/
music_wav.c
638 lines (554 loc) · 16.8 KB
2
SDL_mixer: An audio mixer library based on the SDL library
3
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
20
21
*/
22
#ifdef MUSIC_WAV
24
/* This file supports streaming WAV files */
26
#include "music_wav.h"
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
typedef struct {
SDL_bool active;
Uint32 start;
Uint32 stop;
Uint32 initial_play_count;
Uint32 current_play_count;
} WAVLoopPoint;
typedef struct {
SDL_RWops *src;
SDL_bool freesrc;
SDL_AudioSpec spec;
int volume;
42
int play_count;
43
44
Sint64 start;
Sint64 stop;
45
46
Uint8 *buffer;
SDL_AudioStream *stream;
47
48
int numloops;
WAVLoopPoint *loops;
51
52
53
54
55
56
57
58
59
60
61
/*
Taken with permission from SDL_wave.h, part of the SDL library,
available at: http://www.libsdl.org/
and placed under the same license as this mixer library.
*/
/* WAVE files are little-endian */
/*******************************************/
/* Define values for Microsoft WAVE format */
/*******************************************/
62
63
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
64
#define FMT 0x20746D66 /* "fmt " */
65
#define DATA 0x61746164 /* "data" */
66
#define SMPL 0x6c706d73 /* "smpl" */
67
68
69
70
#define PCM_CODE 1
#define ADPCM_CODE 2
#define WAVE_MONO 1
#define WAVE_STEREO 2
72
typedef struct {
73
/* Not saved in the chunk we read:
74
75
Uint32 chunkID;
Uint32 chunkLen;
77
78
79
80
81
82
Uint16 encoding;
Uint16 channels; /* 1 = mono, 2 = stereo */
Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */
Uint32 byterate; /* Average bytes per second */
Uint16 blockalign; /* Bytes per sample block */
Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */
83
84
} WaveFMT;
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
typedef struct {
Uint32 identifier;
Uint32 type;
Uint32 start;
Uint32 end;
Uint32 fraction;
Uint32 play_count;
} SampleLoop;
typedef struct {
/* Not saved in the chunk we read:
Uint32 chunkID;
Uint32 chunkLen;
*/
Uint32 manufacturer;
Uint32 product;
Uint32 sample_period;
Uint32 MIDI_unity_note;
Uint32 MIDI_pitch_fraction;
Uint32 SMTPE_format;
Uint32 SMTPE_offset;
Uint32 sample_loops;
Uint32 sampler_data;
108
SampleLoop loops[1];
109
} SamplerChunk;
110
111
112
113
/*********************************************/
/* Define values for AIFF (IFF audio) format */
/*********************************************/
114
115
116
117
#define FORM 0x4d524f46 /* "FORM" */
#define AIFF 0x46464941 /* "AIFF" */
#define SSND 0x444e5353 /* "SSND" */
#define COMM 0x4d4d4f43 /* "COMM" */
120
/* Function to load the WAV/AIFF stream */
121
122
static SDL_bool LoadWAVMusic(WAV_Music *wave);
static SDL_bool LoadAIFFMusic(WAV_Music *wave);
124
static void WAV_Delete(void *context);
126
/* Load a WAV stream from the given RWops object */
127
static void *WAV_CreateFromRW(SDL_RWops *src, int freesrc)
128
{
129
130
WAV_Music *music;
Uint32 magic;
131
SDL_bool loaded = SDL_FALSE;
133
134
135
136
137
music = (WAV_Music *)SDL_calloc(1, sizeof(*music));
if (!music) {
SDL_OutOfMemory();
return NULL;
}
138
music->src = src;
139
music->volume = MIX_MAX_VOLUME;
141
142
143
144
145
magic = SDL_ReadLE32(src);
if (magic == RIFF || magic == WAVE) {
loaded = LoadWAVMusic(music);
} else if (magic == FORM) {
loaded = LoadAIFFMusic(music);
146
} else {
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
Mix_SetError("Unknown WAVE format");
}
if (!loaded) {
SDL_free(music);
return NULL;
}
music->buffer = (Uint8*)SDL_malloc(music->spec.size);
if (!music->buffer) {
WAV_Delete(music);
return NULL;
}
music->stream = SDL_NewAudioStream(
music->spec.format, music->spec.channels, music->spec.freq,
music_spec.format, music_spec.channels, music_spec.freq);
if (!music->stream) {
WAV_Delete(music);
return NULL;
165
166
167
music->freesrc = freesrc;
return music;
170
static void WAV_SetVolume(void *context, int volume)
172
173
WAV_Music *music = (WAV_Music *)context;
music->volume = volume;
174
175
176
}
/* Start playback of a given WAV stream */
177
static int WAV_Play(void *context, int play_count)
178
{
179
WAV_Music *music = (WAV_Music *)context;
181
182
for (i = 0; i < music->numloops; ++i) {
WAVLoopPoint *loop = &music->loops[i];
183
184
185
loop->active = SDL_TRUE;
loop->current_play_count = loop->initial_play_count;
}
186
music->play_count = play_count;
187
188
189
190
if (SDL_RWseek(music->src, music->start, RW_SEEK_SET) < 0) {
return -1;
}
return 0;
191
192
}
193
194
/* Play some of a stream previously started with WAV_Play() */
static int WAV_GetSome(void *context, void *data, int bytes, SDL_bool *done)
195
{
196
WAV_Music *music = (WAV_Music *)context;
197
198
199
200
Sint64 pos, stop;
WAVLoopPoint *loop;
Sint64 loop_start;
Sint64 loop_stop;
201
SDL_bool looped = SDL_FALSE;
203
int filled, amount, result;
205
206
207
208
209
210
211
212
213
214
215
216
217
filled = SDL_AudioStreamGet(music->stream, data, bytes);
if (filled != 0) {
return filled;
}
if (!music->play_count) {
/* All done */
*done = SDL_TRUE;
return 0;
}
pos = SDL_RWtell(music->src);
stop = music->stop;
218
loop = NULL;
219
220
for (i = 0; i < music->numloops; ++i) {
loop = &music->loops[i];
221
if (loop->active) {
222
223
224
const int bytes_per_sample = (SDL_AUDIO_BITSIZE(music->spec.format) / 8) * music->spec.channels;
loop_start = music->start + loop->start * bytes_per_sample;
loop_stop = music->start + (loop->stop + 1) * bytes_per_sample;
225
226
227
228
if (pos >= loop_start && pos < loop_stop)
{
stop = loop_stop;
break;
230
231
232
233
}
loop = NULL;
}
234
235
236
237
238
239
240
241
242
amount = music->spec.size;
if ((stop - pos) < amount) {
amount = (int)(stop - pos);
}
amount = (int)SDL_RWread(music->src, music->buffer, 1, amount);
if (amount > 0) {
result = SDL_AudioStreamPut(music->stream, music->buffer, amount);
if (result < 0) {
return -1;
243
244
}
} else {
245
/* We might be looping, continue */
248
if (loop && SDL_RWtell(music->src) >= stop) {
249
250
if (loop->current_play_count == 1) {
loop->active = SDL_FALSE;
251
} else {
252
253
if (loop->current_play_count > 0) {
--loop->current_play_count;
255
256
SDL_RWseek(music->src, loop_start, RW_SEEK_SET);
looped = SDL_TRUE;
260
261
262
263
264
265
266
267
268
269
270
271
272
273
if (!looped && SDL_RWtell(music->src) >= music->stop) {
if (music->play_count == 1) {
music->play_count = 0;
SDL_AudioStreamFlush(music->stream);
} else {
int play_count = -1;
if (music->play_count > 0) {
play_count = (music->play_count - 1);
}
if (WAV_Play(music, play_count) < 0) {
return -1;
}
}
}
275
276
277
/* We'll get called again in the case where we looped or have more data */
return 0;
}
279
280
281
282
static int WAV_GetAudio(void *context, void *data, int bytes)
{
WAV_Music *music = (WAV_Music *)context;
return music_pcm_getaudio(context, data, bytes, music->volume, WAV_GetSome);
283
284
285
}
/* Close the given WAV stream */
286
static void WAV_Delete(void *context)
287
{
288
WAV_Music *music = (WAV_Music *)context;
290
/* Clean up associated data */
291
292
293
294
295
if (music->loops) {
SDL_free(music->loops);
}
if (music->stream) {
SDL_FreeAudioStream(music->stream);
297
298
if (music->buffer) {
SDL_free(music->buffer);
300
301
if (music->freesrc) {
SDL_RWclose(music->src);
303
SDL_free(music);
304
305
}
306
static SDL_bool ParseFMT(WAV_Music *wave, Uint32 chunk_length)
307
{
308
309
310
311
312
313
314
315
SDL_AudioSpec *spec = &wave->spec;
WaveFMT *format;
Uint8 *data;
SDL_bool loaded = SDL_FALSE;
if (chunk_length < sizeof(*format)) {
Mix_SetError("Wave format chunk too small");
return SDL_FALSE;
318
319
320
321
322
323
324
325
326
327
data = (Uint8 *)SDL_malloc(chunk_length);
if (!data) {
Mix_SetError("Out of memory");
return SDL_FALSE;
}
if (!SDL_RWread(wave->src, data, chunk_length, 1)) {
Mix_SetError("Couldn't read %d bytes from WAV file", chunk_length);
return SDL_FALSE;
}
format = (WaveFMT *)data;
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
/* Decode the audio data format */
switch (SDL_SwapLE16(format->encoding)) {
case PCM_CODE:
/* We can understand this */
break;
default:
Mix_SetError("Unknown WAVE data format");
goto done;
}
spec->freq = SDL_SwapLE32(format->frequency);
switch (SDL_SwapLE16(format->bitspersample)) {
case 8:
spec->format = AUDIO_U8;
break;
case 16:
spec->format = AUDIO_S16;
break;
default:
Mix_SetError("Unknown PCM data format");
goto done;
}
spec->channels = (Uint8) SDL_SwapLE16(format->channels);
spec->samples = 4096; /* Good default buffer size */
352
353
354
355
/* SDL_CalculateAudioSpec */
spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;
spec->size *= spec->channels;
spec->size *= spec->samples;
357
358
359
360
361
362
363
loaded = SDL_TRUE;
done:
SDL_free(data);
return loaded;
}
364
static SDL_bool ParseDATA(WAV_Music *wave, Uint32 chunk_length)
365
366
367
368
369
370
371
{
wave->start = SDL_RWtell(wave->src);
wave->stop = wave->start + chunk_length;
SDL_RWseek(wave->src, chunk_length, RW_SEEK_CUR);
return SDL_TRUE;
}
372
static SDL_bool AddLoopPoint(WAV_Music *wave, Uint32 play_count, Uint32 start, Uint32 stop)
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
{
WAVLoopPoint *loop;
WAVLoopPoint *loops = SDL_realloc(wave->loops, (wave->numloops + 1)*sizeof(*wave->loops));
if (!loops) {
Mix_SetError("Out of memory");
return SDL_FALSE;
}
loop = &loops[ wave->numloops ];
loop->start = start;
loop->stop = stop;
loop->initial_play_count = play_count;
loop->current_play_count = play_count;
wave->loops = loops;
++wave->numloops;
return SDL_TRUE;
}
392
static SDL_bool ParseSMPL(WAV_Music *wave, Uint32 chunk_length)
393
394
395
{
SamplerChunk *chunk;
Uint8 *data;
396
Uint32 i;
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
SDL_bool loaded = SDL_FALSE;
data = (Uint8 *)SDL_malloc(chunk_length);
if (!data) {
Mix_SetError("Out of memory");
return SDL_FALSE;
}
if (!SDL_RWread(wave->src, data, chunk_length, 1)) {
Mix_SetError("Couldn't read %d bytes from WAV file", chunk_length);
return SDL_FALSE;
}
chunk = (SamplerChunk *)data;
for (i = 0; i < SDL_SwapLE32(chunk->sample_loops); ++i) {
const Uint32 LOOP_TYPE_FORWARD = 0;
Uint32 loop_type = SDL_SwapLE32(chunk->loops[i].type);
if (loop_type == LOOP_TYPE_FORWARD) {
AddLoopPoint(wave, SDL_SwapLE32(chunk->loops[i].play_count), SDL_SwapLE32(chunk->loops[i].start), SDL_SwapLE32(chunk->loops[i].end));
416
417
418
419
420
421
422
}
loaded = SDL_TRUE;
SDL_free(data);
return loaded;
}
423
static SDL_bool LoadWAVMusic(WAV_Music *wave)
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
{
SDL_RWops *src = wave->src;
Uint32 chunk_type;
Uint32 chunk_length;
SDL_bool found_FMT = SDL_FALSE;
SDL_bool found_DATA = SDL_FALSE;
/* WAV magic header */
Uint32 wavelen;
Uint32 WAVEmagic;
/* Check the magic header */
wavelen = SDL_ReadLE32(src);
WAVEmagic = SDL_ReadLE32(src);
/* Read the chunks */
for (; ;) {
chunk_type = SDL_ReadLE32(src);
chunk_length = SDL_ReadLE32(src);
if (chunk_length == 0)
break;
switch (chunk_type)
{
case FMT:
found_FMT = SDL_TRUE;
if (!ParseFMT(wave, chunk_length))
return SDL_FALSE;
break;
case DATA:
found_DATA = SDL_TRUE;
if (!ParseDATA(wave, chunk_length))
return SDL_FALSE;
break;
case SMPL:
if (!ParseSMPL(wave, chunk_length))
return SDL_FALSE;
break;
default:
SDL_RWseek(src, chunk_length, RW_SEEK_CUR);
break;
}
468
469
470
471
if (!found_FMT) {
Mix_SetError("Bad WAV file (no FMT chunk)");
return SDL_FALSE;
473
474
475
476
477
478
479
if (!found_DATA) {
Mix_SetError("Bad WAV file (no DATA chunk)");
return SDL_FALSE;
}
return SDL_TRUE;
480
481
}
482
483
484
485
486
/* I couldn't get SANE_to_double() to work, so I stole this from libsndfile.
* I don't pretend to fully understand it.
*/
static Uint32 SANE_to_Uint32 (Uint8 *sanebuf)
487
{
488
489
490
/* Negative number? */
if (sanebuf[0] & 0x80)
return 0;
492
493
494
/* Less than 1? */
if (sanebuf[0] <= 0x3F)
return 1;
496
497
498
/* Way too big? */
if (sanebuf[0] > 0x40)
return 0x4000000;
500
501
502
/* Still too big? */
if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)
return 800000000;
504
505
return ((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7) |
(sanebuf[5] >> 1)) >> (29 - sanebuf[1]);
506
507
}
508
static SDL_bool LoadAIFFMusic(WAV_Music *wave)
509
{
510
511
512
513
SDL_RWops *src = wave->src;
SDL_AudioSpec *spec = &wave->spec;
SDL_bool found_SSND = SDL_FALSE;
SDL_bool found_COMM = SDL_FALSE;
514
515
516
Uint32 chunk_type;
Uint32 chunk_length;
517
Sint64 next_chunk;
518
519
520
521
522
523
524
525
526
527
528
529
530
531
/* AIFF magic header */
Uint32 AIFFmagic;
/* SSND chunk */
Uint32 offset;
Uint32 blocksize;
/* COMM format chunk */
Uint16 channels = 0;
Uint32 numsamples = 0;
Uint16 samplesize = 0;
Uint8 sane_freq[10];
Uint32 frequency = 0;
/* Check the magic header */
532
533
534
chunk_length = SDL_ReadBE32(src);
AIFFmagic = SDL_ReadLE32(src);
if (AIFFmagic != AIFF) {
535
Mix_SetError("Unrecognized file type (not AIFF)");
536
return SDL_FALSE;
537
538
539
}
/* From what I understand of the specification, chunks may appear in
540
* any order, and we should just ignore unknown ones.
541
542
543
*
* TODO: Better sanity-checking. E.g. what happens if the AIFF file
* contains compressed sound data?
545
546
547
548
do {
chunk_type = SDL_ReadLE32(src);
chunk_length = SDL_ReadBE32(src);
next_chunk = SDL_RWtell(src) + chunk_length;
550
551
/* Paranoia to avoid infinite loops */
if (chunk_length == 0)
554
switch (chunk_type) {
555
case SSND:
556
557
558
559
found_SSND = SDL_TRUE;
offset = SDL_ReadBE32(src);
blocksize = SDL_ReadBE32(src);
wave->start = SDL_RWtell(src) + offset;
560
561
562
break;
case COMM:
563
found_COMM = SDL_TRUE;
564
565
/* Read the audio data format chunk */
566
567
568
channels = SDL_ReadBE16(src);
numsamples = SDL_ReadBE32(src);
samplesize = SDL_ReadBE16(src);
569
SDL_RWread(src, sane_freq, sizeof(sane_freq), 1);
570
frequency = SANE_to_Uint32(sane_freq);
571
572
573
574
575
576
577
578
579
580
break;
default:
break;
}
} while ((!found_SSND || !found_COMM)
&& SDL_RWseek(src, next_chunk, RW_SEEK_SET) != -1);
if (!found_SSND) {
Mix_SetError("Bad AIFF file (no SSND chunk)");
581
return SDL_FALSE;
582
583
584
585
}
if (!found_COMM) {
Mix_SetError("Bad AIFF file (no COMM chunk)");
586
return SDL_FALSE;
589
wave->stop = wave->start + channels * numsamples * (samplesize / 8);
590
591
/* Decode the audio data format */
592
SDL_memset(spec, 0, (sizeof *spec));
593
594
595
596
597
598
599
600
601
602
spec->freq = frequency;
switch (samplesize) {
case 8:
spec->format = AUDIO_S8;
break;
case 16:
spec->format = AUDIO_S16MSB;
break;
default:
Mix_SetError("Unknown samplesize in data format");
603
return SDL_FALSE;
604
605
606
}
spec->channels = (Uint8) channels;
spec->samples = 4096; /* Good default buffer size */
608
return SDL_TRUE;
609
}
611
612
613
614
615
616
617
618
619
620
Mix_MusicInterface Mix_MusicInterface_WAV =
{
"WAVE",
MIX_MUSIC_WAVE,
MUS_WAV,
SDL_FALSE,
SDL_FALSE,
NULL, /* Load */
NULL, /* Open */
621
WAV_CreateFromRW,
622
NULL, /* CreateFromFile */
623
624
WAV_SetVolume,
WAV_Play,
625
NULL, /* IsPlaying */
627
628
629
630
NULL, /* Seek */
NULL, /* Pause */
NULL, /* Resume */
NULL, /* Stop */
632
633
634
635
636
637
638
NULL, /* Close */
NULL, /* Unload */
};
#endif /* MUSIC_WAV */
/* vi: set ts=4 sw=4 expandtab: */