/
music_mad.c
469 lines (387 loc) · 14.1 KB
2
SDL_mixer: An audio mixer library based on the SDL library
3
Copyright (C) 1997-2019 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.
22
#ifdef MUSIC_MP3_MAD
25
#include "mp3utils.h"
27
28
#include "mad.h"
29
30
31
32
33
34
35
36
37
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
90
91
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
123
124
125
126
127
128
129
/* NOTE: The dithering functions are GPL, which should be fine if your
application is GPL (which would need to be true if you enabled
libmad support in SDL_mixer). If you're using libmad under the
commercial license, you need to disable this code.
*/
/************************ dithering functions ***************************/
#ifdef MUSIC_MP3_MAD_GPL_DITHERING
/* All dithering done here is taken from the GPL'ed xmms-mad plugin. */
/* Copyright (C) 1997 Makoto Matsumoto and Takuji Nishimura. */
/* Any feedback is very welcome. For any question, comments, */
/* see http://www.math.keio.ac.jp/matumoto/emt.html or email */
/* matumoto@math.keio.ac.jp */
/* Period parameters */
#define MP3_DITH_N 624
#define MP3_DITH_M 397
#define MATRIX_A 0x9908b0df /* constant vector a */
#define UPPER_MASK 0x80000000 /* most significant w-r bits */
#define LOWER_MASK 0x7fffffff /* least significant r bits */
/* Tempering parameters */
#define TEMPERING_MASK_B 0x9d2c5680
#define TEMPERING_MASK_C 0xefc60000
#define TEMPERING_SHIFT_U(y) (y >> 11)
#define TEMPERING_SHIFT_S(y) (y << 7)
#define TEMPERING_SHIFT_T(y) (y << 15)
#define TEMPERING_SHIFT_L(y) (y >> 18)
static unsigned long mt[MP3_DITH_N]; /* the array for the state vector */
static int mti=MP3_DITH_N+1; /* mti==MP3_DITH_N+1 means mt[MP3_DITH_N] is not initialized */
/* initializing the array with a NONZERO seed */
static void sgenrand(unsigned long seed)
{
/* setting initial seeds to mt[MP3_DITH_N] using */
/* the generator Line 25 of Table 1 in */
/* [KNUTH 1981, The Art of Computer Programming */
/* Vol. 2 (2nd Ed.), pp102] */
mt[0]= seed & 0xffffffff;
for (mti=1; mti<MP3_DITH_N; mti++)
mt[mti] = (69069 * mt[mti-1]) & 0xffffffff;
}
static unsigned long genrand(void)
{
unsigned long y;
static unsigned long mag01[2]={0x0, MATRIX_A};
/* mag01[x] = x * MATRIX_A for x=0,1 */
if (mti >= MP3_DITH_N) { /* generate MP3_DITH_N words at one time */
int kk;
if (mti == MP3_DITH_N+1) /* if sgenrand() has not been called, */
sgenrand(4357); /* a default initial seed is used */
for (kk=0;kk<MP3_DITH_N-MP3_DITH_M;kk++) {
y = (mt[kk]&UPPER_MASK)|(mt[kk+1]&LOWER_MASK);
mt[kk] = mt[kk+MP3_DITH_M] ^ (y >> 1) ^ mag01[y & 0x1];
}
for (;kk<MP3_DITH_N-1;kk++) {
y = (mt[kk]&UPPER_MASK)|(mt[kk+1]&LOWER_MASK);
mt[kk] = mt[kk+(MP3_DITH_M-MP3_DITH_N)] ^ (y >> 1) ^ mag01[y & 0x1];
}
y = (mt[MP3_DITH_N-1]&UPPER_MASK)|(mt[0]&LOWER_MASK);
mt[MP3_DITH_N-1] = mt[MP3_DITH_M-1] ^ (y >> 1) ^ mag01[y & 0x1];
mti = 0;
}
y = mt[mti++];
y ^= TEMPERING_SHIFT_U(y);
y ^= TEMPERING_SHIFT_S(y) & TEMPERING_MASK_B;
y ^= TEMPERING_SHIFT_T(y) & TEMPERING_MASK_C;
y ^= TEMPERING_SHIFT_L(y);
return y;
}
static long triangular_dither_noise(int nbits) {
/* parameter nbits : the peak-to-peak amplitude desired (in bits)
* use with nbits set to 2 + nber of bits to be trimmed.
* (because triangular is made from two uniformly distributed processes,
* it starts at 2 bits peak-to-peak amplitude)
* see The Theory of Dithered Quantization by Robert Alexander Wannamaker
* for complete proof of why that's optimal
*/
long v = (genrand()/2 - genrand()/2); /* in ]-2^31, 2^31[ */
long P = 1 << (32 - nbits); /* the power of 2 */
v /= P;
/* now v in ]-2^(nbits-1), 2^(nbits-1) [ */
return v;
}
#endif /* MUSIC_MP3_MAD_GPL_DITHERING */
130
131
132
133
134
#define MAD_INPUT_BUFFER_SIZE (5*8192)
enum {
MS_input_eof = 0x0001,
MS_input_error = 0x0001,
135
MS_decode_error = 0x0002,
136
MS_error_flags = 0x000f
137
138
139
};
typedef struct {
140
struct mp3file_t mp3file;
141
int play_count;
142
143
144
145
146
147
148
int freesrc;
struct mad_stream stream;
struct mad_frame frame;
struct mad_synth synth;
mad_timer_t next_frame_start;
int volume;
int status;
149
SDL_AudioStream *audiostream;
150
151
unsigned char input_buffer[MAD_INPUT_BUFFER_SIZE + MAD_BUFFER_GUARD];
152
} MAD_Music;
153
154
155
156
static int MAD_Seek(void *context, double position);
157
static void *MAD_CreateFromRW(SDL_RWops *src, int freesrc)
158
{
159
160
161
162
163
164
MAD_Music *music;
music = (MAD_Music *)SDL_calloc(1, sizeof(MAD_Music));
if (!music) {
SDL_OutOfMemory();
return NULL;
165
}
166
music->mp3file.src = src;
167
168
music->volume = MIX_MAX_VOLUME;
169
170
music->mp3file.length = SDL_RWsize(src);
if (mp3_skiptags(&music->mp3file) < 0) {
171
SDL_free(music);
172
Mix_SetError("music_mad: corrupt mp3 file (bad tags.)");
173
174
175
return NULL;
}
176
177
178
179
180
181
182
mad_stream_init(&music->stream);
mad_frame_init(&music->frame);
mad_synth_init(&music->synth);
mad_timer_reset(&music->next_frame_start);
music->freesrc = freesrc;
return music;
185
static void MAD_SetVolume(void *context, int volume)
186
{
187
188
MAD_Music *music = (MAD_Music *)context;
music->volume = volume;
192
static int MAD_Play(void *context, int play_count)
193
{
194
195
196
MAD_Music *music = (MAD_Music *)context;
music->play_count = play_count;
return MAD_Seek(music, 0.0);
199
200
201
202
203
/* Reads the next frame from the file.
Returns true on success or false on failure.
*/
static SDL_bool read_next_frame(MAD_Music *music)
204
{
205
206
if (music->stream.buffer == NULL ||
music->stream.error == MAD_ERROR_BUFLEN) {
207
208
209
210
211
212
213
size_t read_size;
size_t remaining;
unsigned char *read_start;
/* There might be some bytes in the buffer left over from last
time. If so, move them down and read more bytes following
them. */
214
215
216
217
if (music->stream.next_frame != NULL) {
remaining = music->stream.bufend - music->stream.next_frame;
memmove(music->input_buffer, music->stream.next_frame, remaining);
read_start = music->input_buffer + remaining;
218
219
220
221
read_size = MAD_INPUT_BUFFER_SIZE - remaining;
} else {
read_size = MAD_INPUT_BUFFER_SIZE;
222
read_start = music->input_buffer;
223
224
remaining = 0;
}
225
226
/* Now read additional bytes from the input file. */
227
read_size = MP3_RWread(&music->mp3file, read_start, 1, read_size);
228
229
if (read_size == 0) {
230
if ((music->status & (MS_input_eof | MS_input_error)) == 0) {
231
/* FIXME: how to detect error? */
232
music->status |= MS_input_eof;
233
234
235
236
237
238
239
/* At the end of the file, we must stuff MAD_BUFFER_GUARD
number of 0 bytes. */
SDL_memset(read_start + read_size, 0, MAD_BUFFER_GUARD);
read_size += MAD_BUFFER_GUARD;
}
}
240
241
/* Now feed those bytes into the libmad stream. */
242
mad_stream_buffer(&music->stream, music->input_buffer,
243
read_size + remaining);
244
music->stream.error = MAD_ERROR_NONE;
245
}
246
247
248
/* Now ask libmad to extract a frame from the data we just put in
its buffer. */
249
250
if (mad_frame_decode(&music->frame, &music->stream)) {
if (MAD_RECOVERABLE(music->stream.error)) {
251
mad_stream_sync(&music->stream); /* to frame seek mode */
252
return SDL_FALSE;
253
254
255
} else if (music->stream.error == MAD_ERROR_BUFLEN) {
return SDL_FALSE;
256
257
} else {
258
259
260
Mix_SetError("mad_frame_decode() failed, corrupt stream?");
music->status |= MS_decode_error;
return SDL_FALSE;
261
}
262
263
}
264
mad_timer_add(&music->next_frame_start, music->frame.header.duration);
266
return SDL_TRUE;
270
271
272
273
static Sint16 scale(mad_fixed_t sample)
{
const int n_bits_to_loose = MAD_F_FRACBITS + 1 - 16;
274
/* round */
275
276
277
278
279
sample += (1L << (n_bits_to_loose - 1));
#ifdef MUSIC_MP3_MAD_GPL_DITHERING
sample += triangular_dither_noise(n_bits_to_loose + 1);
#endif
281
282
283
284
285
/* clip */
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
287
/* quantize */
288
return (Sint16)(sample >> n_bits_to_loose);
291
/* Once the frame has been read, copies its samples into the output buffer. */
292
293
static SDL_bool decode_frame(MAD_Music *music)
{
294
struct mad_pcm *pcm;
295
unsigned int i, nchannels, nsamples;
296
mad_fixed_t const *left_ch, *right_ch;
297
298
Sint16 *buffer, *dst;
int result;
299
300
301
mad_synth_frame(&music->synth, &music->frame);
pcm = &music->synth.pcm;
302
303
if (!music->audiostream) {
304
music->audiostream = SDL_NewAudioStream(AUDIO_S16, (Uint8)pcm->channels, (int)pcm->samplerate, music_spec.format, music_spec.channels, music_spec.freq);
305
306
307
308
if (!music->audiostream) {
return SDL_FALSE;
}
}
309
310
311
312
313
314
315
316
317
nchannels = pcm->channels;
nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
buffer = SDL_stack_alloc(Sint16, nsamples*nchannels);
if (!buffer) {
SDL_OutOfMemory();
return SDL_FALSE;
318
319
}
320
321
322
323
324
325
326
327
328
dst = buffer;
if (nchannels == 1) {
for (i = nsamples; i--;) {
*dst++ = scale(*left_ch++);
}
} else {
for (i = nsamples; i--;) {
*dst++ = scale(*left_ch++);
*dst++ = scale(*right_ch++);
329
330
}
}
332
result = SDL_AudioStreamPut(music->audiostream, buffer, (int)(nsamples * nchannels * sizeof(Sint16)));
333
SDL_stack_free(buffer);
335
336
337
338
339
if (result < 0) {
return SDL_FALSE;
}
return SDL_TRUE;
}
341
342
343
344
static int MAD_GetSome(void *context, void *data, int bytes, SDL_bool *done)
{
MAD_Music *music = (MAD_Music *)context;
int filled;
346
347
348
349
if (music->audiostream) {
filled = SDL_AudioStreamGet(music->audiostream, data, bytes);
if (filled != 0) {
return filled;
350
}
353
354
355
if (!music->play_count) {
/* All done */
*done = SDL_TRUE;
356
357
358
return 0;
}
359
360
361
if (read_next_frame(music)) {
if (!decode_frame(music)) {
return -1;
362
}
363
} else if (music->status & MS_input_eof) {
364
365
366
367
368
369
int play_count = -1;
if (music->play_count > 0) {
play_count = (music->play_count - 1);
}
if (MAD_Play(music, play_count) < 0) {
return -1;
370
}
371
372
} else if (music->status & MS_decode_error) {
return -1;
373
}
374
375
return 0;
}
376
377
378
379
380
381
static int MAD_GetAudio(void *context, void *data, int bytes)
{
MAD_Music *music = (MAD_Music *)context;
return music_pcm_getaudio(context, data, bytes, music->volume, MAD_GetSome);
}
382
383
384
static int MAD_Seek(void *context, double position)
{
385
MAD_Music *music = (MAD_Music *)context;
386
387
388
389
mad_timer_t target;
int int_part;
int_part = (int)position;
390
mad_timer_set(&target, (unsigned long)int_part, (unsigned long)((position - int_part) * 1000000), 1000000);
391
392
if (mad_timer_compare(music->next_frame_start, target) > 0) {
393
394
395
396
397
398
/* In order to seek backwards in a VBR file, we have to rewind and
start again from the beginning. This isn't necessary if the
file happens to be CBR, of course; in that case we could seek
directly to the frame we want. But I leave that little
optimization for the future developer who discovers she really
needs it. */
399
400
mad_timer_reset(&music->next_frame_start);
music->status &= ~MS_error_flags;
401
402
MP3_RWseek(&music->mp3file, 0, RW_SEEK_SET);
403
}
405
406
/* Now we have to skip frames until we come to the right one.
Again, only truly necessary if the file is VBR. */
407
408
409
while (mad_timer_compare(music->next_frame_start, target) < 0) {
if (!read_next_frame(music)) {
if ((music->status & MS_error_flags) != 0) {
410
411
412
413
414
/* Couldn't read a frame; either an error condition or
end-of-file. Stop. */
return Mix_SetError("Seek position out of range");
}
}
415
}
417
418
419
420
421
/* Here we are, at the beginning of the frame that contains the
target time. Ehh, I say that's close enough. If we wanted to,
we could get more precise by decoding the frame now and counting
the appropriate number of samples out of it. */
return 0;
424
425
static void MAD_Delete(void *context)
{
426
MAD_Music *music = (MAD_Music *)context;
428
429
430
431
432
433
434
435
mad_stream_finish(&music->stream);
mad_frame_finish(&music->frame);
mad_synth_finish(&music->synth);
if (music->audiostream) {
SDL_FreeAudioStream(music->audiostream);
}
if (music->freesrc) {
436
SDL_RWclose(music->mp3file.src);
437
}
438
SDL_free(music);
439
440
441
442
443
444
445
446
447
448
}
Mix_MusicInterface Mix_MusicInterface_MAD =
{
"MAD",
MIX_MUSIC_MAD,
MUS_MP3,
SDL_FALSE,
SDL_FALSE,
449
450
NULL, /* Load */
NULL, /* Open */
451
MAD_CreateFromRW,
452
NULL, /* CreateFromFile */
453
454
MAD_SetVolume,
MAD_Play,
455
NULL, /* IsPlaying */
456
457
MAD_GetAudio,
MAD_Seek,
458
NULL /* Duration */
459
460
461
NULL, /* Pause */
NULL, /* Resume */
NULL, /* Stop */
462
MAD_Delete,
463
NULL, /* Close */
464
NULL /* Unload */
465
466
467
468
469
};
#endif /* MUSIC_MP3_MAD */
/* vi: set ts=4 sw=4 expandtab: */