/*

TiMidity -- Experimental MIDI to WAVE converter

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

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

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

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

(at your option) any later version.

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

but WITHOUT ANY WARRANTY; without even the implied warranty of

MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the

GNU General Public License for more details.

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

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

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

instrum.c

Code to load and unload GUS-compatible instrument patches.

*/

#include <stdio.h>

#include <string.h>

#include <stdlib.h>

#include "config.h"

#include "common.h"

#include "instrum.h"

#include "playmidi.h"

#include "output.h"

#include "ctrlmode.h"

#include "resample.h"

#include "tables.h"

#include "filter.h"

/* Some functions get aggravated if not even the standard banks are

available. */

static ToneBank standard_tonebank, standard_drumset;

ToneBank

*tonebank[MAXBANK]={&standard_tonebank},

*drumset[MAXBANK]={&standard_drumset};

/* This is a special instrument, used for all melodic programs */

InstrumentLayer *default_instrument=0;

/* This is only used for tracks that don't specify a program */

int default_program=DEFAULT_PROGRAM;

int antialiasing_allowed=0;

#ifdef FAST_DECAY

int fast_decay=1;

#else

int fast_decay=0;

#endif

int current_tune_number = 0;

int last_tune_purged = 0;

int current_patch_memory = 0;

int max_patch_memory = 60000000;

static void purge_as_required(void);

static void free_instrument(Instrument *ip)

{

Sample *sp;

int i;

if (!ip) return;

if (!ip->contents)

for (i=0; i<ip->samples; i++)

{

sp=&(ip->sample[i]);

if (sp->data) free(sp->data);

}

free(ip->sample);

if (!ip->contents)

for (i=0; i<ip->right_samples; i++)

{

sp=&(ip->right_sample[i]);

if (sp->data) free(sp->data);

}

if (ip->right_sample)

free(ip->right_sample);

free(ip);

}

static void free_layer(InstrumentLayer *lp)

{

InstrumentLayer *next;

current_patch_memory -= lp->size;

for (; lp; lp = next)

{

next = lp->next;

free_instrument(lp->instrument);

free(lp);

}

}

static void free_bank(int dr, int b)

{

int i;

ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);

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

{

if (bank->tone[i].layer)

{

/* Not that this could ever happen, of course */

if (bank->tone[i].layer != MAGIC_LOAD_INSTRUMENT)

{

free_layer(bank->tone[i].layer);

bank->tone[i].layer=NULL;

bank->tone[i].last_used=-1;

}

}

if (bank->tone[i].name)

{

free(bank->tone[i].name);

bank->tone[i].name = NULL;

}

}

}

static void free_old_bank(int dr, int b, int how_old)

{

int i;

ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);

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

if (bank->tone[i].layer && bank->tone[i].last_used < how_old)

{

if (bank->tone[i].layer != MAGIC_LOAD_INSTRUMENT)

{

ctl->cmsg(CMSG_INFO, VERB_DEBUG,

"Unloading %s %s[%d,%d] - last used %d.",

(dr)? "drum" : "inst", bank->tone[i].name,

i, b, bank->tone[i].last_used);

free_layer(bank->tone[i].layer);

bank->tone[i].layer=NULL;

bank->tone[i].last_used=-1;

}

}

}

int32 convert_envelope_rate_attack(uint8 rate, uint8 fastness)

{

int32 r;

r=3-((rate>>6) & 0x3);

r*=3;

r = (int32)(rate & 0x3f) << r; /* 6.9 fixed point */

/* 15.15 fixed point. */

return (((r * 44100) / play_mode->rate) * control_ratio)

<< 10;

}

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. */

return (((r * 44100) / play_mode->rate) * control_ratio)

<< ((fast_decay) ? 10 : 9);

}

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);

}

int32 convert_tremolo_sweep(uint8 sweep)

{

if (!sweep)

return 0;

return

((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /

(play_mode->rate * sweep);

}

int32 convert_vibrato_sweep(uint8 sweep, int32 vib_control_ratio)

{

if (!sweep)

return 0;

return

(int32) (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); */

}

int32 convert_tremolo_rate(uint8 rate)

{

return

((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) /

(TREMOLO_RATE_TUNING * play_mode->rate);

}

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;

sp+=ls;

le-=ls;

le/=2;

while (le--)

{

s=*sp;

*sp++=*ep;

*ep--=s;

}

}

/*

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 InstrumentLayer *load_instrument(char *name, int font_type, int percussion,

int panning, int amp, int cfg_tuning, int note_to_use,

int strip_loop, int strip_envelope,

int strip_tail, int bank, int gm_num, int sf_ix)

{

InstrumentLayer *lp, *lastlp, *headlp = 0;

Instrument *ip;

FILE *fp;

uint8 tmp[1024];

int i,j,noluck=0;

#ifdef PATCH_EXT_LIST

static char *patch_ext[] = PATCH_EXT_LIST;

#endif

int sf2flag = 0;

int right_samples = 0;

int stereo_channels = 1, stereo_layer;

int vlayer_list[19][4], vlayer, vlayer_count = 0;

if (!name) return 0;

/* Open patch file */

if ((fp=open_file(name, 1, OF_NORMAL)) == NULL)

{

noluck=1;

#ifdef PATCH_EXT_LIST

/* Try with various extensions */

for (i=0; patch_ext[i]; i++)

{

if (strlen(name)+strlen(patch_ext[i])<1024)

{

char path[1024];

strcpy(path, name);

strcat(path, patch_ext[i]);

if ((fp=open_file(path, 1, OF_NORMAL)) != NULL)

{

noluck=0;

break;

}

}

}

#endif

}

if (noluck)

{

ctl->cmsg(CMSG_ERROR, VERB_NORMAL,

"Instrument `%s' can't be found.", name);

return 0;

}

/*ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", current_filename);*/

/* Read some headers and do cursory sanity checks. There are loads

of magic offsets. This could be rewritten... */

if ((239 != fread(tmp, 1, 239, fp)) ||

(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);

return 0;

}

/* patch layout:

* bytes: info: starts at offset:

* 22 id (see above) 0

* 60 copyright 22

* 1 instruments 82

* 1 voices 83

* 1 channels 84

* 2 number of waveforms 85

* 2 master volume 87

* 4 datasize 89

* 36 reserved, but now: 93

* 7 "SF2EXT\0" id 93

* 1 right samples 100

* 28 reserved 101

* 2 instrument number 129

* 16 instrument name 131

* 4 instrument size 147

* 1 number of layers 151

* 40 reserved 152

* 1 layer duplicate 192

* 1 layer number 193

* 4 layer size 194

* 1 number of samples 198

* 40 reserved 199

* 239

* THEN, for each sample, see below

*/

if (!memcmp(tmp + 93, "SF2EXT", 6))

{

sf2flag = 1;

vlayer_count = tmp[152];

}

if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers,

0 means 1 */

{

ctl->cmsg(CMSG_ERROR, VERB_NORMAL,

"Can't handle patches with %d instruments", tmp[82]);

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]);

return 0;

}

if (sf2flag && vlayer_count > 0) {

for (i = 0; i < 9; i++)

for (j = 0; j < 4; j++)

vlayer_list[i][j] = tmp[153+i*4+j];

for (i = 9; i < 19; i++)

for (j = 0; j < 4; j++)

vlayer_list[i][j] = tmp[199+(i-9)*4+j];

}

else {

for (i = 0; i < 19; i++)

for (j = 0; j < 4; j++)

vlayer_list[i][j] = 0;

vlayer_list[0][0] = 0;

vlayer_list[0][1] = 127;

vlayer_list[0][2] = tmp[198];

vlayer_list[0][3] = 0;

vlayer_count = 1;

}

lastlp = 0;

for (vlayer = 0; vlayer < vlayer_count; vlayer++) {

lp=(InstrumentLayer *)safe_malloc(sizeof(InstrumentLayer));

lp->size = sizeof(InstrumentLayer);

lp->lo = vlayer_list[vlayer][0];

lp->hi = vlayer_list[vlayer][1];

ip=(Instrument *)safe_malloc(sizeof(Instrument));

lp->size += sizeof(Instrument);

lp->instrument = ip;

lp->next = 0;

if (lastlp) lastlp->next = lp;

else headlp = lp;

lastlp = lp;

if (sf2flag) ip->type = INST_SF2;

else ip->type = INST_GUS;

ip->samples = vlayer_list[vlayer][2];

ip->sample = (Sample *)safe_malloc(sizeof(Sample) * ip->samples);

lp->size += sizeof(Sample) * ip->samples;

ip->left_samples = ip->samples;

ip->left_sample = ip->sample;

right_samples = vlayer_list[vlayer][3];

ip->right_samples = right_samples;

if (right_samples)

{

ip->right_sample = (Sample *)safe_malloc(sizeof(Sample) * right_samples);

lp->size += sizeof(Sample) * right_samples;

stereo_channels = 2;

}

else ip->right_sample = 0;

ip->contents = 0;

ctl->cmsg(CMSG_INFO, VERB_NOISY, "%s%s[%d,%d] %s(%d-%d layer %d of %d)",

(percussion)? " ":"", name,

(percussion)? note_to_use : gm_num, bank,

(right_samples)? "(2) " : "",

lp->lo, lp->hi, vlayer+1, vlayer_count);

for (stereo_layer = 0; stereo_layer < stereo_channels; stereo_layer++)

{

int sample_count = 0;

if (stereo_layer == 0) sample_count = ip->left_samples;

else if (stereo_layer == 1) sample_count = ip->right_samples;

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

{

uint8 fractions;

int32 tmplong;

uint16 tmpshort;

uint16 sample_volume = 0;

uint8 tmpchar;

Sample *sp = 0;

uint8 sf2delay = 0;

#define READ_CHAR(thing) \

if (1 != fread(&tmpchar, 1, 1, fp)) goto fail; \

thing = tmpchar;

#define READ_SHORT(thing) \

if (1 != fread(&tmpshort, 2, 1, fp)) goto fail; \

thing = LE_SHORT(tmpshort);

#define READ_LONG(thing) \

if (1 != fread(&tmplong, 4, 1, fp)) goto fail; \

thing = LE_LONG(tmplong);

/*

* 7 sample name

* 1 fractions

* 4 length

* 4 loop start

* 4 loop end

* 2 sample rate

* 4 low frequency

* 4 high frequency

* 2 finetune

* 1 panning

* 6 envelope rates |

* 6 envelope offsets | 18 bytes

* 3 tremolo sweep, rate, depth |

* 3 vibrato sweep, rate, depth |

* 1 sample mode

* 2 scale frequency

* 2 scale factor

* 2 sample volume (??)

* 34 reserved

* Now: 1 delay

* 33 reserved

*/

skip(fp, 7); /* Skip the wave name */

if (1 != fread(&fractions, 1, 1, fp))

{

fail:

ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d", i);

if (stereo_layer == 1)

{

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

free(ip->right_sample[j].data);

free(ip->right_sample);

i = ip->left_samples;

}

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

free(ip->left_sample[j].data);

free(ip->left_sample);

free(ip);

free(lp);

return 0;

}

if (stereo_layer == 0) sp=&(ip->left_sample[i]);

else if (stereo_layer == 1) sp=&(ip->right_sample[i]);

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(fp, 2); /* Why have a "root frequency" and then "tuning"?? */

READ_CHAR(tmp[0]);

if (panning==-1)

sp->panning = (tmp[0] * 8 + 4) & 0x7f;

else

sp->panning=(uint8)(panning & 0x7F);

sp->resonance=0;

sp->cutoff_freq=0;

sp->reverberation=0;

sp->chorusdepth=0;

sp->exclusiveClass=0;

sp->keyToModEnvHold=0;

sp->keyToModEnvDecay=0;

sp->keyToVolEnvHold=0;

sp->keyToVolEnvDecay=0;

if (cfg_tuning)

{

double tune_factor = (double)(cfg_tuning)/1200.0;

tune_factor = pow(2.0, tune_factor);

sp->root_freq = (uint32)( tune_factor * (double)sp->root_freq );

}

/* envelope, tremolo, and vibrato */

if (18 != fread(tmp, 1, 18, fp)) goto fail;

if (!tmp[13] || !tmp[14])

{

sp->tremolo_sweep_increment=

sp->tremolo_phase_increment=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=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);

READ_SHORT(sp->freq_center);

READ_SHORT(sp->freq_scale);

if (sf2flag)

{

READ_SHORT(sample_volume);

READ_CHAR(sf2delay);

READ_CHAR(sp->exclusiveClass);

skip(fp, 32);

}

else

{

skip(fp, 36);

}

/* Mark this as a fixed-pitch instrument if such a deed is desired. */

if (note_to_use!=-1)

sp->note_to_use=(uint8)(note_to_use);

else

sp->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)))

{

ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain");

sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING |

MODES_PINGPONG | MODES_REVERSE);

}

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");

}

}

sp->attenuation = 0;

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

{

sp->envelope_rate[j]=

(j<3)? convert_envelope_rate_attack(tmp[j], 11) : convert_envelope_rate(tmp[j]);

sp->envelope_offset[j]=

convert_envelope_offset(tmp[6+j]);

}

if (sf2flag)

{

if (sf2delay > 5) sf2delay = 5;

sp->envelope_rate[DELAY] = (int32)( (sf2delay*play_mode->rate) / 1000 );

}

else

{

sp->envelope_rate[DELAY]=0;

}

sp->envelope_offset[DELAY]=0;

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

{

sp->modulation_rate[j]=sp->envelope_rate[j];

sp->modulation_offset[j]=sp->envelope_offset[j];

}

sp->modulation_rate[DELAY] = sp->modulation_offset[DELAY] = 0;

sp->modEnvToFilterFc=0;

sp->modEnvToPitch=0;

sp->lfo_sweep_increment = 0;

sp->lfo_phase_increment = 0;

sp->modLfoToFilterFc = 0;

sp->vibrato_delay = 0;

/* Then read the sample data */

if (sp->data_length/2 > MAX_SAMPLE_SIZE)

{

goto fail;

}

sp->data = safe_malloc(sp->data_length + 1);

lp->size += sp->data_length + 1;

if (1 != fread(sp->data, sp->data_length, 1, fp))

goto fail;

if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */

{

int32 i=sp->data_length;

uint8 *cp=(uint8 *)(sp->data);

uint16 *tmp,*newdta;

tmp=newdta=safe_malloc(sp->data_length*2 + 2);

while (i--)

*tmp++ = (uint16)(*cp++) << 8;

cp=(uint8 *)(sp->data);

sp->data = (sample_t *)newdta;

free(cp);

sp->data_length *= 2;

sp->loop_start *= 2;

sp->loop_end *= 2;

}

#if SDL_BYTEORDER == SDL_BIG_ENDIAN

else

/* convert to machine byte order */

{

int32 i=sp->data_length/2;

int16 *tmp=(int16 *)sp->data,s;

while (i--)

{

s=LE_SHORT(*tmp);

*tmp++=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 reverse loops and pass them off as normal loops */

if (sp->modes & MODES_REVERSE)

{

int32 t;

/* The GUS apparently plays reverse loops by reversing the

whole sample. We do the same because the GUS does not SUCK. */

ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name);

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 */

}

/* If necessary do some anti-aliasing filtering */

if (antialiasing_allowed)

antialiasing(sp,play_mode->rate);

#ifdef ADJUST_SAMPLE_VOLUMES

if (amp!=-1)

sp->volume=(FLOAT_T)((amp) / 100.0);

else if (sf2flag)

sp->volume=(FLOAT_T)((sample_volume) / 255.0);

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. */

uint32 i, numsamps=sp->data_length/2;

uint32 higher=0, highcount=0;

int16 maxamp=0,a;

int16 *tmp=(int16 *)sp->data;

i = numsamps;

while (i--)

{

a=*tmp++;

if (a<0) a=-a;

if (a>maxamp)

maxamp=a;

}

tmp=(int16 *)sp->data;

i = numsamps;

while (i--)

{

a=*tmp++;

if (a<0) a=-a;

if (a > 3*maxamp/4)

{

higher += a;

highcount++;

}

}

if (highcount) higher /= highcount;

else higher = 10000;

sp->volume = (32768.0 * 0.875) / (double)higher ;

ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f", sp->volume);

}

#else

if (amp!=-1)

sp->volume=(double)(amp) / 100.0;

else

sp->volume=1.0;

#endif

sp->data_length /= 2; /* These are in bytes. Convert into samples. */

sp->loop_start /= 2;

sp->loop_end /= 2;

sp->data[sp->data_length] = sp->data[sp->data_length-1];

/* Then fractional samples */

sp->data_length <<= FRACTION_BITS;

sp->loop_start <<= FRACTION_BITS;

sp->loop_end <<= FRACTION_BITS;

/* trim off zero data at end */

{

int ls = sp->loop_start>>FRACTION_BITS;

int le = sp->loop_end>>FRACTION_BITS;

int se = sp->data_length>>FRACTION_BITS;

while (se > 1 && !sp->data[se-1]) se--;

if (le > se) le = se;

if (ls >= le) sp->modes &= ~MODES_LOOPING;

sp->loop_end = le<<FRACTION_BITS;

sp->data_length = se<<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);

#ifdef LOOKUP_HACK

/* Squash the 16-bit data into 8 bits. */

{

uint8 *gulp,*ulp;

int16 *swp;

int l=sp->data_length >> FRACTION_BITS;

gulp=ulp=safe_malloc(l+1);

swp=(int16 *)sp->data;

while(l--)

*ulp++ = (*swp++ >> 8) & 0xFF;

free(sp->data);

sp->data=(sample_t *)gulp;

}

#endif

if (strip_tail==1)

{

/* Let's not really, just say we did. */

ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail");

sp->data_length = sp->loop_end;

}

} /* end of sample loop */

} /* end of stereo layer loop */

} /* end of vlayer loop */

close_file(fp);

return headlp;

}

static int fill_bank(int dr, int b)

{

int i, errors=0;

ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);

if (!bank)

{

ctl->cmsg(CMSG_ERROR, VERB_NORMAL,

"Huh. Tried to load instruments in non-existent %s %d",

(dr) ? "drumset" : "tone bank", b);

return 0;

}

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

{

if (bank->tone[i].layer==MAGIC_LOAD_INSTRUMENT)

{

if (!(bank->tone[i].name))

{

ctl->cmsg(CMSG_WARNING, (b!=0) ? VERB_VERBOSE : VERB_NORMAL,

"No instrument mapped to %s %d, program %d%s",

(dr)? "drum set" : "tone bank", b, i,

(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].layer))

standard_tonebank.tone[i].layer=

MAGIC_LOAD_INSTRUMENT;

}

else

{

if (!(standard_drumset.tone[i].layer))

standard_drumset.tone[i].layer=

MAGIC_LOAD_INSTRUMENT;

}

}

bank->tone[i].layer=0;

errors++;

}

else if (!(bank->tone[i].layer=

load_instrument(bank->tone[i].name,

bank->tone[i].font_type,

(dr) ? 1 : 0,

bank->tone[i].pan,

bank->tone[i].amp,

bank->tone[i].tuning,

(bank->tone[i].note!=-1) ?

bank->tone[i].note :

((dr) ? i : -1),

(bank->tone[i].strip_loop!=-1) ?

bank->tone[i].strip_loop :

((dr) ? 1 : -1),

(bank->tone[i].strip_envelope != -1) ?

bank->tone[i].strip_envelope :

((dr) ? 1 : -1),

bank->tone[i].strip_tail,

b,

((dr) ? i + 128 : i),

bank->tone[i].sf_ix

)))

{

ctl->cmsg(CMSG_ERROR, VERB_NORMAL,

"Couldn't load instrument %s (%s %d, program %d)",

bank->tone[i].name,

(dr)? "drum set" : "tone bank", b, i);

errors++;

}

else

{ /* it's loaded now */

bank->tone[i].last_used = current_tune_number;

current_patch_memory += bank->tone[i].layer->size;

purge_as_required();

if (current_patch_memory > max_patch_memory) {

ctl->cmsg(CMSG_ERROR, VERB_NORMAL,

"Not enough memory to load instrument %s (%s %d, program %d)",

bank->tone[i].name,

(dr)? "drum set" : "tone bank", b, i);

errors++;

free_layer(bank->tone[i].layer);

bank->tone[i].layer=0;

bank->tone[i].last_used=-1;

}

#if 0

if (check_for_rc()) {

free_layer(bank->tone[i].layer);

bank->tone[i].layer=0;

bank->tone[i].last_used=-1;

return 0;

}

#endif

}

}

}

return errors;

}

static void free_old_instruments(int how_old)

{

int i=MAXBANK;

while(i--)

{

if (tonebank[i])

free_old_bank(0, i, how_old);

if (drumset[i])

free_old_bank(1, i, how_old);

}

}

static void purge_as_required(void)

{

if (!max_patch_memory) return;

while (last_tune_purged < current_tune_number

&& current_patch_memory > max_patch_memory)

{

last_tune_purged++;

free_old_instruments(last_tune_purged);

}

}

int load_missing_instruments(void)

{

int i=MAXBANK,errors=0;

while (i--)

{

if (tonebank[i])

errors+=fill_bank(0,i);

if (drumset[i])

errors+=fill_bank(1,i);

}

current_tune_number++;

return errors;