www.pudn.com > gspmidi101src.zip > instrum.c
/*
TiMidity -- Experimental MIDI to WAVE converter
Copyright (C) 1995 Tuukka Toivonen
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
#ifndef _WIN32_WCE
#include
#endif
#if defined(__FreeBSD__) || defined (__WIN32__)
#include
#else
#include
#endif
#include "config.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "output.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[128]={&standard_tonebank},
*drumset[128]={&standard_drumset};
/* This is a special instrument, used for all melodic programs */
Instrument *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
static void free_instrument(Instrument *ip)
{
Sample *sp;
int i;
if (!ip) return;
for (i=0; isamples; i++)
{
sp=&(ip->sample[i]);
free(sp->data);
}
free(ip->sample);
free(ip);
}
static void free_bank(int dr, int b)
{
int i;
ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
for (i=0; i<128; i++)
if (bank->tone[i].instrument)
{
/* Not that this could ever happen, of course */
if (bank->tone[i].instrument != MAGIC_LOAD_INSTRUMENT)
free_instrument(bank->tone[i].instrument);
bank->tone[i].instrument=0;
}
}
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. */
return (((r * 44100) / play_mode->rate) * control_ratio)
<< ((fast_decay) ? 10 : 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) (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;
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(char *name, int percussion,
int panning, int amp, int note_to_use,
int strip_loop, int strip_envelope,
int strip_tail)
{
Instrument *ip;
Sample *sp;
FILE *fp;
uint8 tmp[1024];
int i,j,noluck=0;
#ifdef PATCH_EXT_LIST
static char *patch_ext[] = PATCH_EXT_LIST;
#endif
if (!name) return 0;
/* Open patch file */
if (!(fp=open_file(name, 1, OF_NORMAL)))
{
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)
{
strcpy(tmp, name);
strcat(tmp, patch_ext[i]);
if ((fp=open_file(tmp, 1, OF_NORMAL)))
{
noluck=0;
break;
}
}
}
#endif
}
if (noluck)
{
return 0;
}
/* 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 */
{
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; isamples; i++)
{
uint8 fractions;
int32 tmplong;
uint16 tmpshort;
uint8 tmpchar;
#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);
skip(fp, 7); /* Skip the wave name */
if (1 != fread(&fractions, 1, 1, fp))
{
fail:
for (j=0; jsample[j].data);
free(ip->sample);
free(ip);
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);
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);
/* 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;
}
else
{
sp->tremolo_sweep_increment=convert_tremolo_sweep(tmp[12]);
sp->tremolo_phase_increment=convert_tremolo_rate(tmp[13]);
sp->tremolo_depth=tmp[14];
}
if (!tmp[16] || !tmp[17])
{
sp->vibrato_sweep_increment=
sp->vibrato_control_ratio=sp->vibrato_depth=0;
}
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];
}
READ_CHAR(sp->modes);
skip(fp, 40); /* 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)
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)))
{
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;
}
}
for (j=0; j<6; j++)
{
sp->envelope_rate[j]=
convert_envelope_rate(tmp[j]);
sp->envelope_offset[j]=
convert_envelope_offset(tmp[6+j]);
}
/* Then read the sample data */
sp->data = safe_malloc(sp->data_length);
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,*new;
tmp=new=safe_malloc(sp->data_length*2);
while (i--)
*tmp++ = (uint16)(*cp++) << 8;
cp=(uint8 *)(sp->data);
sp->data = (sample_t *)new;
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=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. */
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=(double)(amp) / 100.0;
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. */
int32 i=sp->data_length/2;
int16 maxamp=0,a;
int16 *tmp=(int16 *)sp->data;
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a>maxamp)
maxamp=a;
}
sp->volume=32768.0 / (double)(maxamp);
}
#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;
/* 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);
#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. */
sp->data_length = sp->loop_end;
}
}
close_file(fp);
return ip;
}
static int fill_bank(int dr, int b)
{
int i, errors=0;
ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
if (!bank)
{
return 0;
}
for (i=0; i<128; i++)
{
if (bank->tone[i].instrument==MAGIC_LOAD_INSTRUMENT)
{
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 (!(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;
errors++;
}
else if (!(bank->tone[i].instrument=
load_instrument(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),
bank->tone[i].strip_tail )))
{
errors++;
}
}
}
return errors;
}
#if 0
// original
int load_missing_instruments(void)
{
int i=128,errors=0;
while (i--)
{
if (tonebank[i])
errors+=fill_bank(0,i);
if (drumset[i])
errors+=fill_bank(1,i);
}
return errors;
}
#else
int load_instruments = 0;
int load_errors = 0;
void start_load_missing_instruments(void)
{
load_instruments = 127;
load_errors = 0;
}
int load_missing_instruments(void)
{
if (load_instruments >= 0) {
if (tonebank[load_instruments])
load_errors += fill_bank(0, load_instruments);
if (drumset[load_instruments])
load_errors += fill_bank(1, load_instruments);
load_instruments--;
return 1;
}
return 0;
}
#endif
void free_instruments(void)
{
int i=128;
while (i--)
{
if (tonebank[i])
free_bank(0, i);
if (drumset[i])
free_bank(1, i);
}
}
int set_default_instrument(char *name)
{
Instrument *ip;
if (!(ip=load_instrument(name, 0, -1, -1, -1, 0, 0, 0)))
return -1;
if (default_instrument)
free_instrument(default_instrument);
default_instrument=ip;
default_program=SPECIAL_PROGRAM;
return 0;
}