instrum.c
1
2
3
4
5
/*
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
6
7
it under the terms of the Perl Artistic License, available in COPYING.
*/
8
9
10
11
12
13
14
15
16
17
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include "config.h"
#include "common.h"
#include "instrum.h"
#include "playmidi.h"
#include "output.h"
18
#include "ctrlmode.h"
19
20
21
22
23
24
25
26
#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
27
28
*tonebank[MAXBANK]={&standard_tonebank},
*drumset[MAXBANK]={&standard_drumset};
29
30
/* This is a special instrument, used for all melodic programs */
31
InstrumentLayer *default_instrument=0;
32
33
34
35
36
37
38
39
40
41
42
/* 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
43
44
45
46
47
48
49
50
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);
51
52
53
54
55
static void free_instrument(Instrument *ip)
{
Sample *sp;
int i;
if (!ip) return;
56
57
if (!ip->contents)
58
59
60
for (i=0; i<ip->samples; i++)
{
sp=&(ip->sample[i]);
61
if (sp->data) free(sp->data);
62
63
}
free(ip->sample);
64
65
66
67
68
69
70
71
72
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);
73
74
75
free(ip);
}
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
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);
}
}
91
92
93
94
static void free_bank(int dr, int b)
{
int i;
ToneBank *bank=((dr) ? drumset[b] : tonebank[b]);
95
for (i=0; i<MAXPROG; i++)
97
if (bank->tone[i].layer)
98
99
100
{
/* Not that this could ever happen, of course */
if (bank->tone[i].layer != MAGIC_LOAD_INSTRUMENT)
101
102
{
free_layer(bank->tone[i].layer);
103
bank->tone[i].layer=NULL;
104
105
bank->tone[i].last_used=-1;
}
106
107
108
109
110
111
112
}
if (bank->tone[i].name)
{
free(bank->tone[i].name);
bank->tone[i].name = NULL;
}
}
113
114
}
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
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);
130
bank->tone[i].layer=NULL;
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
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)
151
152
153
154
155
156
157
158
159
160
161
162
{
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);
}
163
int32 convert_envelope_offset(uint8 offset)
164
165
166
167
168
169
170
171
{
/* 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);
}
172
int32 convert_tremolo_sweep(uint8 sweep)
173
174
175
176
177
178
179
180
181
{
if (!sweep)
return 0;
return
((control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
(play_mode->rate * sweep);
}
182
int32 convert_vibrato_sweep(uint8 sweep, int32 vib_control_ratio)
183
184
185
186
187
188
189
190
191
192
193
194
195
196
{
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); */
}
197
int32 convert_tremolo_rate(uint8 rate)
198
199
200
201
202
203
{
return
((SINE_CYCLE_LENGTH * control_ratio * rate) << RATE_SHIFT) /
(TREMOLO_RATE_TUNING * play_mode->rate);
}
204
int32 convert_vibrato_rate(uint8 rate)
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
{
/* 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 */
236
static InstrumentLayer *load_instrument(const char *name, int font_type, int percussion,
237
int panning, int amp, int cfg_tuning, int note_to_use,
238
int strip_loop, int strip_envelope,
239
int strip_tail, int bank, int gm_num, int sf_ix)
240
{
241
InstrumentLayer *lp, *lastlp, *headlp = 0;
242
243
244
245
246
247
248
Instrument *ip;
FILE *fp;
uint8 tmp[1024];
int i,j,noluck=0;
#ifdef PATCH_EXT_LIST
static char *patch_ext[] = PATCH_EXT_LIST;
#endif
249
250
251
int sf2flag = 0;
int right_samples = 0;
int stereo_channels = 1, stereo_layer;
252
int vlayer_list[19][4], vlayer, vlayer_count = 0;
253
254
255
256
if (!name) return 0;
/* Open patch file */
257
if ((fp=open_file(name, 1, OF_NORMAL)) == NULL)
258
259
260
261
262
263
{
noluck=1;
#ifdef PATCH_EXT_LIST
/* Try with various extensions */
for (i=0; patch_ext[i]; i++)
{
264
if (strlen(name)+strlen(patch_ext[i])<PATH_MAX)
265
{
266
char path[PATH_MAX];
267
268
269
strcpy(path, name);
strcat(path, patch_ext[i]);
if ((fp=open_file(path, 1, OF_NORMAL)) != NULL)
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
{
noluck=0;
break;
}
}
}
#endif
}
if (noluck)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Instrument `%s' can't be found.", name);
return 0;
}
286
/*ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", current_filename);*/
287
288
289
290
291
292
293
294
295
296
297
298
/* 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;
}
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
/* 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];
}
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
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;
}
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
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)
397
{
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
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++)
{
413
int sample_count = 0;
414
415
416
if (stereo_layer == 0) sample_count = ip->left_samples;
else if (stereo_layer == 1) sample_count = ip->right_samples;
418
419
for (i=0; i < sample_count; i++)
{
420
421
422
uint8 fractions;
int32 tmplong;
uint16 tmpshort;
423
uint16 sample_volume = 0;
424
uint8 tmpchar;
425
426
Sample *sp = 0;
uint8 sf2delay = 0;
427
428
429
430
431
432
433
434
435
436
437
#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);
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
/*
* 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
*/
461
462
463
464
465
466
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);
467
468
469
470
471
472
473
if (stereo_layer == 1)
{
for (j=0; j<i; j++)
free(ip->right_sample[j].data);
free(ip->right_sample);
i = ip->left_samples;
}
474
for (j=0; j<i; j++)
475
476
free(ip->left_sample[j].data);
free(ip->left_sample);
477
free(ip);
479
480
481
return 0;
}
482
483
484
if (stereo_layer == 0) sp=&(ip->left_sample[i]);
else if (stereo_layer == 1) sp=&(ip->right_sample[i]);
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
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);
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
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 );
}
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
/* 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);
558
559
560
561
562
563
564
565
566
567
568
569
570
571
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);
}
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
/* 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");
}
}
634
635
636
sp->attenuation = 0;
for (j=ATTACK; j<DELAY; j++)
637
638
{
sp->envelope_rate[j]=
639
(j<3)? convert_envelope_rate_attack(tmp[j], 11) : convert_envelope_rate(tmp[j]);
640
641
642
sp->envelope_offset[j]=
convert_envelope_offset(tmp[6+j]);
}
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
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;
666
667
/* Then read the sample data */
668
669
670
671
672
673
674
if (sp->data_length/2 > MAX_SAMPLE_SIZE)
{
goto fail;
}
sp->data = safe_malloc(sp->data_length + 1);
lp->size += sp->data_length + 1;
675
676
677
678
679
680
681
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);
682
683
uint16 *tmp,*newdta;
tmp=newdta=safe_malloc(sp->data_length*2 + 2);
684
685
686
while (i--)
*tmp++ = (uint16)(*cp++) << 8;
cp=(uint8 *)(sp->data);
687
sp->data = (sample_t *)newdta;
688
689
690
691
692
free(cp);
sp->data_length *= 2;
sp->loop_start *= 2;
sp->loop_end *= 2;
}
693
#if SDL_BYTEORDER == SDL_BIG_ENDIAN
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
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)
740
741
742
sp->volume=(FLOAT_T)((amp) / 100.0);
else if (sf2flag)
sp->volume=(FLOAT_T)((sample_volume) / 255.0);
743
744
745
746
747
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. */
748
749
uint32 i, numsamps=sp->data_length/2;
uint32 higher=0, highcount=0;
750
751
int16 maxamp=0,a;
int16 *tmp=(int16 *)sp->data;
752
i = numsamps;
753
754
755
756
757
758
759
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a>maxamp)
maxamp=a;
}
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
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 ;
775
776
777
778
779
780
781
782
783
784
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. */
786
787
sp->loop_start /= 2;
sp->loop_end /= 2;
788
sp->data[sp->data_length] = sp->data[sp->data_length-1];
789
790
791
792
793
794
/* Then fractional samples */
sp->data_length <<= FRACTION_BITS;
sp->loop_start <<= FRACTION_BITS;
sp->loop_end <<= FRACTION_BITS;
795
796
797
798
799
800
801
802
803
804
805
806
/* 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;
}
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
/* 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;
}
840
841
842
843
} /* end of sample loop */
} /* end of stereo layer loop */
} /* end of vlayer loop */
844
845
close_file(fp);
846
return headlp;
847
848
849
850
851
852
853
854
855
856
857
858
859
}
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;
}
860
for (i=0; i<MAXPROG; i++)
861
{
862
if (bank->tone[i].layer==MAGIC_LOAD_INSTRUMENT)
863
864
865
866
867
868
869
870
871
872
873
874
875
{
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)
{
876
877
if (!(standard_tonebank.tone[i].layer))
standard_tonebank.tone[i].layer=
878
879
880
881
MAGIC_LOAD_INSTRUMENT;
}
else
{
882
883
if (!(standard_drumset.tone[i].layer))
standard_drumset.tone[i].layer=
884
885
886
MAGIC_LOAD_INSTRUMENT;
}
}
887
bank->tone[i].layer=0;
888
889
errors++;
}
890
else if (!(bank->tone[i].layer=
891
load_instrument(bank->tone[i].name,
892
bank->tone[i].font_type,
893
894
895
(dr) ? 1 : 0,
bank->tone[i].pan,
bank->tone[i].amp,
896
bank->tone[i].tuning,
897
(bank->tone[i].note!=-1) ?
898
899
bank->tone[i].note :
((dr) ? i : -1),
900
901
902
903
904
905
(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),
906
907
908
909
910
bank->tone[i].strip_tail,
b,
((dr) ? i + 128 : i),
bank->tone[i].sf_ix
)))
911
912
913
914
915
916
917
{
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++;
}
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
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
}
942
943
944
945
946
}
}
return errors;
}
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
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);
}
}
972
973
int load_missing_instruments(void)
{
974
int i=MAXBANK,errors=0;
975
976
977
978
979
980
981
while (i--)
{
if (tonebank[i])
errors+=fill_bank(0,i);
if (drumset[i])
errors+=fill_bank(1,i);
}
982
current_tune_number++;
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
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);
}
}
998
int set_default_instrument(const char *name)
999
{
1000
InstrumentLayer *lp;