if (i < tone->envkeyfnum) {
for (j = 0; j < 6; j++)
if (tone->envkeyf[i][j] != -1)
sp->envelope_keyf[j] = tone->envkeyf[i][j];
}
}
if (tone->envvelfnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->envvelfnum == 1) {
for (j = 0; j < 6; j++)
if (tone->envvelf[0][j] != -1)
sp->envelope_velf[j] = tone->envvelf[0][j];
} else if (i < tone->envvelfnum) {
for (j = 0; j < 6; j++)
if (tone->envvelf[i][j] != -1)
sp->envelope_velf[j] = tone->envvelf[i][j];
}
}
if (tone->modenvkeyfnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->modenvkeyfnum == 1) {
for (j = 0; j < 6; j++)
if (tone->modenvkeyf[0][j] != -1)
sp->modenv_keyf[j] = tone->modenvkeyf[0][j];
} else if (i < tone->modenvkeyfnum) {
for (j = 0; j < 6; j++)
if (tone->modenvkeyf[i][j] != -1)
sp->modenv_keyf[j] = tone->modenvkeyf[i][j];
}
}
if (tone->modenvvelfnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->modenvvelfnum == 1) {
for (j = 0; j < 6; j++)
if (tone->modenvvelf[0][j] != -1)
sp->modenv_velf[j] = tone->modenvvelf[0][j];
} else if (i < tone->modenvvelfnum) {
for (j = 0; j < 6; j++)
if (tone->modenvvelf[i][j] != -1)
sp->modenv_velf[j] = tone->modenvvelf[i][j];
}
}
if (tone->trempitchnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->trempitchnum == 1)
sp->tremolo_to_pitch = tone->trempitch[0];
else if (i < tone->trempitchnum)
sp->tremolo_to_pitch = tone->trempitch[i];
}
if (tone->tremfcnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->tremfcnum == 1)
sp->tremolo_to_fc = tone->tremfc[0];
else if (i < tone->tremfcnum)
sp->tremolo_to_fc = tone->tremfc[i];
}
if (tone->modpitchnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->modpitchnum == 1)
sp->modenv_to_pitch = tone->modpitch[0];
else if (i < tone->modpitchnum)
sp->modenv_to_pitch = tone->modpitch[i];
}
if (tone->modfcnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->modfcnum == 1)
sp->modenv_to_fc = tone->modfc[0];
else if (i < tone->modfcnum)
sp->modenv_to_fc = tone->modfc[i];
}
if (tone->fcnum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->fcnum == 1)
sp->cutoff_freq = adjust_fc(tone->fc[0]);
else if (i < tone->fcnum)
sp->cutoff_freq = adjust_fc(tone->fc[i]);
}
if (tone->resonum)
for (i = 0; i < ip->samples; i++) {
sp = &ip->sample[i];
if (tone->resonum == 1)
sp->resonance = adjust_reso(tone->reso[0]);
else if (i < tone->resonum)
sp->resonance = adjust_reso(tone->reso[i]);
}
}
#define READ_CHAR(thing) { \
uint8 tmpchar; \
\
if (tf_read(&tmpchar, 1, 1, tf) != 1) \
goto fail; \
thing = tmpchar; \
}
#define READ_SHORT(thing) { \
uint16 tmpshort; \
\
if (tf_read(&tmpshort, 2, 1, tf) != 1) \
goto fail; \
thing = LE_SHORT(tmpshort); \
}
#define READ_LONG(thing) { \
int32 tmplong; \
\
if (tf_read(&tmplong, 4, 1, tf) != 1) \
goto fail; \
thing = LE_LONG(tmplong); \
}
/* If panning or note_to_use != -1, it will be used for all samples,
* instead of the sample-specific values in the instrument file.
*
* For note_to_use, any value < 0 or > 127 will be forced to 0.
*
* For other parameters, 1 means yes, 0 means no, other values are
* undefined.
*
* TODO: do reverse loops right
*/
static Instrument *load_gus_instrument(char *name,
ToneBank *bank, int dr, int prog, char *infomsg)
{
ToneBankElement *tone;
int amp, note_to_use, panning, strip_envelope, strip_loop, strip_tail;
Instrument *ip;
struct timidity_file *tf;
uint8 tmp[1024], fractions;
Sample *sp;
int i, j, noluck = 0;
if (! name)
return 0;
if (infomsg != NULL)
ctl->cmsg(CMSG_INFO, VERB_NOISY, "%s: %s", infomsg, name);
else
ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", name);
if (bank) {
tone = &bank->tone[prog];
amp = tone->amp;
note_to_use = (tone->note != -1) ? tone->note : ((dr) ? prog : -1);
panning = tone->pan;
strip_envelope = (tone->strip_envelope != -1)
? tone->strip_envelope : ((dr) ? 1 : -1);
strip_loop = (tone->strip_loop != -1)
? tone->strip_loop : ((dr) ? 1 : -1);
strip_tail = tone->strip_tail;
} else {
tone = NULL;
amp = note_to_use = panning = -1;
strip_envelope = strip_loop = strip_tail = 0;
}
if (tone && tone->tunenum == 0
&& tone->envratenum == 0 && tone->envofsnum == 0
&& tone->tremnum == 0 && tone->vibnum == 0
&& tone->sclnotenum == 0 && tone->scltunenum == 0
&& tone->modenvratenum == 0 && tone->modenvofsnum == 0
&& tone->envkeyfnum == 0 && tone->envvelfnum == 0
&& tone->modenvkeyfnum == 0 && tone->modenvvelfnum == 0
&& tone->trempitchnum == 0 && tone->tremfcnum == 0
&& tone->modpitchnum == 0 && tone->modfcnum == 0
&& tone->fcnum == 0 && tone->resonum == 0)
if ((ip = search_instrument_cache(name, panning, amp, note_to_use,
strip_loop, strip_envelope, strip_tail)) != NULL) {
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * Cached");
return ip;
}
/* Open patch file */
if (! (tf = open_file(name, 2, OF_NORMAL))) {
#ifdef PATCH_EXT_LIST
int name_len, ext_len;
static char *patch_ext[] = PATCH_EXT_LIST;
#endif
noluck = 1;
#ifdef PATCH_EXT_LIST
name_len = strlen(name);
/* Try with various extensions */
for (i = 0; patch_ext[i]; i++) {
ext_len = strlen(patch_ext[i]);
if (name_len + ext_len < 1024) {
if (name_len >= ext_len && strcmp(name + name_len - ext_len,
patch_ext[i]) == 0)
continue; /* duplicated ext. */
strcpy((char *) tmp, name);
strcat((char *) tmp, patch_ext[i]);
if ((tf = open_file((char *) tmp, 1, OF_NORMAL))) {
noluck = 0;
break;
}
}
}
#endif
}
if (noluck) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Instrument `%s' can't be found.", name);
return 0;
}
/* Read some headers and do cursory sanity checks. There are loads
* of magic offsets. This could be rewritten...
*/
tmp[0] = tf_getc(tf);
if (tmp[0] == '\0') {
/* for Mac binary */
skip(tf, 127);
tmp[0] = tf_getc(tf);
}
if ((tf_read(tmp + 1, 1, 238, tf) != 238)
|| (memcmp(tmp, "GF1PATCH110\0ID#000002", 22)
&& memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) {
/* don't know what the differences are */
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: not an instrument", name);
close_file(tf);
return 0;
}
/* instruments. To some patch makers, 0 means 1 */
if (tmp[82] != 1 && tmp[82] != 0) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle patches with %d instruments", tmp[82]);
close_file(tf);
return 0;
}
if (tmp[151] != 1 && tmp[151] != 0) { /* layers. What's a layer? */
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle instruments with %d layers", tmp[151]);
close_file(tf);
return 0;
}
ip = (Instrument *) safe_malloc(sizeof(Instrument));
ip->type = INST_GUS;
ip->samples = tmp[198];
ip->sample = (Sample *) safe_malloc(sizeof(Sample) * ip->samples);
memset(ip->sample, 0, sizeof(Sample) * ip->samples);
for (i = 0; i < ip->samples; i++) {
skip(tf, 7); /* Skip the wave name */
if (tf_read(&fractions, 1, 1, tf) != 1) {
fail:
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d", i);
for (j = 0; j < i; j++)
free(ip->sample[j].data);
free(ip->sample);
free(ip);
close_file(tf);
return 0;
}
sp = &(ip->sample[i]);
sp->low_vel = 0;
sp->high_vel = 127;
sp->cutoff_freq = sp->resonance = 0;
sp->tremolo_to_pitch = sp->tremolo_to_fc = 0;
sp->modenv_to_pitch = sp->modenv_to_fc = 0;
sp->vel_to_fc = sp->key_to_fc = sp->vel_to_resonance = 0;
sp->envelope_velf_bpo = sp->modenv_velf_bpo = 64;
sp->vel_to_fc_threshold = 64;
sp->key_to_fc_bpo = 60;
sp->envelope_delay = sp->modenv_delay = 0;
sp->tremolo_delay = sp->vibrato_delay = 0;
sp->inst_type = INST_GUS;
sp->sample_type = SF_SAMPLETYPE_MONO;
sp->sf_sample_link = -1;
sp->sf_sample_index = 0;
memset(sp->envelope_velf, 0, sizeof(sp->envelope_velf));
memset(sp->envelope_keyf, 0, sizeof(sp->envelope_keyf));
memset(sp->modenv_velf, 0, sizeof(sp->modenv_velf));
memset(sp->modenv_keyf, 0, sizeof(sp->modenv_keyf));
memset(sp->modenv_rate, 0, sizeof(sp->modenv_rate));
memset(sp->modenv_offset, 0, sizeof(sp->modenv_offset));
READ_LONG(sp->data_length);
READ_LONG(sp->loop_start);
READ_LONG(sp->loop_end);
READ_SHORT(sp->sample_rate);
READ_LONG(sp->low_freq);
READ_LONG(sp->high_freq);
READ_LONG(sp->root_freq);
skip(tf, 2); /* Why have a "root frequency" and then "tuning"?? */
READ_CHAR(tmp[0]);
ctl->cmsg(CMSG_INFO, VERB_DEBUG, "Rate/Low/Hi/Root = %d/%d/%d/%d",
sp->sample_rate, sp->low_freq, sp->high_freq, sp->root_freq);
if (panning == -1)
/* 0x07 and 0x08 are both center panning */
sp->panning = ((tmp[0] - ((tmp[0] < 8) ? 7 : 8)) * 63) / 7 + 64;
else
sp->panning = (uint8) (panning & 0x7f);
/* envelope, tremolo, and vibrato */
if (tf_read(tmp, 1, 18, tf) != 18)
goto fail;
if (! tmp[13] || ! tmp[14]) {
sp->tremolo_sweep_increment = sp->tremolo_phase_increment = 0;
sp->tremolo_depth = 0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no tremolo");
} else {
sp->tremolo_sweep_increment = convert_tremolo_sweep(tmp[12]);
sp->tremolo_phase_increment = convert_tremolo_rate(tmp[13]);
sp->tremolo_depth = tmp[14];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * tremolo: sweep %d, phase %d, depth %d",
sp->tremolo_sweep_increment, sp->tremolo_phase_increment,
sp->tremolo_depth);
}
if (! tmp[16] || ! tmp[17]) {
sp->vibrato_sweep_increment = sp->vibrato_control_ratio = 0;
sp->vibrato_depth = 0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no vibrato");
} else {
sp->vibrato_control_ratio = convert_vibrato_rate(tmp[16]);
sp->vibrato_sweep_increment = convert_vibrato_sweep(tmp[15],
sp->vibrato_control_ratio);
sp->vibrato_depth = tmp[17];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * vibrato: sweep %d, ctl %d, depth %d",
sp->vibrato_sweep_increment, sp->vibrato_control_ratio,
sp->vibrato_depth);
}
READ_CHAR(sp->modes);
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * mode: 0x%02x", sp->modes);
READ_SHORT(sp->scale_freq);
READ_SHORT(sp->scale_factor);
skip(tf, 36); /* skip reserved space */
/* Mark this as a fixed-pitch instrument if such a deed is desired. */
sp->note_to_use = (note_to_use != -1) ? (uint8) note_to_use : 0;
/* seashore.pat in the Midia patch set has no Sustain. I don't
* understand why, and fixing it by adding the Sustain flag to
* all looped patches probably breaks something else. We do it
* anyway.
*/
if (sp->modes & MODES_LOOPING)
sp->modes |= MODES_SUSTAIN;
/* Strip any loops and envelopes we're permitted to */
if ((strip_loop == 1) && (sp->modes & (MODES_SUSTAIN | MODES_LOOPING
| MODES_PINGPONG | MODES_REVERSE))) {
sp->modes &= ~(MODES_SUSTAIN | MODES_LOOPING
| MODES_PINGPONG | MODES_REVERSE);
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - Removing loop and/or sustain");
}
if (strip_envelope == 1) {
if (sp->modes & MODES_ENVELOPE)
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing envelope");
sp->modes &= ~MODES_ENVELOPE;
} else if (strip_envelope != 0) {
/* Have to make a guess. */
if (! (sp->modes & (MODES_LOOPING
| MODES_PINGPONG | MODES_REVERSE))) {
/* No loop? Then what's there to sustain?
* No envelope needed either...
*/
sp->modes &= ~(MODES_SUSTAIN|MODES_ENVELOPE);
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No loop, removing sustain and envelope");
} else if (! memcmp(tmp, "??????", 6) || tmp[11] >= 100) {
/* Envelope rates all maxed out?
* Envelope end at a high "offset"?
* That's a weird envelope. Take it out.
*/
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - Weirdness, removing envelope");
} else if (! (sp->modes & MODES_SUSTAIN)) {
/* No sustain? Then no envelope. I don't know if this is
* justified, but patches without sustain usually don't need
* the envelope either... at least the Gravis ones. They're
* mostly drums. I think.
*/
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No sustain, removing envelope");
}
}
for (j = 0; j < 6; j++) {
sp->envelope_rate[j]= convert_envelope_rate(tmp[j]);
sp->envelope_offset[j] = convert_envelope_offset(tmp[j + 6]);
}
/* this envelope seems to give reverb like effects to most patches
* use the same method as soundfont
*/
if (modify_release) {
sp->envelope_offset[3] = to_offset(5);
sp->envelope_rate[3] = calc_rate(255, modify_release);
sp->envelope_offset[4] = to_offset(4);
sp->envelope_rate[4] = to_offset(200);
sp->envelope_offset[5] = to_offset(4);
sp->envelope_rate[5] = to_offset(200);
}
/* Then read the sample data */
sp->data = (sample_t *) safe_malloc(sp->data_length + 4);
sp->data_alloced = 1;
if ((j = tf_read(sp->data, 1, sp->data_length, tf))
!= sp->data_length) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Too small this patch length: %d < %d",
j, sp->data_length);
goto fail;
}
if (! (sp->modes & MODES_16BIT)) { /* convert to 16-bit data */
int32 i;
uint16 *tmp;
uint8 *cp = (uint8 *) sp->data;
tmp = (uint16 *) safe_malloc(sp->data_length * 2 + 4);
for (i = 0; i < sp->data_length; i++)
tmp[i] = (uint16) cp[i] << 8;
sp->data = (sample_t *) tmp;
free(cp);
sp->data_length *= 2;
sp->loop_start *= 2;
sp->loop_end *= 2;
}
#ifndef LITTLE_ENDIAN
else { /* convert to machine byte order */
int32 i;
int16 *tmp = (int16 *) sp->data, s;
for (i = 0; i < sp->data_length / 2; i++)
s = LE_SHORT(tmp[i]), tmp[i] = s;
}
#endif
if (sp->modes & MODES_UNSIGNED) { /* convert to signed data */
int32 i = sp->data_length / 2;
int16 *tmp = (int16 *) sp->data;
while (i--)
*tmp++ ^= 0x8000;
}
/* Reverse loops and pass them off as normal loops */
if (sp->modes & MODES_REVERSE) {
/* The GUS apparently plays reverse loops by reversing the
* whole sample. We do the same because the GUS does not SUCK.
*/
int32 t;
reverse_data((int16 *) sp->data, 0, sp->data_length / 2);
t = sp->loop_start;
sp->loop_start = sp->data_length - sp->loop_end;
sp->loop_end = sp->data_length - t;
sp->modes &= ~MODES_REVERSE;
sp->modes |= MODES_LOOPING; /* just in case */
ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name);
}
/* If necessary do some anti-aliasing filtering */
if (antialiasing_allowed)
antialiasing((int16 *) sp->data, sp->data_length / 2,
sp->sample_rate, play_mode->rate);
#ifdef ADJUST_SAMPLE_VOLUMES
if (amp != -1)
sp->volume = (double) amp / 100;
else {
/* Try to determine a volume scaling factor for the sample.
* This is a very crude adjustment, but things sound more
* balanced with it. Still, this should be a runtime option.
*/
int32 i, a, maxamp = 0;
int16 *tmp = (int16 *) sp->data;
for (i = 0; i < sp->data_length / 2; i++)
if ((a = abs(tmp[i])) > maxamp)
maxamp = a;
sp->volume = 32768 / (double) maxamp;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * volume comp: %f", sp->volume);
}
#else
sp->volume = (amp != -1) ? (double) amp / 100 : 1.0;
#endif
/* These are in bytes. Convert into samples. */
sp->data_length /= 2;
sp->loop_start /= 2;
sp->loop_end /= 2;
/* The sample must be padded out by 2 extra sample, so that
* round off errors in the offsets used in interpolation will not
* cause a "pop" by reading random data beyond data_length
*/
sp->data[sp->data_length] = sp->data[sp->data_length + 1] = 0;
/* Remove abnormal loops which cause pop noise
* in long sustain stage
*/
if (! (sp->modes & MODES_LOOPING)) {
sp->loop_start = sp->data_length - 1;
sp->loop_end = sp->data_length;
sp->data[sp->data_length - 1] = 0;
}
/* Then fractional samples */
sp->data_length <<= FRACTION_BITS;
sp->loop_start <<= FRACTION_BITS;
sp->loop_end <<= FRACTION_BITS;
/* Adjust for fractional loop points. This is a guess. Does anyone
* know what "fractions" really stands for?
*/
sp->loop_start |= (fractions & 0x0f) << (FRACTION_BITS - 4);
sp->loop_end |= ((fractions >> 4) & 0x0f) << (FRACTION_BITS - 4);
/* If this instrument will always be played on the same note,
* and it's not looped, we can resample it now.
*/
if (sp->note_to_use && ! (sp->modes & MODES_LOOPING))
pre_resample(sp);
/* do pitch detection on drums if surround chorus is used */
if (dr && opt_surround_chorus)
{
sp->chord = -1;
sp->root_freq_detected = freq_fourier(sp, &(sp->chord));
sp->transpose_detected =
assign_pitch_to_freq(sp->root_freq_detected) -
assign_pitch_to_freq(sp->root_freq / 1024.0);
}
#ifdef LOOKUP_HACK
squash_sample_16to8(sp);
#endif
if (strip_tail == 1) {
/* Let's not really, just say we did. */
sp->data_length = sp->loop_end;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail");
}
}
close_file(tf);
store_instrument_cache(ip, name, panning, amp, note_to_use,
strip_loop, strip_envelope, strip_tail);
return ip;
}
#ifdef LOOKUP_HACK
/*! Squash the 16-bit data into 8 bits. */
void squash_sample_16to8(Sample *sp)
{
uint8 *gulp, *ulp;
int16 *swp;
int l = sp->data_length >> FRACTION_BITS;
gulp = ulp = (uint8 *)safe_malloc(l + 1);
swp = (int16 *)sp->data;
while (l--)
*ulp++ = (*swp++ >> 8) & 0xff;
free(sp->data);
sp->data = (sample_t *)gulp;
}
#endif
Instrument *load_instrument(int dr, int b, int prog)
{
ToneBank *bank = ((dr) ? drumset[b] : tonebank[b]);
Instrument *ip;
int i, font_bank, font_preset, font_keynote;
extern Instrument *extract_sample_file(char *);
FLOAT_T volume_max;
int pan, panning;
char infomsg[256];
#ifndef CFG_FOR_SF
if (play_system_mode == GS_SYSTEM_MODE && (b == 64 || b == 65))
if (! dr) /* User Instrument */
recompute_userinst(b, prog);
else /* User Drumset */
recompute_userdrum(b, prog);
#endif
if (bank->tone[prog].instype == 1 || bank->tone[prog].instype == 2) {
if (bank->tone[prog].instype == 1) { /* Font extention */
font_bank = bank->tone[prog].font_bank;
font_preset = bank->tone[prog].font_preset;
font_keynote = bank->tone[prog].font_keynote;
ip = extract_soundfont(bank->tone[prog].name,
font_bank, font_preset, font_keynote);
} else /* Sample extension */
ip = extract_sample_file(bank->tone[prog].name);
/* amp tuning */
if (ip != NULL && bank->tone[prog].amp != -1) {
for (i = 0, volume_max = 0; i < ip->samples; i++)
if (volume_max < ip->sample[i].volume)
volume_max = ip->sample[i].volume;
if (volume_max != 0)
for (i = 0; i < ip->samples; i++)
ip->sample[i].volume *= bank->tone[prog].amp
/ 100.0 / volume_max;
}
/* panning */
if (ip != NULL && bank->tone[prog].pan != -1) {
pan = ((int) bank->tone[prog].pan & 0x7f) - 64;
for (i = 0; i < ip->samples; i++) {
panning = (int) ip->sample[i].panning + pan;
panning = (panning < 0) ? 0
: ((panning > 127) ? 127 : panning);
ip->sample[i].panning = panning;
}
}
/* note to use */
if (ip != NULL && bank->tone[prog].note != -1)
for (i = 0; i < ip->samples; i++)
ip->sample[i].root_freq =
freq_table[bank->tone[prog].note & 0x7f];
/* filter key-follow */
if (ip != NULL && bank->tone[prog].key_to_fc != 0)
for (i = 0; i < ip->samples; i++)
ip->sample[i].key_to_fc = bank->tone[prog].key_to_fc;
/* filter velocity-follow */
if (ip != NULL && bank->tone[prog].vel_to_fc != 0)
for (i = 0; i < ip->samples; i++)
ip->sample[i].key_to_fc = bank->tone[prog].vel_to_fc;
/* resonance velocity-follow */
if (ip != NULL && bank->tone[prog].vel_to_resonance != 0)
for (i = 0; i < ip->samples; i++)
ip->sample[i].vel_to_resonance =
bank->tone[prog].vel_to_resonance;
/* strip tail */
if (ip != NULL && bank->tone[prog].strip_tail == 1)
for (i = 0; i < ip->samples; i++)
ip->sample[i].data_length = ip->sample[i].loop_end;
if (ip != NULL) {
i = (dr) ? 0 : prog;
if (bank->tone[i].comment)
free(bank->tone[i].comment);
bank->tone[i].comment = safe_strdup(ip->instname);
apply_bank_parameter(ip, &bank->tone[prog]);
}
return ip;
}
if (! dr) {
font_bank = b;
font_preset = prog;
font_keynote = -1;
} else {
font_bank = 128;
font_preset = b;
font_keynote = prog;
}
/* preload soundfont */
ip = load_soundfont_inst(0, font_bank, font_preset, font_keynote);
if (ip != NULL) {
if (bank->tone[prog].comment)
free(bank->tone[prog].comment);
bank->tone[prog].comment = safe_strdup(ip->instname);
}
if (ip == NULL) { /* load GUS/patch file */
if (! dr)
sprintf(infomsg, "Tonebank %d %d", b, prog + progbase);
else
sprintf(infomsg, "Drumset %d %d(%s)",
b + progbase, prog, note_name[prog % 12]);
ip = load_gus_instrument(bank->tone[prog].name,
bank, dr, prog, infomsg);
if (ip == NULL) { /* no patch; search soundfont again */
ip = load_soundfont_inst(1, font_bank, font_preset, font_keynote);
if (ip != NULL) {
if (bank->tone[0].comment)
free(bank->tone[0].comment);
bank->tone[0].comment = safe_strdup(ip->instname);
}
}
}
if (ip != NULL)
apply_bank_parameter(ip, &bank->tone[prog]);
return ip;
}
static int fill_bank(int dr, int b, int *rc)
{
int i, errors = 0;
ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
if(rc != NULL)
*rc = RC_NONE;
for(i = 0; i < 128; i++)
{
if(bank->tone[i].instrument == MAGIC_LOAD_INSTRUMENT)
{
if(!(bank->tone[i].name))
{
bank->tone[i].instrument = load_instrument(dr, b, i);
if(bank->tone[i].instrument == NULL)
{
ctl->cmsg(CMSG_WARNING,
(b != 0) ? VERB_VERBOSE : VERB_NORMAL,
"No instrument mapped to %s %d, program %d%s",
dr ? "drum set" : "tone bank",
dr ? b+progbase : b,
dr ? i : i+progbase,
(b != 0) ? "" :
" - this instrument will not be heard");
if(b != 0)
{
/* Mark the corresponding instrument in the default
bank / drumset for loading (if it isn't already) */
if(!dr)
{
if(!(standard_tonebank.tone[i].instrument))
standard_tonebank.tone[i].instrument =
MAGIC_LOAD_INSTRUMENT;
}
else
{
if(!(standard_drumset.tone[i].instrument))
standard_drumset.tone[i].instrument =
MAGIC_LOAD_INSTRUMENT;
}
bank->tone[i].instrument = 0;
}
else
bank->tone[i].instrument = MAGIC_ERROR_INSTRUMENT;
errors++;
}
}
else
{
if(rc != NULL)
{
*rc = check_apply_control();
if(RC_IS_SKIP_FILE(*rc))
return errors;
}
bank->tone[i].instrument = load_instrument(dr, b, i);
if(!bank->tone[i].instrument)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Couldn't load instrument %s "
"(%s %d, program %d)", bank->tone[i].name,
dr ? "drum set" : "tone bank",
dr ? b+progbase : b,
dr ? i : i+progbase);
errors++;
}
}
}
}
return errors;
}
int load_missing_instruments(int *rc)
{
int i = 128 + map_bank_counter, errors = 0;
if(rc != NULL)
*rc = RC_NONE;
while (i--)
{
if (tonebank[i])
errors+=fill_bank(0,i,rc);
if(rc != NULL && RC_IS_SKIP_FILE(*rc))
return errors;
if (drumset[i])
errors+=fill_bank(1,i,rc);
if(rc != NULL && RC_IS_SKIP_FILE(*rc))
return errors;
}
return errors;
}
static void *safe_memdup(void *s, size_t size)
{
return memcpy(safe_malloc(size), s, size);
}
/*! Copy ToneBankElement src to elm. The original elm is released. */
void copy_tone_bank_element(ToneBankElement *elm, const ToneBankElement *src)
{
int i;
free_tone_bank_element(elm);
memcpy(elm, src, sizeof(ToneBankElement));
if (elm->name)
elm->name = safe_strdup(elm->name);
if (elm->tunenum)
elm->tune = (float *) safe_memdup(elm->tune,
elm->tunenum * sizeof(float));
if (elm->envratenum) {
elm->envrate = (int **) safe_memdup(elm->envrate,
elm->envratenum * sizeof(int *));
for (i = 0; i < elm->envratenum; i++)
elm->envrate[i] = (int *) safe_memdup(elm->envrate[i],
6 * sizeof(int));
}
if (elm->envofsnum) {
elm->envofs = (int **) safe_memdup(elm->envofs,
elm->envofsnum * sizeof(int *));
for (i = 0; i < elm->envofsnum; i++)
elm->envofs[i] = (int *) safe_memdup(elm->envofs[i],
6 * sizeof(int));
}
if (elm->tremnum) {
elm->trem = (Quantity **) safe_memdup(elm->trem,
elm->tremnum * sizeof(Quantity *));
for (i = 0; i < elm->tremnum; i++)
elm->trem[i] = (Quantity *) safe_memdup(elm->trem[i],
3 * sizeof(Quantity));
}
if (elm->vibnum) {
elm->vib = (Quantity **) safe_memdup(elm->vib,
elm->vibnum * sizeof(Quantity *));
for (i = 0; i < elm->vibnum; i++)
elm->vib[i] = (Quantity *) safe_memdup(elm->vib[i],
3 * sizeof(Quantity));
}
if (elm->sclnotenum)
elm->sclnote = (int16 *) safe_memdup(elm->sclnote,
elm->sclnotenum * sizeof(int16));
if (elm->scltunenum)
elm->scltune = (int16 *) safe_memdup(elm->scltune,
elm->scltunenum * sizeof(int16));
if (elm->comment)
elm->comment = safe_strdup(elm->comment);
if (elm->modenvratenum) {
elm->modenvrate = (int **) safe_memdup(elm->modenvrate,
elm->modenvratenum * sizeof(int *));
for (i = 0; i < elm->modenvratenum; i++)
elm->modenvrate[i] = (int *) safe_memdup(elm->modenvrate[i],
6 * sizeof(int));
}
if (elm->modenvofsnum) {
elm->modenvofs = (int **) safe_memdup(elm->modenvofs,
elm->modenvofsnum * sizeof(int *));
for (i = 0; i < elm->modenvofsnum; i++)
elm->modenvofs[i] = (int *) safe_memdup(elm->modenvofs[i],
6 * sizeof(int));
}
if (elm->envkeyfnum) {
elm->envkeyf = (int **) safe_memdup(elm->envkeyf,
elm->envkeyfnum * sizeof(int *));
for (i = 0; i < elm->envkeyfnum; i++)
elm->envkeyf[i] = (int *) safe_memdup(elm->envkeyf[i],
6 * sizeof(int));
}
if (elm->envvelfnum) {
elm->envvelf = (int **) safe_memdup(elm->envvelf,
elm->envvelfnum * sizeof(int *));
for (i = 0; i < elm->envvelfnum; i++)
elm->envvelf[i] = (int *) safe_memdup(elm->envvelf[i],
6 * sizeof(int));
}
if (elm->modenvkeyfnum) {
elm->modenvkeyf = (int **) safe_memdup(elm->modenvkeyf,
elm->modenvkeyfnum * sizeof(int *));
for (i = 0; i < elm->modenvkeyfnum; i++)
elm->modenvkeyf[i] = (int *) safe_memdup(elm->modenvkeyf[i],
6 * sizeof(int));
}
if (elm->modenvvelfnum) {
elm->modenvvelf = (int **) safe_memdup(elm->modenvvelf,
elm->modenvvelfnum * sizeof(int *));
for (i = 0; i < elm->modenvvelfnum; i++)
elm->modenvvelf[i] = (int *) safe_memdup(elm->modenvvelf[i],
6 * sizeof(int));
}
if (elm->trempitchnum)
elm->trempitch = (int16 *) safe_memdup(elm->trempitch,
elm->trempitchnum * sizeof(int16));
if (elm->tremfcnum)
elm->tremfc = (int16 *) safe_memdup(elm->tremfc,
elm->tremfcnum * sizeof(int16));
if (elm->modpitchnum)
elm->modpitch = (int16 *) safe_memdup(elm->modpitch,
elm->modpitchnum * sizeof(int16));
if (elm->modfcnum)
elm->modfc = (int16 *) safe_memdup(elm->modfc,
elm->modfcnum * sizeof(int16));
if (elm->fcnum)
elm->fc = (int16 *) safe_memdup(elm->fc,
elm->fcnum * sizeof(int16));
if (elm->resonum)
elm->reso = (int16 *) safe_memdup(elm->reso,
elm->resonum * sizeof(int16));
}
/*! Release ToneBank[128 + MAP_BANK_COUNT] */
static void free_tone_bank_list(ToneBank *tb[])
{
int i, j;
ToneBank *bank;
for (i = 0; i < 128 + map_bank_counter; i++)
{
bank = tb[i];
if (!bank)
continue;
for (j = 0; j < 128; j++)
free_tone_bank_element(&bank->tone[j]);
if (i > 0)
{
free(bank);
tb[i] = NULL;
}
}
}
/*! Release tonebank and drumset */
void free_tone_bank(void)
{
free_tone_bank_list(tonebank);
free_tone_bank_list(drumset);
}
/*! Release ToneBankElement. */
void free_tone_bank_element(ToneBankElement *elm)
{
elm->instype = 0;
if (elm->name)
free(elm->name);
elm->name = NULL;
if (elm->tune)
free(elm->tune);
elm->tune = NULL, elm->tunenum = 0;
if (elm->envratenum)
free_ptr_list(elm->envrate, elm->envratenum);
elm->envrate = NULL, elm->envratenum = 0;
if (elm->envofsnum)
free_ptr_list(elm->envofs, elm->envofsnum);
elm->envofs = NULL, elm->envofsnum = 0;
if (elm->tremnum)
free_ptr_list(elm->trem, elm->tremnum);
elm->trem = NULL, elm->tremnum = 0;
if (elm->vibnum)
free_ptr_list(elm->vib, elm->vibnum);
elm->vib = NULL, elm->vibnum = 0;
if (elm->sclnote)
free(elm->sclnote);
elm->sclnote = NULL, elm->sclnotenum = 0;
if (elm->scltune)
free(elm->scltune);
elm->scltune = NULL, elm->scltunenum = 0;
if (elm->comment)
free(elm->comment);
elm->comment = NULL;
if (elm->modenvratenum)
free_ptr_list(elm->modenvrate, elm->modenvratenum);
elm->modenvrate = NULL, elm->modenvratenum = 0;
if (elm->modenvofsnum)
free_ptr_list(elm->modenvofs, elm->modenvofsnum);
elm->modenvofs = NULL, elm->modenvofsnum = 0;
if (elm->envkeyfnum)
free_ptr_list(elm->envkeyf, elm->envkeyfnum);
elm->envkeyf = NULL, elm->envkeyfnum = 0;
if (elm->envvelfnum)
free_ptr_list(elm->envvelf, elm->envvelfnum);
elm->envvelf = NULL, elm->envvelfnum = 0;
if (elm->modenvkeyfnum)
free_ptr_list(elm->modenvkeyf, elm->modenvkeyfnum);
elm->modenvkeyf = NULL, elm->modenvkeyfnum = 0;
if (elm->modenvvelfnum)
free_ptr_list(elm->modenvvelf, elm->modenvvelfnum);
elm->modenvvelf = NULL, elm->modenvvelfnum = 0;
if (elm->trempitch)
free(elm->trempitch);
elm->trempitch = NULL, elm->trempitchnum = 0;
if (elm->tremfc)
free(elm->tremfc);
elm->tremfc = NULL, elm->tremfcnum = 0;
if (elm->modpitch)
free(elm->modpitch);
elm->modpitch = NULL, elm->modpitchnum = 0;
if (elm->modfc)
free(elm->modfc);
elm->modfc = NULL, elm->modfcnum = 0;
if (elm->fc)
free(elm->fc);
elm->fc = NULL, elm->fcnum = 0;
if (elm->reso)
free(elm->reso);
elm->reso = NULL, elm->resonum = 0;
}
void free_instruments(int reload_default_inst)
{
int i = 128 + map_bank_counter, j;
struct InstrumentCache *p;
ToneBank *bank;
Instrument *ip;
struct InstrumentCache *default_entry;
int default_entry_addr;
clear_magic_instruments();
/* Free soundfont instruments */
while(i--)
{
/* Note that bank[*]->tone[j].instrument may pointer to
bank[0]->tone[j].instrument. See play_midi_load_instrument()
at playmidi.c for the implementation */
if((bank = tonebank[i]) != NULL)
for(j = 127; j >= 0; j--)
{
ip = bank->tone[j].instrument;
if(ip != NULL && ip->type == INST_SF2 &&
(i == 0 || ip != tonebank[0]->tone[j].instrument))
free_instrument(ip);
bank->tone[j].instrument = NULL;
}
if((bank = drumset[i]) != NULL)
for(j = 127; j >= 0; j--)
{
ip = bank->tone[j].instrument;
if(ip != NULL && ip->type == INST_SF2 &&
(i == 0 || ip != drumset[0]->tone[j].instrument))
free_instrument(ip);
bank->tone[j].instrument = NULL;
}
}
/* Free GUS/patch instruments */
default_entry = NULL;
default_entry_addr = 0;
for(i = 0; i < INSTRUMENT_HASH_SIZE; i++)
{
p = instrument_cache[i];
while(p != NULL)
{
if(!reload_default_inst && p->ip == default_instrument)
{
default_entry = p;
default_entry_addr = i;
p = p->next;
}
else
{
struct InstrumentCache *tmp;
tmp = p;
p = p->next;
free_instrument(tmp->ip);
free(tmp);
}
}
instrument_cache[i] = NULL;
}
if(reload_default_inst)
set_default_instrument(NULL);
else if(default_entry)
{
default_entry->next = NULL;
instrument_cache[default_entry_addr] = default_entry;
}
}
void free_special_patch(int id)
{
int i, j, start, end;
if(id >= 0)
start = end = id;
else
{
start = 0;
end = NSPECIAL_PATCH - 1;
}
for(i = start; i <= end; i++)
if(special_patch[i] != NULL)
{
Sample *sp;
int n;
if(special_patch[i]->name != NULL)
free(special_patch[i]->name);
special_patch[i]->name = NULL;
n = special_patch[i]->samples;
sp = special_patch[i]->sample;
if(sp)
{
for(j = 0; j < n; j++)
if(sp[j].data_alloced && sp[j].data)
free(sp[j].data);
free(sp);
}
free(special_patch[i]);
special_patch[i] = NULL;
}
}
int set_default_instrument(char *name)
{
Instrument *ip;
int i;
static char *last_name;
if(name == NULL)
{
name = last_name;
if(name == NULL)
return 0;
}
if(!(ip = load_gus_instrument(name, NULL, 0, 0, NULL)))
return -1;
if(default_instrument)
free_instrument(default_instrument);
default_instrument = ip;
for(i = 0; i < MAX_CHANNELS; i++)
default_program[i] = SPECIAL_PROGRAM;
last_name = name;
return 0;
}
/*! search mapped bank.
returns negative value indicating free bank if not found,
0 if no free bank was available */
int find_instrument_map_bank(int dr, int map, int bk)
{
struct bank_map_elem *bm;
int i;
if (map == INST_NO_MAP)
return 0;
bm = dr ? map_drumset : map_bank;
for(i = 0; i < MAP_BANK_COUNT; i++)
{
if (!bm[i].used)
return -(128 + i);
else if (bm[i].mapid == map && bm[i].bankno == bk)
return 128 + i;
}
return 0;
}
/*! allocate mapped bank if needed. returns -1 if allocation failed. */
int alloc_instrument_map_bank(int dr, int map, int bk)
{
struct bank_map_elem *bm;
int i;
if (map == INST_NO_MAP)
{
alloc_instrument_bank(dr, bk);
return bk;
}
i = find_instrument_map_bank(dr, map, bk);
if (i == 0)
return -1;
if (i < 0)
{
i = -i - 128;
bm = dr ? map_drumset : map_bank;
bm[i].used = 1;
bm[i].mapid = map;
bm[i].bankno = bk;
if (map_bank_counter < i + 1)
map_bank_counter = i + 1;
i += 128;
alloc_instrument_bank(dr, i);
}
return i;
}
void alloc_instrument_bank(int dr, int bk)
{
ToneBank *b;
if(dr)
{
if((b = drumset[bk]) == NULL)
{
b = drumset[bk] = (ToneBank *)safe_malloc(sizeof(ToneBank));
memset(b, 0, sizeof(ToneBank));
}
}
else
{
if((b = tonebank[bk]) == NULL)
{
b = tonebank[bk] = (ToneBank *)safe_malloc(sizeof(ToneBank));
memset(b, 0, sizeof(ToneBank));
}
}
}
/* Instrument alias map - Written by Masanao Izumo */
int instrument_map(int mapID, int *set, int *elem)
{
int s, e;
struct inst_map_elem *p;
if(mapID == INST_NO_MAP)
return 0; /* No map */
s = *set;
e = *elem;
p = inst_map_table[mapID][s];
if(p != NULL && p[e].mapped)
{
*set = p[e].set;
*elem = p[e].elem;
return 1;
}
if(s != 0)
{
p = inst_map_table[mapID][0];
if(p != NULL && p[e].mapped)
{
*set = p[e].set;
*elem = p[e].elem;
}
return 2;
}
return 0;
}
void set_instrument_map(int mapID,
int set_from, int elem_from,
int set_to, int elem_to)
{
struct inst_map_elem *p;
p = inst_map_table[mapID][set_from];
if(p == NULL)
{
p = (struct inst_map_elem *)
safe_malloc(128 * sizeof(struct inst_map_elem));
memset(p, 0, 128 * sizeof(struct inst_map_elem));
inst_map_table[mapID][set_from] = p;
}
p[elem_from].set = set_to;
p[elem_from].elem = elem_to;
p[elem_from].mapped = 1;
}
void free_instrument_map(void)
{
int i, j;
for(i = 0; i < map_bank_counter; i++)
map_bank[i].used = map_drumset[i].used = 0;
/* map_bank_counter = 0; never shrinks rather than assuming tonebank was already freed */
for (i = 0; i < NUM_INST_MAP; i++) {
for (j = 0; j < 128; j++) {
struct inst_map_elem *map;
map = inst_map_table[i][j];
if (map) {
free(map);
inst_map_table[i][j] = NULL;
}
}
}
}
/* Alternate assign - Written by Masanao Izumo */
AlternateAssign *add_altassign_string(AlternateAssign *old,
char **params, int n)
{
int i, j;
char *p;
int beg, end;
AlternateAssign *alt;
if(n == 0)
return old;
if(!strcmp(*params, "clear")) {
while(old) {
AlternateAssign *next;
next = old->next;
free(old);
old = next;
}
params++;
n--;
if(n == 0)
return NULL;
}
alt = (AlternateAssign *)safe_malloc(sizeof(AlternateAssign));
memset(alt, 0, sizeof(AlternateAssign));
for(i = 0; i < n; i++) {
p = params[i];
if(*p == '-') {
beg = 0;
p++;
} else
beg = atoi(p);
if((p = strchr(p, '-')) != NULL) {
if(p[1] == '\0')
end = 127;
else
end = atoi(p + 1);
} else
end = beg;
if(beg > end) {
int t;
t = beg;
beg = end;
end = t;
}
if(beg < 0)
beg = 0;
if(end > 127)
end = 127;
for(j = beg; j <= end; j++)
alt->bits[(j >> 5) & 0x3] |= 1 << (j & 0x1F);
}
alt->next = old;
return alt;
}
AlternateAssign *find_altassign(AlternateAssign *altassign, int note)
{
AlternateAssign *p;
uint32 mask;
int idx;
mask = 1 << (note & 0x1F);
idx = (note >> 5) & 0x3;
for(p = altassign; p != NULL; p = p->next)
if(p->bits[idx] & mask)
return p;
return NULL;
}