/*
    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_r(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_r((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 */
			ip = recompute_userdrum(b, prog);
			if (ip != NULL) {
				return ip;
			}
		}
        }
#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].name == NULL) /* this should not be NULL to play the instrument */
			bank->tone[prog].name = safe_strdup(DYNAMIC_INSTRUMENT_NAME);
		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 (bank->alt)
			free (bank->alt);
		bank->alt = NULL;

		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->tone[j].name && !bank->tone[j].name[0]) /* DYNAMIC_INSTRUMENT_NAME */
		{
			free(bank->tone[j].name);
			bank->tone[j].name = 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;
		if(bank->tone[j].name && !bank->tone[j].name[0]) /* DYNAMIC_INSTRUMENT_NAME */
		{
			free(bank->tone[j].name);
			bank->tone[j].name = NULL;
		}
	    }
#if 0
		if ((drumset[i] != NULL) && (drumset[i]->alt != NULL)) {
			free(drumset[i]->alt);
			drumset[i]->alt = NULL;
		}
#endif
    }

    /* 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;
}
