gzdoom-gles/libraries/timidityplus/instrum.cpp

/*
    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.
*/

#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <string.h>

#include "timidity.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "resample.h"
#include "tables.h"
#include "filter.h"
#include "quantity.h"
#include "freq.h"

namespace TimidityPlus
{


Instruments::Instruments()
{
	// init one-time global stuff - this should go to the device class once it exists.
	initialize_resampler_coeffs();
	init_tables();

	memset(&standard_tonebank, 0, sizeof(standard_tonebank));
	memset(&standard_drumset, 0, sizeof(standard_drumset));
	memcpy(layer_items, static_layer_items, sizeof(layer_items));
}

bool Instruments::load(SoundFontReaderInterface *sf)
{
	sfreader = sf;
	if (read_config_file(nullptr, 0, 0) == RC_OK)
	{
		init_load_soundfont();
		set_default_instrument();
		return true;
	}
	return false;
}

Instruments::~Instruments()
{
	free_instruments(0);
	free_soundfonts();

	free_tone_bank();
	free_instrument_map();

	if (sfreader != nullptr) delete sfreader;
}

void Instruments::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);
}



/* calculate ramp rate in fractional unit;
 * diff = 8bit, time = msec
 */
int32_t Instruments::calc_rate_i(int diff, double msec)
{
    double rate;

    if(msec < 6)
	msec = 6;
    if(diff == 0)
	diff = 255;
    diff <<= (7+15);
    rate = ((double)diff / playback_rate) * control_ratio * 1000.0 / msec;
    if(fast_decay)
		rate *= 2;
    return (int32_t)rate;
}
/*End of Pseudo Reverb*/


void Instruments::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;
		}
	}
}


int32_t Instruments::convert_envelope_rate(uint8_t rate)
{
	const int32_t GUS_ENVRATE_MAX = (int32_t)(0x3FFFFFFF >> 9);

	int32_t r;

	r = 3 - ((rate >> 6) & 0x3);
	r *= 3;
	r = (int32_t)(rate & 0x3f) << r; /* 6.9 fixed point */

	/* 15.15 fixed point. */
	r = r * 44100 / playback_rate * control_ratio * (1 << fast_decay);
	if (r > GUS_ENVRATE_MAX) { r = GUS_ENVRATE_MAX; }
	return (r << 9);
}

int32_t Instruments::convert_envelope_offset(uint8_t 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);
}

int32_t Instruments::convert_tremolo_sweep(uint8_t sweep)
{
	if (!sweep)
		return 0;

	return
		((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
		(playback_rate * sweep);
}

int32_t Instruments::convert_vibrato_sweep(uint8_t sweep, int32_t vib_control_ratio)
{
  if (!sweep)
    return 0;

  return (int32_t)(TIM_FSCALE((double) (vib_control_ratio)
			    * SWEEP_TUNING, SWEEP_SHIFT)
		 / (double)(playback_rate * sweep));

  /* this was overflowing with seashore.pat

      ((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
      (playback_rate * sweep); */
}

int32_t Instruments::convert_tremolo_rate(uint8_t rate)
{
	return
		((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) /
		(TREMOLO_RATE_TUNING * playback_rate);
}

int32_t Instruments::convert_vibrato_rate(uint8_t rate)
{
	/* Return a suitable vibrato_control_ratio value */
	return
		(VIBRATO_RATE_TUNING * playback_rate) /
		(rate * 2 * VIBRATO_SAMPLE_INCREMENTS);
}

void Instruments::reverse_data(int16_t *sp, int32_t ls, int32_t le)
{
	int16_t s, *ep = sp + le;
	int32_t i;
	sp += ls;
	le -= ls;
	le /= 2;
	for (i = 0; i < le; i++)
	{
		s = *sp;
		*sp++ = *ep;
		*ep-- = s;
	}
}

int Instruments::name_hash(char *name)
{
    unsigned int addr = 0;

    while(*name)
	addr += *name++;
    return addr % INSTRUMENT_HASH_SIZE;
}

Instrument *Instruments::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;
}

void Instruments::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_t adjust_tune_freq(int32_t val, float tune)
{
	if (! tune)
		return val;
	return val / pow(2.0, tune / 12.0);
}

static int16_t adjust_scale_tune(int16_t val)
{
	return 1024 * (double) val / 100 + 0.5;
}

static int16_t adjust_fc(int16_t val)
{
	if (val < 0 || val > playback_rate / 2) {
		return 0;
	} else {
		return val;
	}
}

static int16_t adjust_reso(int16_t val)
{
	if (val < 0 || val > 960) {
		return 0;
	} else {
		return val;
	}
}

int32_t Instruments::to_rate(int rate)
{
	return (rate) ? (int32_t) (0x200 * pow(2.0, rate / 17.0)
			* 44100 / playback_rate * control_ratio) << fast_decay : 0;
}


void Instruments::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_22(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_22(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_22(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_22(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_t tmpchar; \
		\
		if (tf_read(&tmpchar, 1, 1, tf) != 1) \
			goto fail; \
		thing = tmpchar; \
}
#define READ_SHORT(thing) { \
		uint16_t tmpshort; \
		\
		if (tf_read(&tmpshort, 2, 1, tf) != 1) \
			goto fail; \
		thing = LE_SHORT(tmpshort); \
}
#define READ_LONG(thing) { \
		int32_t 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
 */
Instrument *Instruments::load_gus_instrument(char *name, ToneBank *bank, int dr, int prog)
{
	ToneBankElement *tone;
	int amp, note_to_use, panning, strip_envelope, strip_loop, strip_tail;
	Instrument *ip;
	struct timidity_file *tf;
	uint8_t tmp[1024], fractions;
	Sample *sp;
	int i, j, noluck = 0;

	if (!name)
		return 0;

	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 */
	tf = open_file(name, sfreader);
	if (!tf) 
	{
		int name_len, ext_len;
		static const char *patch_ext[] = { ".pat", 0 };

		noluck = 1;
		name_len = (int)strlen(name);
		/* Try with various extensions */
		for (i = 0; patch_ext[i]; i++) 
		{
			ext_len = (int)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]);
				tf = open_file((char *)tmp, sfreader);
				if (tf)
				{
					noluck = 0;
					break;
				}
			}
		}
	}
	if (noluck) 
	{
		ctl_cmsg(CMSG_INFO, VERB_DEBUG, "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);
		tf_close(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]);
		tf_close(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]);
		tf_close(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);
			tf_close(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_t)(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_t)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_22(5);
			sp->envelope_rate[3] = calc_rate_i(255, modify_release);
			sp->envelope_offset[4] = to_offset_22(4);
			sp->envelope_rate[4] = to_offset_22(200);
			sp->envelope_offset[5] = to_offset_22(4);
			sp->envelope_rate[5] = to_offset_22(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))	!= (int)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 */
			uint16_t *tmp;
			uint8_t *cp = (uint8_t *)sp->data;

			tmp = (uint16_t *)safe_malloc(sp->data_length * 2 + 4);
			for (splen_t i = 0; i < sp->data_length; i++)
				tmp[i] = (uint16_t)cp[i] << 8;
			sp->data = (sample_t *)tmp;
			free(cp);
			sp->data_length *= 2;
			sp->loop_start *= 2;
			sp->loop_end *= 2;
		}
#ifdef _BIG_ENDIAN_
		else {	/* convert to machine byte order */
			int32_t i;
			int16_t *tmp = (int16_t *)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_t i = sp->data_length / 2;
			int16_t *tmp = (int16_t *)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 t;

			reverse_data((int16_t *)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_t *)sp->data, sp->data_length / 2,
				sp->sample_rate, playback_rate);
		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_t a, maxamp = 0;
			int16_t *tmp = (int16_t *)sp->data;

			for (splen_t 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);
		}
		/* 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 && timidity_surround_chorus)
		{
			Freq freq;
			sp->chord = -1;
			sp->root_freq_detected = freq.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);
		}

		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");
		}
	}
	tf_close(tf);
	store_instrument_cache(ip, name, panning, amp, note_to_use,
		strip_loop, strip_envelope, strip_tail);
	return ip;
}


Instrument *Instruments::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;
	double volume_max;
	int pan, panning;

#if 0
	// This cannot possibly work as implemented.
	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 */
		ip = load_gus_instrument(bank->tone[prog].name, bank, dr, prog);
		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;
}

int Instruments::fill_bank(int dr, int b, int *rc)
{
	int i, errors = 0;
	ToneBank *bank = ((dr) ? drumset[b] : tonebank[b]);

	if (rc != NULL)
		*rc = RC_OK;

	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)
				{
					// This would be too annoying on 'warning' level.
					ctl_cmsg(CMSG_WARNING, VERB_DEBUG,
						"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 = RC_OK;
				}

				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 Instruments::load_missing_instruments(int *rc)
{
	int i = 128 + map_bank_counter, errors = 0;
	if (rc != NULL)
		*rc = RC_OK;
	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;
}

// The precaching code is from ZDoom's Timidity-based GUS emulation
void Instruments::MarkInstrument(int banknum, int percussion, int instr)
{
	ToneBank *bank;

	if (banknum >= 128)
	{
		return;
	}
	if (banknum != 0)
	{
		/* Mark the standard bank in case it's not defined by this one. */
		MarkInstrument(0, percussion, instr);
	}
	if (percussion)
	{
		bank = drumset[banknum];
	}
	else
	{
		bank = tonebank[banknum];
	}
	if (bank == NULL)
	{
		return;
	}
	if (bank->tone[instr].instrument == NULL)
	{
		bank->tone[instr].instrument = MAGIC_LOAD_INSTRUMENT;
	}
}

void Instruments::PrecacheInstruments(const uint16_t *instruments, int count)
{
	for (int i = 0; i < count; ++i)
	{
		MarkInstrument((instruments[i] >> 7) & 127, instruments[i] >> 14, instruments[i] & 127);
	}
	load_missing_instruments(nullptr);
}



void *Instruments::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 Instruments::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_t *)safe_memdup(elm->sclnote,
			elm->sclnotenum * sizeof(int16_t));
	if (elm->scltunenum)
		elm->scltune = (int16_t *)safe_memdup(elm->scltune,
			elm->scltunenum * sizeof(int16_t));
	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_t *)safe_memdup(elm->trempitch,
			elm->trempitchnum * sizeof(int16_t));
	if (elm->tremfcnum)
		elm->tremfc = (int16_t *)safe_memdup(elm->tremfc,
			elm->tremfcnum * sizeof(int16_t));
	if (elm->modpitchnum)
		elm->modpitch = (int16_t *)safe_memdup(elm->modpitch,
			elm->modpitchnum * sizeof(int16_t));
	if (elm->modfcnum)
		elm->modfc = (int16_t *)safe_memdup(elm->modfc,
			elm->modfcnum * sizeof(int16_t));
	if (elm->fcnum)
		elm->fc = (int16_t *)safe_memdup(elm->fc,
			elm->fcnum * sizeof(int16_t));
	if (elm->resonum)
		elm->reso = (int16_t *)safe_memdup(elm->reso,
			elm->resonum * sizeof(int16_t));

}

/*! Release ToneBank[128 + MAP_BANK_COUNT] */
void Instruments::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 Instruments::free_tone_bank(void)
{
	free_tone_bank_list(tonebank);
	free_tone_bank_list(drumset);
}

/*! Release ToneBankElement. */
void Instruments::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 Instruments::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 Instruments::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 Instruments::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)))
		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 Instruments::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 Instruments::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 Instruments::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 Instruments::instrument_map(int mapID, int *set, int *elem) const
{
	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 Instruments::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 Instruments::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 *Instruments::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 *Instruments::find_altassign(AlternateAssign *altassign, int note)
{
	AlternateAssign *p;
	uint32_t 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;
}

Instrument *Instruments::play_midi_load_instrument(int dr, int bk, int prog, bool *pLoad_success)
{
	ToneBank **bank = (dr) ? drumset : tonebank;
	ToneBankElement *tone;
	Instrument *ip;
	bool load_success = false;

	if (bank[bk] == NULL)
		alloc_instrument_bank(dr, bk);

	tone = &bank[bk]->tone[prog];
	/* tone->name is NULL if "soundfont" directive is used, and ip is NULL when not preloaded */
	/* dr: not sure but only drumsets are concerned at the moment */
	if (dr && !tone->name && ((ip = tone->instrument) == MAGIC_LOAD_INSTRUMENT || ip == NULL)
		&& (ip = load_instrument(dr, bk, prog)) != NULL) {
		tone->instrument = ip;
		tone->name = safe_strdup(DYNAMIC_INSTRUMENT_NAME);
		load_success = 1;
	}
	else if (tone->name) {
		/* Instrument is found. */
		if ((ip = tone->instrument) == MAGIC_LOAD_INSTRUMENT
#ifndef SUPPRESS_CHANNEL_LAYER
			|| ip == NULL	/* see also readmidi.c: groom_list(). */
#endif
			) {
			ip = tone->instrument = load_instrument(dr, bk, prog);
		}
		if (ip == NULL || IS_MAGIC_INSTRUMENT(ip)) {
			tone->instrument = MAGIC_ERROR_INSTRUMENT;
		}
		else {
			load_success = true;
		}
	}
	else {
		/* Instrument is not found.
		Try to load the instrument from bank 0 */
		ToneBankElement *tone0 = &bank[0]->tone[prog];
		if ((ip = tone0->instrument) == NULL
			|| ip == MAGIC_LOAD_INSTRUMENT)
			ip = tone0->instrument = load_instrument(dr, 0, prog);
		if (ip == NULL || IS_MAGIC_INSTRUMENT(ip)) {
			tone0->instrument = MAGIC_ERROR_INSTRUMENT;
		}
		else {
			copy_tone_bank_element(tone, tone0);
			tone->instrument = ip;
			load_success = 1;
		}
	}

	*pLoad_success = load_success;

	if (ip == MAGIC_ERROR_INSTRUMENT)
		return NULL;

	return ip;
}



//void recompute_userinst_altassign(int bank, int group);


/*! initialize GS user drumset. */
void Instruments::init_userdrum()
{
	int i;

	free_userdrum();

	for (i = 0; i<2; i++) {	/* allocate alternative assign */
		memset(&alt[i], 0, sizeof(AlternateAssign));
		alloc_instrument_bank(1, 64 + i);
		drumset[64 + i]->alt = &alt[i];
	}
}


/*! free GS user drumset. */
void Instruments::free_userdrum()
{
	UserDrumset *p, *next;

	for (p = userdrum_first; p != NULL; p = next) {
		next = p->next;
		free(p);
	}
	userdrum_first = userdrum_last = NULL;
}

/*! free GS user instrument. */
void Instruments::free_userinst()
{
	UserInstrument *p, *next;

	for (p = userinst_first; p != NULL; p = next) {
		next = p->next;
		free(p);
	}
	userinst_first = userinst_last = NULL;
}




/*! recompute GS user instrument. */
/*! get pointer to requested GS user instrument.
if it's not found, allocate a new item first. */
Instruments::UserInstrument *Instruments::get_userinst(int bank, int prog)
{
	UserInstrument *p;

	for (p = userinst_first; p != NULL; p = p->next) {
		if (p->bank == bank && p->prog == prog) { return p; }
	}

	p = (UserInstrument *)safe_malloc(sizeof(UserInstrument));
	memset(p, 0, sizeof(UserInstrument));
	p->next = NULL;
	if (userinst_first == NULL) {
		userinst_first = p;
		userinst_last = p;
	}
	else {
		userinst_last->next = p;
		userinst_last = p;
	}
	p->bank = bank;
	p->prog = prog;

	return p;
}



void Instruments::recompute_userinst(int bank, int prog)
{
	auto p = get_userinst(bank, prog);
	auto source_bank = p->source_bank;
	auto source_prog = p->source_prog;
	free_tone_bank_element(&tonebank[bank]->tone[prog]);
	if (tonebank[source_bank]) {
		if (tonebank[source_bank]->tone[source_prog].name) 
		{
			copy_tone_bank_element(&tonebank[bank]->tone[prog], &tonebank[source_bank]->tone[source_prog]);
		}
		else if (tonebank[0]->tone[source_prog].name) 
		{
			copy_tone_bank_element(&tonebank[bank]->tone[prog], &tonebank[0]->tone[source_prog]);
		}
	}
}

/*! get pointer to requested GS user drumset.
if it's not found, allocate a new item first. */
Instruments::UserDrumset *Instruments::get_userdrum(int bank, int prog)
{
	UserDrumset *p;

	for (p = userdrum_first; p != NULL; p = p->next) {
		if (p->bank == bank && p->prog == prog) { return p; }
	}

	p = (UserDrumset *)safe_malloc(sizeof(UserDrumset));
	memset(p, 0, sizeof(UserDrumset));
	p->next = NULL;
	if (userdrum_first == NULL) {
		userdrum_first = p;
		userdrum_last = p;
	}
	else {
		userdrum_last->next = p;
		userdrum_last = p;
	}
	p->bank = bank;
	p->prog = prog;

	return p;
}


/*! recompute GS user drumset. */
Instrument *Instruments::recompute_userdrum(int bank, int prog)
{
	Instrument *ip = NULL;

	auto p = get_userdrum(bank, prog);
	auto source_note = p->source_note;
	auto source_prog = p->source_prog;

	free_tone_bank_element(&drumset[bank]->tone[prog]);
	if (drumset[source_prog]) {
		ToneBankElement *source_tone = &drumset[source_prog]->tone[source_note];

		if (source_tone->name == NULL /* NULL if "soundfont" directive is used */
			&& source_tone->instrument == NULL) {
			if ((ip = load_instrument(1, source_prog, source_note)) == NULL) {
				ip = MAGIC_ERROR_INSTRUMENT;
			}
			source_tone->instrument = ip;
		}
		if (source_tone->name) 
		{
			copy_tone_bank_element(&drumset[bank]->tone[prog], source_tone);
		}
		else if (drumset[0]->tone[source_note].name) 
		{
			copy_tone_bank_element(&drumset[bank]->tone[prog], &drumset[0]->tone[source_note]);
		}
		else {
			ctl_cmsg(CMSG_WARNING, VERB_NORMAL, "Referring user drum set %d, note %d not found - this instrument will not be heard as expected", bank, prog);
		}
	}
	return ip;
}

/*! convert GS user drumset assign groups to internal "alternate assign". */
void Instruments::recompute_userdrum_altassign(int bank, int group)
{
	int number = 0, i;
	char *params[131], param[10];
	ToneBank *bk;
	UserDrumset *p;

	for (p = userdrum_first; p != NULL; p = p->next) {
		if (p->assign_group == group) {
			sprintf(param, "%d", p->prog);
			params[number] = safe_strdup(param);
			number++;
		}
	}
	params[number] = NULL;

	alloc_instrument_bank(1, bank);
	bk = drumset[bank];
	bk->alt = add_altassign_string(bk->alt, params, number);
	for (i = number - 1; i >= 0; i--)
		free(params[i]);
}



}