/
wavestream.c
536 lines (470 loc) · 12.7 KB
1
/*
2
3
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library 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
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
20
slouken@libsdl.org
21
22
*/
23
/* $Id$ */
24
25
26
27
28
29
/* This file supports streaming WAV files, without volume adjustment */
#include <stdlib.h>
#include <string.h>
30
31
32
33
#include "SDL_audio.h"
#include "SDL_mutex.h"
#include "SDL_rwops.h"
#include "SDL_endian.h"
34
35
#include "SDL_mixer.h"
36
37
#include "wavestream.h"
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
/*
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 */
/*******************************************/
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
#define PCM_CODE 1
#define ADPCM_CODE 2
#define WAVE_MONO 1
#define WAVE_STEREO 2
/* Normally, these three chunks come consecutively in a WAVE file */
typedef struct WaveFMT {
/* Not saved in the chunk we read:
Uint32 FMTchunk;
Uint32 fmtlen;
*/
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 */
} WaveFMT;
/* The general chunk found in the WAVE file */
typedef struct Chunk {
Uint32 magic;
Uint32 length;
Uint8 *data; /* Data includes magic and length */
} Chunk;
/*********************************************/
/* Define values for AIFF (IFF audio) format */
/*********************************************/
#define FORM 0x4d524f46 /* "FORM" */
#define AIFF 0x46464941 /* "AIFF" */
#define SSND 0x444e5353 /* "SSND" */
#define COMM 0x4d4d4f43 /* "COMM" */
90
/* Currently we only support a single stream at a time */
91
static WAVStream *music = NULL;
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
/* This is the format of the audio mixer data */
static SDL_AudioSpec mixer;
/* Function to load the WAV/AIFF stream */
static FILE *LoadWAVStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop);
static FILE *LoadAIFFStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop);
/* Initialize the WAVStream player, with the given mixer settings
This function returns 0, or -1 if there was an error.
*/
int WAVStream_Init(SDL_AudioSpec *mixerfmt)
{
mixer = *mixerfmt;
return(0);
}
/* Unimplemented */
extern void WAVStream_SetVolume(int volume)
{
}
/* Load a WAV stream from the given file */
extern WAVStream *WAVStream_LoadSong(const char *file, const char *magic)
{
WAVStream *wave;
SDL_AudioSpec wavespec;
if ( ! mixer.format ) {
123
Mix_SetError("WAV music output not started");
124
125
126
127
128
129
130
131
132
133
134
135
return(NULL);
}
wave = (WAVStream *)malloc(sizeof *wave);
if ( wave ) {
memset(wave, 0, (sizeof *wave));
if ( strcmp(magic, "RIFF") == 0 ) {
wave->wavefp = LoadWAVStream(file, &wavespec,
&wave->start, &wave->stop);
} else
if ( strcmp(magic, "FORM") == 0 ) {
wave->wavefp = LoadAIFFStream(file, &wavespec,
&wave->start, &wave->stop);
136
137
} else {
Mix_SetError("Unknown WAVE format");
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
}
if ( wave->wavefp == NULL ) {
free(wave);
return(NULL);
}
SDL_BuildAudioCVT(&wave->cvt,
wavespec.format, wavespec.channels, wavespec.freq,
mixer.format, mixer.channels, mixer.freq);
}
return(wave);
}
/* Start playback of a given WAV stream */
extern void WAVStream_Start(WAVStream *wave)
{
clearerr(wave->wavefp);
fseek(wave->wavefp, wave->start, SEEK_SET);
155
music = wave;
156
157
}
158
/* Play some of a stream previously started with WAVStream_Start() */
159
extern void WAVStream_PlaySome(Uint8 *stream, int len)
160
161
162
{
long pos;
163
164
if ( music && ((pos=ftell(music->wavefp)) < music->stop) ) {
if ( music->cvt.needed ) {
165
166
int original_len;
167
168
original_len=(int)((double)len/music->cvt.len_ratio);
if ( music->cvt.len != original_len ) {
169
int worksize;
170
171
if ( music->cvt.buf != NULL ) {
free(music->cvt.buf);
172
}
173
174
175
worksize = original_len*music->cvt.len_mult;
music->cvt.buf=(Uint8 *)malloc(worksize);
if ( music->cvt.buf == NULL ) {
176
177
return;
}
178
music->cvt.len = original_len;
179
}
180
181
if ( (music->stop - pos) < original_len ) {
original_len = (music->stop - pos);
182
}
183
original_len = fread(music->cvt.buf,1,original_len,music->wavefp);
184
185
186
187
188
189
/* At least at the time of writing, SDL_ConvertAudio()
does byte-order swapping starting at the end of the
buffer. Thus, if we are reading 16-bit samples, we
had better make damn sure that we get an even
number of bytes, or we'll get garbage.
*/
190
if ( (music->cvt.src_format & 0x0010) && (original_len & 1) ) {
191
192
original_len--;
}
193
194
195
music->cvt.len = original_len;
SDL_ConvertAudio(&music->cvt);
memcpy(stream, music->cvt.buf, music->cvt.len_cvt);
196
} else {
197
198
if ( (music->stop - pos) < len ) {
len = (music->stop - pos);
199
}
200
fread(stream, len, 1, music->wavefp);
201
202
203
204
205
206
207
}
}
}
/* Stop playback of a stream previously started with WAVStream_Start() */
extern void WAVStream_Stop(void)
{
208
music = NULL;
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
}
/* Close the given WAV stream */
extern void WAVStream_FreeSong(WAVStream *wave)
{
if ( wave ) {
/* Clean up associated data */
if ( wave->wavefp ) {
fclose(wave->wavefp);
}
if ( wave->cvt.buf ) {
free(wave->cvt.buf);
}
free(wave);
}
}
/* Return non-zero if a stream is currently playing */
227
extern int WAVStream_Active(void)
228
229
230
231
{
int active;
active = 0;
232
if ( music && (ftell(music->wavefp) < music->stop) ) {
233
234
235
236
237
238
239
240
241
242
243
244
active = 1;
}
return(active);
}
static int ReadChunk(SDL_RWops *src, Chunk *chunk, int read_data)
{
chunk->magic = SDL_ReadLE32(src);
chunk->length = SDL_ReadLE32(src);
if ( read_data ) {
chunk->data = (Uint8 *)malloc(chunk->length);
if ( chunk->data == NULL ) {
245
Mix_SetError("Out of memory");
246
247
248
return(-1);
}
if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {
249
Mix_SetError("Couldn't read chunk");
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
free(chunk->data);
return(-1);
}
} else {
SDL_RWseek(src, chunk->length, SEEK_CUR);
}
return(chunk->length);
}
static FILE *LoadWAVStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop)
{
int was_error;
FILE *wavefp;
SDL_RWops *src;
Chunk chunk;
int lenread;
/* WAV magic header */
Uint32 RIFFchunk;
Uint32 wavelen;
Uint32 WAVEmagic;
/* FMT chunk */
WaveFMT *format = NULL;
/* Make sure we are passed a valid data source */
was_error = 0;
wavefp = fopen(file, "rb");
src = NULL;
if ( wavefp ) {
src = SDL_RWFromFP(wavefp, 0);
}
if ( src == NULL ) {
was_error = 1;
goto done;
}
287
288
289
290
291
292
/* Check the magic header */
RIFFchunk = SDL_ReadLE32(src);
wavelen = SDL_ReadLE32(src);
WAVEmagic = SDL_ReadLE32(src);
if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) {
293
Mix_SetError("Unrecognized file type (not WAVE)");
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
was_error = 1;
goto done;
}
/* Read the audio data format chunk */
chunk.data = NULL;
do {
/* FIXME! Add this logic to SDL_LoadWAV_RW() */
if ( chunk.data ) {
free(chunk.data);
}
lenread = ReadChunk(src, &chunk, 1);
if ( lenread < 0 ) {
was_error = 1;
goto done;
}
} while ( (chunk.magic == FACT) || (chunk.magic == LIST) );
/* Decode the audio data format */
format = (WaveFMT *)chunk.data;
if ( chunk.magic != FMT ) {
free(chunk.data);
316
Mix_SetError("Complex WAVE files not supported");
317
318
319
320
321
322
323
324
was_error = 1;
goto done;
}
switch (SDL_SwapLE16(format->encoding)) {
case PCM_CODE:
/* We can understand this */
break;
default:
325
Mix_SetError("Unknown WAVE data format");
326
327
328
329
330
331
332
333
334
335
336
337
338
was_error = 1;
goto done;
}
memset(spec, 0, (sizeof *spec));
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:
339
Mix_SetError("Unknown PCM data format");
340
341
342
was_error = 1;
goto done;
}
343
spec->channels = (Uint8) SDL_SwapLE16(format->channels);
344
345
346
347
348
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
spec->samples = 4096; /* Good default buffer size */
/* Set the file offset to the DATA chunk data */
chunk.data = NULL;
do {
*start = SDL_RWtell(src) + 2*sizeof(Uint32);
lenread = ReadChunk(src, &chunk, 0);
if ( lenread < 0 ) {
was_error = 1;
goto done;
}
} while ( chunk.magic != DATA );
*stop = SDL_RWtell(src);
done:
if ( format != NULL ) {
free(format);
}
if ( src ) {
SDL_RWclose(src);
}
if ( was_error ) {
if ( wavefp ) {
fclose(wavefp);
wavefp = NULL;
}
}
return(wavefp);
}
374
375
376
377
378
/* 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)
379
{
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
/* Negative number? */
if (sanebuf[0] & 0x80)
return 0;
/* Less than 1? */
if (sanebuf[0] <= 0x3F)
return 1;
/* Way too big? */
if (sanebuf[0] > 0x40)
return 0x4000000;
/* Still too big? */
if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)
return 800000000;
return ((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7)
| (sanebuf[5] >> 1)) >> (29 - sanebuf[1]);
398
399
400
401
402
403
}
static FILE *LoadAIFFStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop)
{
int was_error;
404
405
int found_SSND;
int found_COMM;
406
407
408
FILE *wavefp;
SDL_RWops *src;
409
410
411
412
Uint32 chunk_type;
Uint32 chunk_length;
long next_chunk;
413
414
415
416
417
418
419
/* AIFF magic header */
Uint32 FORMchunk;
Uint32 AIFFmagic;
/* SSND chunk */
Uint32 offset;
Uint32 blocksize;
/* COMM format chunk */
420
421
422
Uint16 channels = 0;
Uint32 numsamples = 0;
Uint16 samplesize = 0;
423
Uint8 sane_freq[10];
424
Uint32 frequency = 0;
425
426
427
428
429
430
431
432
433
434
435
436
437
/* Make sure we are passed a valid data source */
was_error = 0;
wavefp = fopen(file, "rb");
src = NULL;
if ( wavefp ) {
src = SDL_RWFromFP(wavefp, 0);
}
if ( src == NULL ) {
was_error = 1;
goto done;
}
438
439
440
/* Check the magic header */
FORMchunk = SDL_ReadLE32(src);
441
chunk_length = SDL_ReadBE32(src);
442
443
AIFFmagic = SDL_ReadLE32(src);
if ( (FORMchunk != FORM) || (AIFFmagic != AIFF) ) {
444
Mix_SetError("Unrecognized file type (not AIFF)");
445
446
447
448
was_error = 1;
goto done;
}
449
450
/* From what I understand of the specification, chunks may appear in
* any order, and we should just ignore unknown ones.
451
452
453
*
* TODO: Better sanity-checking. E.g. what happens if the AIFF file
* contains compressed sound data?
454
*/
455
456
457
found_SSND = 0;
found_COMM = 0;
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
do {
chunk_type = SDL_ReadLE32(src);
chunk_length = SDL_ReadBE32(src);
next_chunk = SDL_RWtell(src) + chunk_length;
/* Paranoia to avoid infinite loops */
if (chunk_length == 0)
break;
switch (chunk_type) {
case SSND:
found_SSND = 1;
offset = SDL_ReadBE32(src);
blocksize = SDL_ReadBE32(src);
*start = SDL_RWtell(src) + offset;
break;
case COMM:
found_COMM = 1;
/* Read the audio data format chunk */
channels = SDL_ReadBE16(src);
numsamples = SDL_ReadBE32(src);
samplesize = SDL_ReadBE16(src);
SDL_RWread(src, sane_freq, sizeof(sane_freq), 1);
frequency = SANE_to_Uint32(sane_freq);
break;
default:
break;
}
} while ((!found_SSND || !found_COMM)
&& SDL_RWseek(src, next_chunk, SEEK_SET) != -1);
if (!found_SSND) {
494
Mix_SetError("Bad AIFF file (no SSND chunk)");
495
496
497
498
499
was_error = 1;
goto done;
}
if (!found_COMM) {
500
Mix_SetError("Bad AIFF file (no COMM chunk)");
501
502
was_error = 1;
goto done;
503
}
504
505
*stop = *start + channels * numsamples * (samplesize / 8);
506
507
508
509
510
511
/* Decode the audio data format */
memset(spec, 0, (sizeof *spec));
spec->freq = frequency;
switch (samplesize) {
case 8:
512
spec->format = AUDIO_S8;
513
514
break;
case 16:
515
spec->format = AUDIO_S16MSB;
516
517
break;
default:
518
Mix_SetError("Unknown samplesize in data format");
519
520
521
was_error = 1;
goto done;
}
522
spec->channels = (Uint8) channels;
523
524
525
526
527
528
529
530
531
532
533
534
535
536
spec->samples = 4096; /* Good default buffer size */
done:
if ( src ) {
SDL_RWclose(src);
}
if ( was_error ) {
if ( wavefp ) {
fclose(wavefp);
wavefp = NULL;
}
}
return(wavefp);
}