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

instrum.c

Code to load and unload GUS-compatible instrument patches.

*/

#if HAVE_CONFIG_H

# include <config.h>

#endif

#include <stdio.h>

#include <string.h>

#include <stdlib.h>

#include "SDL.h"

#include "timidity.h"

#include "options.h"

#include "common.h"

#include "instrum.h"

#include "resample.h"

#include "tables.h"

static void free_instrument(Instrument *ip)

{

Sample *sp;

int i;

if (!ip) return;

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

{

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

}

free(ip->sample);

free(ip);

}

{

int i;

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

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

if (bank->instrument[i])

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

if (bank->instrument[i] != MAGIC_LOAD_INSTRUMENT)

free_instrument(bank->instrument[i]);

bank->instrument[i]=0;

}

}

Sint32 r;

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

r *= 3;

r = (Sint32) (rate & 0x3f) << r; /* 6.9 fixed point */

/* 15.15 fixed point. */

#ifdef FAST_DECAY

return r << 10;

#else

return r << 9;

#endif

}

{

/* 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);

}

{

if (!sweep)

return 0;

return

((song->control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /

(song->rate * sweep);

}

static Sint32 convert_vibrato_sweep(MidiSong *song, Uint8 sweep,

Sint32 vib_control_ratio)

{

if (!sweep)

return 0;

return

(Sint32) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT)

/ (double)(song->rate * sweep));

/* this was overflowing with seashore.pat

((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /

}

{

return

((SINE_CYCLE_LENGTH * song->control_ratio * rate) << RATE_SHIFT) /

(TREMOLO_RATE_TUNING * song->rate);

}

{

/* Return a suitable vibrato_control_ratio value */

return

(rate * 2 * VIBRATO_SAMPLE_INCREMENTS);

}

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 Instrument *load_instrument(MidiSong *song, char *name, int percussion,

int panning, int amp, int note_to_use,

{

Instrument *ip;

Sample *sp;

SDL_RWops *rw;

char tmp[1024];

int i,j,noluck=0;

static char *patch_ext[] = PATCH_EXT_LIST;

if (!name) return 0;

/* Open patch file */

{

noluck=1;

/* Try with various extensions */

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

{

strcpy(tmp, name);

strcat(tmp, patch_ext[i]);

if ((rw=open_file(tmp)) != NULL)

{

noluck=0;

break;

}

}

}

}

if (noluck)

{

return 0;

}

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

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

(memcmp(tmp, "GF1PATCH110\0ID#000002", 22) &&

memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the

differences are */

{

return 0;

}

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

0 means 1 */

{

return 0;

}

if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */

{

return 0;

}

ip=safe_malloc(sizeof(Instrument));

ip->samples = tmp[198];

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

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

Uint8 fractions;

Sint32 tmplong;

Uint16 tmpshort;

Uint8 tmpchar;

#define READ_CHAR(thing) \

thing = tmpchar;

#define READ_SHORT(thing) \

if (1 != SDL_RWread(rw, &tmpshort, 2, 1)) goto fail; \

thing = SDL_SwapLE16(tmpshort);

if (1 != SDL_RWread(rw, &tmplong, 4, 1)) goto fail; \

thing = SDL_SwapLE32(tmplong);

SDL_RWseek(rw, 7, RW_SEEK_CUR); /* Skip the wave name */

if (1 != SDL_RWread(rw, &fractions, 1, 1))

{

fail:

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

free(ip->sample);

return 0;

}

sp=&(ip->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);

SDL_RWseek(rw, 2, RW_SEEK_CUR); /* Why have a "root frequency" and then

* "tuning"?? */

READ_CHAR(tmp[0]);

if (panning==-1)

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

else

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

{

sp->tremolo_sweep_increment=

sp->tremolo_phase_increment=sp->tremolo_depth=0;

}

else

{

sp->tremolo_sweep_increment=convert_tremolo_sweep(song, tmp[12]);

sp->tremolo_phase_increment=convert_tremolo_rate(song, tmp[13]);

}

if (!tmp[16] || !tmp[17])

{

sp->vibrato_sweep_increment=

sp->vibrato_control_ratio=sp->vibrato_depth=0;

}

else

{

}

READ_CHAR(sp->modes);

SDL_RWseek(rw, 40, RW_SEEK_CUR); /* skip the useless scale frequency, scale

factor (what's it mean?), and reserved

space */

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

if (note_to_use!=-1)

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)))

{

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

MODES_PINGPONG | MODES_REVERSE);

}

if (strip_envelope==1)

{

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);

}

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;

}

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;

}

}

{

sp->envelope_rate[j]=

sp->envelope_offset[j]=

convert_envelope_offset(tmp[6+j]);

}

/* Then read the sample data */

goto fail;

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

{

sp->data_length *= 2;

sp->loop_start *= 2;

sp->loop_end *= 2;

tmp16 = new16 = (Uint16 *) safe_malloc(sp->data_length+4);

while (k--)

*tmp16++ = (Uint16)(*cp++) << 8;

free(sp->data);

sp->data = (sample_t *)new16;

else

/* convert to machine byte order */

{

Sint32 k=sp->data_length/2;

Sint16 *tmp16=(Sint16 *)sp->data,s;

while (k--)

{

s=SDL_SwapLE16(*tmp16);

*tmp16++=s;

}

}

#endif

if (sp->modes & MODES_UNSIGNED) /* convert to signed data */

{

Sint32 k=sp->data_length/2;

Sint16 *tmp16=(Sint16 *)sp->data;

while (k--)

*tmp16++ ^= 0x8000;

}

/* Reverse 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. */

SNDDBG(("Reverse loop in %s\n", name));

reverse_data((Sint16 *)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 */

}

#ifdef ADJUST_SAMPLE_VOLUMES

if (amp!=-1)

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. */

Sint16 *tmp16=(Sint16 *)sp->data;

while (k--)

if (a<0) a=-a;

if (a>maxamp)

maxamp=a;

}

sp->volume=(float)(32768.0 / maxamp);

SNDDBG((" * volume comp: %f\n", 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;

/* initialize the added extra sample space (see the +4 bytes in

allocation) using the last actual sample: */

sp->data[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))

if (strip_tail==1)

{

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

sp->data_length = sp->loop_end;

}

SDL_RWclose(rw);

return ip;

}

{

int i, errors=0;

if (!bank)

{

SNDDBG(("Huh. Tried to load instruments in non-existent %s %d\n",

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

return 0;

}

{

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

{

if (b!=0)

{

/* Mark the corresponding instrument in the default

bank / drumset for loading (if it isn't already) */

if (!dr)

{

if (!(song->tonebank[0]->instrument[i]))

song->tonebank[0]->instrument[i] =

MAGIC_LOAD_INSTRUMENT;

}

else

{

if (!(song->drumset[0]->instrument[i]))

song->drumset[0]->instrument[i] =

MAGIC_LOAD_INSTRUMENT;

}

}

errors++;

}

else if (!(bank->instrument[i] =

load_instrument(song,

bank->tone[i].name,

(dr) ? 1 : 0,

bank->tone[i].pan,

bank->tone[i].amp,

(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),

errors++;

}

}

}

return errors;

}

while (i--)

{

if (song->tonebank[i])

errors+=fill_bank(song,0,i);

if (song->drumset[i])

errors+=fill_bank(song,1,i);

}

return errors;

}

while(i--)

{

if (song->tonebank[i])

free_bank(song, 0, i);

if (song->drumset[i])

free_bank(song, 1, i);

}

}

Instrument *ip;

if (!(ip=load_instrument(song, name, 0, -1, -1, -1, 0, 0, 0)))

song->default_instrument = ip;

song->default_program = SPECIAL_PROGRAM;

return 0;

}