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rockbox/apps/plugins/sdl/SDL_mixer/timidity/instrum.c
Franklin Wei a855d62025 Port of Duke Nukem 3D 
This ports Fabien Sanglard's Chocolate Duke to run on a version of SDL
for Rockbox.

Change-Id: I8f2c4c78af19de10c1633ed7bb7a997b43256dd9
2017-12-23 21:01:26 -05:00

1018 lines
25 KiB
C
Raw Blame History

/*
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 Perl Artistic License, available in COPYING.
*/
#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(const 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])<PATH_MAX)
{
char path[PATH_MAX];
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);
fclose(fp);
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);
fclose(fp);
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]);
fclose(fp);
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]);
fclose(fp);
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)) { \
printf("error readc\n"); goto fail; } \
thing = tmpchar;
#define READ_SHORT(thing) \
if (1 != fread(&tmpshort, 2, 1, fp)) { \
printf("error reads\n"); goto fail; } \
thing = LE_SHORT(tmpshort);
#define READ_LONG(thing) \
if (1 != fread(&tmplong, 4, 1, fp)) { \
printf("error readl\n"); 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))
{
printf("error 1\n");
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);
fclose(fp);
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)) { printf("error 2\n"); 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)
{
printf("error 3\n");
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))
{
printf("error 4\n");
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;
}
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(const char *name)
{
InstrumentLayer *lp;
/* if (!(lp=load_instrument(name, 0, -1, -1, -1, 0, 0, 0))) */
if (!(lp=load_instrument(name, FONT_NORMAL, 0, -1, -1, 0, -1, -1, -1, -1, 0, -1, -1)))
return -1;
if (default_instrument)
free_layer(default_instrument);
default_instrument=lp;
default_program=SPECIAL_PROGRAM;
return 0;
}
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