mpegaudiodec.c
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1 /*
2  * MPEG Audio decoder
3  * Copyright (c) 2001, 2002 Fabrice Bellard
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
27 #include "libavutil/audioconvert.h"
28 #include "avcodec.h"
29 #include "get_bits.h"
30 #include "mathops.h"
31 #include "mpegaudiodsp.h"
32 
33 /*
34  * TODO:
35  * - test lsf / mpeg25 extensively.
36  */
37 
38 #include "mpegaudio.h"
39 #include "mpegaudiodecheader.h"
40 
41 #define BACKSTEP_SIZE 512
42 #define EXTRABYTES 24
43 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
44 
45 /* layer 3 "granule" */
46 typedef struct GranuleDef {
47  uint8_t scfsi;
52  uint8_t block_type;
53  uint8_t switch_point;
54  int table_select[3];
55  int subblock_gain[3];
56  uint8_t scalefac_scale;
58  int region_size[3]; /* number of huffman codes in each region */
59  int preflag;
60  int short_start, long_end; /* long/short band indexes */
61  uint8_t scale_factors[40];
62  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
63 } GranuleDef;
64 
65 typedef struct MPADecodeContext {
69  /* next header (used in free format parsing) */
76  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
77  GranuleDef granules[2][2]; /* Used in Layer 3 */
78  int adu_mode;
85 
86 #if CONFIG_FLOAT
87 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
88 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
89 # define FIXR(x) ((float)(x))
90 # define FIXHR(x) ((float)(x))
91 # define MULH3(x, y, s) ((s)*(y)*(x))
92 # define MULLx(x, y, s) ((y)*(x))
93 # define RENAME(a) a ## _float
94 # define OUT_FMT AV_SAMPLE_FMT_FLT
95 #else
96 # define SHR(a,b) ((a)>>(b))
97 /* WARNING: only correct for positive numbers */
98 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
99 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
100 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
101 # define MULH3(x, y, s) MULH((s)*(x), y)
102 # define MULLx(x, y, s) MULL(x,y,s)
103 # define RENAME(a) a ## _fixed
104 # define OUT_FMT AV_SAMPLE_FMT_S16
105 #endif
106 
107 /****************/
108 
109 #define HEADER_SIZE 4
110 
111 #include "mpegaudiodata.h"
112 #include "mpegaudiodectab.h"
113 
114 /* vlc structure for decoding layer 3 huffman tables */
115 static VLC huff_vlc[16];
117  0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
118  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
119  ][2];
120 static const int huff_vlc_tables_sizes[16] = {
121  0, 128, 128, 128, 130, 128, 154, 166,
122  142, 204, 190, 170, 542, 460, 662, 414
123 };
124 static VLC huff_quad_vlc[2];
125 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
126 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
127 /* computed from band_size_long */
128 static uint16_t band_index_long[9][23];
129 #include "mpegaudio_tablegen.h"
130 /* intensity stereo coef table */
131 static INTFLOAT is_table[2][16];
132 static INTFLOAT is_table_lsf[2][2][16];
133 static INTFLOAT csa_table[8][4];
134 
135 static int16_t division_tab3[1<<6 ];
136 static int16_t division_tab5[1<<8 ];
137 static int16_t division_tab9[1<<11];
138 
139 static int16_t * const division_tabs[4] = {
141 };
142 
143 /* lower 2 bits: modulo 3, higher bits: shift */
144 static uint16_t scale_factor_modshift[64];
145 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
146 static int32_t scale_factor_mult[15][3];
147 /* mult table for layer 2 group quantization */
148 
149 #define SCALE_GEN(v) \
150 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
151 
152 static const int32_t scale_factor_mult2[3][3] = {
153  SCALE_GEN(4.0 / 3.0), /* 3 steps */
154  SCALE_GEN(4.0 / 5.0), /* 5 steps */
155  SCALE_GEN(4.0 / 9.0), /* 9 steps */
156 };
157 
163 {
164  int i, k, j = 0;
165  g->region_size[2] = 576 / 2;
166  for (i = 0; i < 3; i++) {
167  k = FFMIN(g->region_size[i], g->big_values);
168  g->region_size[i] = k - j;
169  j = k;
170  }
171 }
172 
174 {
175  if (g->block_type == 2)
176  g->region_size[0] = (36 / 2);
177  else {
178  if (s->sample_rate_index <= 2)
179  g->region_size[0] = (36 / 2);
180  else if (s->sample_rate_index != 8)
181  g->region_size[0] = (54 / 2);
182  else
183  g->region_size[0] = (108 / 2);
184  }
185  g->region_size[1] = (576 / 2);
186 }
187 
188 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
189 {
190  int l;
191  g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
192  /* should not overflow */
193  l = FFMIN(ra1 + ra2 + 2, 22);
194  g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
195 }
196 
198 {
199  if (g->block_type == 2) {
200  if (g->switch_point) {
201  /* if switched mode, we handle the 36 first samples as
202  long blocks. For 8000Hz, we handle the 48 first
203  exponents as long blocks (XXX: check this!) */
204  if (s->sample_rate_index <= 2)
205  g->long_end = 8;
206  else if (s->sample_rate_index != 8)
207  g->long_end = 6;
208  else
209  g->long_end = 4; /* 8000 Hz */
210 
211  g->short_start = 3;
212  } else {
213  g->long_end = 0;
214  g->short_start = 0;
215  }
216  } else {
217  g->short_start = 13;
218  g->long_end = 22;
219  }
220 }
221 
222 /* layer 1 unscaling */
223 /* n = number of bits of the mantissa minus 1 */
224 static inline int l1_unscale(int n, int mant, int scale_factor)
225 {
226  int shift, mod;
227  int64_t val;
228 
229  shift = scale_factor_modshift[scale_factor];
230  mod = shift & 3;
231  shift >>= 2;
232  val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
233  shift += n;
234  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
235  return (int)((val + (1LL << (shift - 1))) >> shift);
236 }
237 
238 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
239 {
240  int shift, mod, val;
241 
242  shift = scale_factor_modshift[scale_factor];
243  mod = shift & 3;
244  shift >>= 2;
245 
246  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
247  /* NOTE: at this point, 0 <= shift <= 21 */
248  if (shift > 0)
249  val = (val + (1 << (shift - 1))) >> shift;
250  return val;
251 }
252 
253 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
254 static inline int l3_unscale(int value, int exponent)
255 {
256  unsigned int m;
257  int e;
258 
259  e = table_4_3_exp [4 * value + (exponent & 3)];
260  m = table_4_3_value[4 * value + (exponent & 3)];
261  e -= exponent >> 2;
262  assert(e >= 1);
263  if (e > 31)
264  return 0;
265  m = (m + (1 << (e - 1))) >> e;
266 
267  return m;
268 }
269 
270 static av_cold void decode_init_static(void)
271 {
272  int i, j, k;
273  int offset;
274 
275  /* scale factors table for layer 1/2 */
276  for (i = 0; i < 64; i++) {
277  int shift, mod;
278  /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
279  shift = i / 3;
280  mod = i % 3;
281  scale_factor_modshift[i] = mod | (shift << 2);
282  }
283 
284  /* scale factor multiply for layer 1 */
285  for (i = 0; i < 15; i++) {
286  int n, norm;
287  n = i + 2;
288  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
289  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
290  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
291  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
292  av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
293  scale_factor_mult[i][0],
294  scale_factor_mult[i][1],
295  scale_factor_mult[i][2]);
296  }
297 
298  RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
299 
300  /* huffman decode tables */
301  offset = 0;
302  for (i = 1; i < 16; i++) {
303  const HuffTable *h = &mpa_huff_tables[i];
304  int xsize, x, y;
305  uint8_t tmp_bits [512];
306  uint16_t tmp_codes[512];
307 
308  memset(tmp_bits , 0, sizeof(tmp_bits ));
309  memset(tmp_codes, 0, sizeof(tmp_codes));
310 
311  xsize = h->xsize;
312 
313  j = 0;
314  for (x = 0; x < xsize; x++) {
315  for (y = 0; y < xsize; y++) {
316  tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
317  tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
318  }
319  }
320 
321  /* XXX: fail test */
322  huff_vlc[i].table = huff_vlc_tables+offset;
323  huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
324  init_vlc(&huff_vlc[i], 7, 512,
325  tmp_bits, 1, 1, tmp_codes, 2, 2,
327  offset += huff_vlc_tables_sizes[i];
328  }
329  assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
330 
331  offset = 0;
332  for (i = 0; i < 2; i++) {
333  huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
334  huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
335  init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
336  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
338  offset += huff_quad_vlc_tables_sizes[i];
339  }
340  assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
341 
342  for (i = 0; i < 9; i++) {
343  k = 0;
344  for (j = 0; j < 22; j++) {
345  band_index_long[i][j] = k;
346  k += band_size_long[i][j];
347  }
348  band_index_long[i][22] = k;
349  }
350 
351  /* compute n ^ (4/3) and store it in mantissa/exp format */
352 
354 
355  for (i = 0; i < 4; i++) {
356  if (ff_mpa_quant_bits[i] < 0) {
357  for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
358  int val1, val2, val3, steps;
359  int val = j;
360  steps = ff_mpa_quant_steps[i];
361  val1 = val % steps;
362  val /= steps;
363  val2 = val % steps;
364  val3 = val / steps;
365  division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
366  }
367  }
368  }
369 
370 
371  for (i = 0; i < 7; i++) {
372  float f;
373  INTFLOAT v;
374  if (i != 6) {
375  f = tan((double)i * M_PI / 12.0);
376  v = FIXR(f / (1.0 + f));
377  } else {
378  v = FIXR(1.0);
379  }
380  is_table[0][ i] = v;
381  is_table[1][6 - i] = v;
382  }
383  /* invalid values */
384  for (i = 7; i < 16; i++)
385  is_table[0][i] = is_table[1][i] = 0.0;
386 
387  for (i = 0; i < 16; i++) {
388  double f;
389  int e, k;
390 
391  for (j = 0; j < 2; j++) {
392  e = -(j + 1) * ((i + 1) >> 1);
393  f = pow(2.0, e / 4.0);
394  k = i & 1;
395  is_table_lsf[j][k ^ 1][i] = FIXR(f);
396  is_table_lsf[j][k ][i] = FIXR(1.0);
397  av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
398  i, j, (float) is_table_lsf[j][0][i],
399  (float) is_table_lsf[j][1][i]);
400  }
401  }
402 
403  for (i = 0; i < 8; i++) {
404  float ci, cs, ca;
405  ci = ci_table[i];
406  cs = 1.0 / sqrt(1.0 + ci * ci);
407  ca = cs * ci;
408 #if !CONFIG_FLOAT
409  csa_table[i][0] = FIXHR(cs/4);
410  csa_table[i][1] = FIXHR(ca/4);
411  csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
412  csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
413 #else
414  csa_table[i][0] = cs;
415  csa_table[i][1] = ca;
416  csa_table[i][2] = ca + cs;
417  csa_table[i][3] = ca - cs;
418 #endif
419  }
420 }
421 
422 static av_cold int decode_init(AVCodecContext * avctx)
423 {
424  static int initialized_tables = 0;
425  MPADecodeContext *s = avctx->priv_data;
426 
427  if (!initialized_tables) {
429  initialized_tables = 1;
430  }
431 
432  s->avctx = avctx;
433 
434  ff_mpadsp_init(&s->mpadsp);
435 
436  avctx->sample_fmt= OUT_FMT;
437  s->err_recognition = avctx->err_recognition;
438 
439  if (avctx->codec_id == CODEC_ID_MP3ADU)
440  s->adu_mode = 1;
441 
443  avctx->coded_frame = &s->frame;
444 
445  return 0;
446 }
447 
448 #define C3 FIXHR(0.86602540378443864676/2)
449 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
450 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
451 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
452 
453 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
454  cases. */
455 static void imdct12(INTFLOAT *out, INTFLOAT *in)
456 {
457  INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
458 
459  in0 = in[0*3];
460  in1 = in[1*3] + in[0*3];
461  in2 = in[2*3] + in[1*3];
462  in3 = in[3*3] + in[2*3];
463  in4 = in[4*3] + in[3*3];
464  in5 = in[5*3] + in[4*3];
465  in5 += in3;
466  in3 += in1;
467 
468  in2 = MULH3(in2, C3, 2);
469  in3 = MULH3(in3, C3, 4);
470 
471  t1 = in0 - in4;
472  t2 = MULH3(in1 - in5, C4, 2);
473 
474  out[ 7] =
475  out[10] = t1 + t2;
476  out[ 1] =
477  out[ 4] = t1 - t2;
478 
479  in0 += SHR(in4, 1);
480  in4 = in0 + in2;
481  in5 += 2*in1;
482  in1 = MULH3(in5 + in3, C5, 1);
483  out[ 8] =
484  out[ 9] = in4 + in1;
485  out[ 2] =
486  out[ 3] = in4 - in1;
487 
488  in0 -= in2;
489  in5 = MULH3(in5 - in3, C6, 2);
490  out[ 0] =
491  out[ 5] = in0 - in5;
492  out[ 6] =
493  out[11] = in0 + in5;
494 }
495 
496 /* return the number of decoded frames */
498 {
499  int bound, i, v, n, ch, j, mant;
500  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
501  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
502 
503  if (s->mode == MPA_JSTEREO)
504  bound = (s->mode_ext + 1) * 4;
505  else
506  bound = SBLIMIT;
507 
508  /* allocation bits */
509  for (i = 0; i < bound; i++) {
510  for (ch = 0; ch < s->nb_channels; ch++) {
511  allocation[ch][i] = get_bits(&s->gb, 4);
512  }
513  }
514  for (i = bound; i < SBLIMIT; i++)
515  allocation[0][i] = get_bits(&s->gb, 4);
516 
517  /* scale factors */
518  for (i = 0; i < bound; i++) {
519  for (ch = 0; ch < s->nb_channels; ch++) {
520  if (allocation[ch][i])
521  scale_factors[ch][i] = get_bits(&s->gb, 6);
522  }
523  }
524  for (i = bound; i < SBLIMIT; i++) {
525  if (allocation[0][i]) {
526  scale_factors[0][i] = get_bits(&s->gb, 6);
527  scale_factors[1][i] = get_bits(&s->gb, 6);
528  }
529  }
530 
531  /* compute samples */
532  for (j = 0; j < 12; j++) {
533  for (i = 0; i < bound; i++) {
534  for (ch = 0; ch < s->nb_channels; ch++) {
535  n = allocation[ch][i];
536  if (n) {
537  mant = get_bits(&s->gb, n + 1);
538  v = l1_unscale(n, mant, scale_factors[ch][i]);
539  } else {
540  v = 0;
541  }
542  s->sb_samples[ch][j][i] = v;
543  }
544  }
545  for (i = bound; i < SBLIMIT; i++) {
546  n = allocation[0][i];
547  if (n) {
548  mant = get_bits(&s->gb, n + 1);
549  v = l1_unscale(n, mant, scale_factors[0][i]);
550  s->sb_samples[0][j][i] = v;
551  v = l1_unscale(n, mant, scale_factors[1][i]);
552  s->sb_samples[1][j][i] = v;
553  } else {
554  s->sb_samples[0][j][i] = 0;
555  s->sb_samples[1][j][i] = 0;
556  }
557  }
558  }
559  return 12;
560 }
561 
563 {
564  int sblimit; /* number of used subbands */
565  const unsigned char *alloc_table;
566  int table, bit_alloc_bits, i, j, ch, bound, v;
567  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
568  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
569  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
570  int scale, qindex, bits, steps, k, l, m, b;
571 
572  /* select decoding table */
573  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
574  s->sample_rate, s->lsf);
575  sblimit = ff_mpa_sblimit_table[table];
576  alloc_table = ff_mpa_alloc_tables[table];
577 
578  if (s->mode == MPA_JSTEREO)
579  bound = (s->mode_ext + 1) * 4;
580  else
581  bound = sblimit;
582 
583  av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
584 
585  /* sanity check */
586  if (bound > sblimit)
587  bound = sblimit;
588 
589  /* parse bit allocation */
590  j = 0;
591  for (i = 0; i < bound; i++) {
592  bit_alloc_bits = alloc_table[j];
593  for (ch = 0; ch < s->nb_channels; ch++)
594  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
595  j += 1 << bit_alloc_bits;
596  }
597  for (i = bound; i < sblimit; i++) {
598  bit_alloc_bits = alloc_table[j];
599  v = get_bits(&s->gb, bit_alloc_bits);
600  bit_alloc[0][i] = v;
601  bit_alloc[1][i] = v;
602  j += 1 << bit_alloc_bits;
603  }
604 
605  /* scale codes */
606  for (i = 0; i < sblimit; i++) {
607  for (ch = 0; ch < s->nb_channels; ch++) {
608  if (bit_alloc[ch][i])
609  scale_code[ch][i] = get_bits(&s->gb, 2);
610  }
611  }
612 
613  /* scale factors */
614  for (i = 0; i < sblimit; i++) {
615  for (ch = 0; ch < s->nb_channels; ch++) {
616  if (bit_alloc[ch][i]) {
617  sf = scale_factors[ch][i];
618  switch (scale_code[ch][i]) {
619  default:
620  case 0:
621  sf[0] = get_bits(&s->gb, 6);
622  sf[1] = get_bits(&s->gb, 6);
623  sf[2] = get_bits(&s->gb, 6);
624  break;
625  case 2:
626  sf[0] = get_bits(&s->gb, 6);
627  sf[1] = sf[0];
628  sf[2] = sf[0];
629  break;
630  case 1:
631  sf[0] = get_bits(&s->gb, 6);
632  sf[2] = get_bits(&s->gb, 6);
633  sf[1] = sf[0];
634  break;
635  case 3:
636  sf[0] = get_bits(&s->gb, 6);
637  sf[2] = get_bits(&s->gb, 6);
638  sf[1] = sf[2];
639  break;
640  }
641  }
642  }
643  }
644 
645  /* samples */
646  for (k = 0; k < 3; k++) {
647  for (l = 0; l < 12; l += 3) {
648  j = 0;
649  for (i = 0; i < bound; i++) {
650  bit_alloc_bits = alloc_table[j];
651  for (ch = 0; ch < s->nb_channels; ch++) {
652  b = bit_alloc[ch][i];
653  if (b) {
654  scale = scale_factors[ch][i][k];
655  qindex = alloc_table[j+b];
656  bits = ff_mpa_quant_bits[qindex];
657  if (bits < 0) {
658  int v2;
659  /* 3 values at the same time */
660  v = get_bits(&s->gb, -bits);
661  v2 = division_tabs[qindex][v];
662  steps = ff_mpa_quant_steps[qindex];
663 
664  s->sb_samples[ch][k * 12 + l + 0][i] =
665  l2_unscale_group(steps, v2 & 15, scale);
666  s->sb_samples[ch][k * 12 + l + 1][i] =
667  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
668  s->sb_samples[ch][k * 12 + l + 2][i] =
669  l2_unscale_group(steps, v2 >> 8 , scale);
670  } else {
671  for (m = 0; m < 3; m++) {
672  v = get_bits(&s->gb, bits);
673  v = l1_unscale(bits - 1, v, scale);
674  s->sb_samples[ch][k * 12 + l + m][i] = v;
675  }
676  }
677  } else {
678  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
679  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
680  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
681  }
682  }
683  /* next subband in alloc table */
684  j += 1 << bit_alloc_bits;
685  }
686  /* XXX: find a way to avoid this duplication of code */
687  for (i = bound; i < sblimit; i++) {
688  bit_alloc_bits = alloc_table[j];
689  b = bit_alloc[0][i];
690  if (b) {
691  int mant, scale0, scale1;
692  scale0 = scale_factors[0][i][k];
693  scale1 = scale_factors[1][i][k];
694  qindex = alloc_table[j+b];
695  bits = ff_mpa_quant_bits[qindex];
696  if (bits < 0) {
697  /* 3 values at the same time */
698  v = get_bits(&s->gb, -bits);
699  steps = ff_mpa_quant_steps[qindex];
700  mant = v % steps;
701  v = v / steps;
702  s->sb_samples[0][k * 12 + l + 0][i] =
703  l2_unscale_group(steps, mant, scale0);
704  s->sb_samples[1][k * 12 + l + 0][i] =
705  l2_unscale_group(steps, mant, scale1);
706  mant = v % steps;
707  v = v / steps;
708  s->sb_samples[0][k * 12 + l + 1][i] =
709  l2_unscale_group(steps, mant, scale0);
710  s->sb_samples[1][k * 12 + l + 1][i] =
711  l2_unscale_group(steps, mant, scale1);
712  s->sb_samples[0][k * 12 + l + 2][i] =
713  l2_unscale_group(steps, v, scale0);
714  s->sb_samples[1][k * 12 + l + 2][i] =
715  l2_unscale_group(steps, v, scale1);
716  } else {
717  for (m = 0; m < 3; m++) {
718  mant = get_bits(&s->gb, bits);
719  s->sb_samples[0][k * 12 + l + m][i] =
720  l1_unscale(bits - 1, mant, scale0);
721  s->sb_samples[1][k * 12 + l + m][i] =
722  l1_unscale(bits - 1, mant, scale1);
723  }
724  }
725  } else {
726  s->sb_samples[0][k * 12 + l + 0][i] = 0;
727  s->sb_samples[0][k * 12 + l + 1][i] = 0;
728  s->sb_samples[0][k * 12 + l + 2][i] = 0;
729  s->sb_samples[1][k * 12 + l + 0][i] = 0;
730  s->sb_samples[1][k * 12 + l + 1][i] = 0;
731  s->sb_samples[1][k * 12 + l + 2][i] = 0;
732  }
733  /* next subband in alloc table */
734  j += 1 << bit_alloc_bits;
735  }
736  /* fill remaining samples to zero */
737  for (i = sblimit; i < SBLIMIT; i++) {
738  for (ch = 0; ch < s->nb_channels; ch++) {
739  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
740  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
741  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
742  }
743  }
744  }
745  }
746  return 3 * 12;
747 }
748 
749 #define SPLIT(dst,sf,n) \
750  if (n == 3) { \
751  int m = (sf * 171) >> 9; \
752  dst = sf - 3 * m; \
753  sf = m; \
754  } else if (n == 4) { \
755  dst = sf & 3; \
756  sf >>= 2; \
757  } else if (n == 5) { \
758  int m = (sf * 205) >> 10; \
759  dst = sf - 5 * m; \
760  sf = m; \
761  } else if (n == 6) { \
762  int m = (sf * 171) >> 10; \
763  dst = sf - 6 * m; \
764  sf = m; \
765  } else { \
766  dst = 0; \
767  }
768 
769 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
770  int n3)
771 {
772  SPLIT(slen[3], sf, n3)
773  SPLIT(slen[2], sf, n2)
774  SPLIT(slen[1], sf, n1)
775  slen[0] = sf;
776 }
777 
779  int16_t *exponents)
780 {
781  const uint8_t *bstab, *pretab;
782  int len, i, j, k, l, v0, shift, gain, gains[3];
783  int16_t *exp_ptr;
784 
785  exp_ptr = exponents;
786  gain = g->global_gain - 210;
787  shift = g->scalefac_scale + 1;
788 
789  bstab = band_size_long[s->sample_rate_index];
790  pretab = mpa_pretab[g->preflag];
791  for (i = 0; i < g->long_end; i++) {
792  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
793  len = bstab[i];
794  for (j = len; j > 0; j--)
795  *exp_ptr++ = v0;
796  }
797 
798  if (g->short_start < 13) {
799  bstab = band_size_short[s->sample_rate_index];
800  gains[0] = gain - (g->subblock_gain[0] << 3);
801  gains[1] = gain - (g->subblock_gain[1] << 3);
802  gains[2] = gain - (g->subblock_gain[2] << 3);
803  k = g->long_end;
804  for (i = g->short_start; i < 13; i++) {
805  len = bstab[i];
806  for (l = 0; l < 3; l++) {
807  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
808  for (j = len; j > 0; j--)
809  *exp_ptr++ = v0;
810  }
811  }
812  }
813 }
814 
815 /* handle n = 0 too */
816 static inline int get_bitsz(GetBitContext *s, int n)
817 {
818  return n ? get_bits(s, n) : 0;
819 }
820 
821 
822 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
823  int *end_pos2)
824 {
825  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
826  s->gb = s->in_gb;
827  s->in_gb.buffer = NULL;
828  assert((get_bits_count(&s->gb) & 7) == 0);
829  skip_bits_long(&s->gb, *pos - *end_pos);
830  *end_pos2 =
831  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
832  *pos = get_bits_count(&s->gb);
833  }
834 }
835 
836 /* Following is a optimized code for
837  INTFLOAT v = *src
838  if(get_bits1(&s->gb))
839  v = -v;
840  *dst = v;
841 */
842 #if CONFIG_FLOAT
843 #define READ_FLIP_SIGN(dst,src) \
844  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
845  AV_WN32A(dst, v);
846 #else
847 #define READ_FLIP_SIGN(dst,src) \
848  v = -get_bits1(&s->gb); \
849  *(dst) = (*(src) ^ v) - v;
850 #endif
851 
853  int16_t *exponents, int end_pos2)
854 {
855  int s_index;
856  int i;
857  int last_pos, bits_left;
858  VLC *vlc;
859  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
860 
861  /* low frequencies (called big values) */
862  s_index = 0;
863  for (i = 0; i < 3; i++) {
864  int j, k, l, linbits;
865  j = g->region_size[i];
866  if (j == 0)
867  continue;
868  /* select vlc table */
869  k = g->table_select[i];
870  l = mpa_huff_data[k][0];
871  linbits = mpa_huff_data[k][1];
872  vlc = &huff_vlc[l];
873 
874  if (!l) {
875  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
876  s_index += 2 * j;
877  continue;
878  }
879 
880  /* read huffcode and compute each couple */
881  for (; j > 0; j--) {
882  int exponent, x, y;
883  int v;
884  int pos = get_bits_count(&s->gb);
885 
886  if (pos >= end_pos){
887 // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
888  switch_buffer(s, &pos, &end_pos, &end_pos2);
889 // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
890  if (pos >= end_pos)
891  break;
892  }
893  y = get_vlc2(&s->gb, vlc->table, 7, 3);
894 
895  if (!y) {
896  g->sb_hybrid[s_index ] =
897  g->sb_hybrid[s_index+1] = 0;
898  s_index += 2;
899  continue;
900  }
901 
902  exponent= exponents[s_index];
903 
904  av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
905  i, g->region_size[i] - j, x, y, exponent);
906  if (y & 16) {
907  x = y >> 5;
908  y = y & 0x0f;
909  if (x < 15) {
910  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
911  } else {
912  x += get_bitsz(&s->gb, linbits);
913  v = l3_unscale(x, exponent);
914  if (get_bits1(&s->gb))
915  v = -v;
916  g->sb_hybrid[s_index] = v;
917  }
918  if (y < 15) {
919  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
920  } else {
921  y += get_bitsz(&s->gb, linbits);
922  v = l3_unscale(y, exponent);
923  if (get_bits1(&s->gb))
924  v = -v;
925  g->sb_hybrid[s_index+1] = v;
926  }
927  } else {
928  x = y >> 5;
929  y = y & 0x0f;
930  x += y;
931  if (x < 15) {
932  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
933  } else {
934  x += get_bitsz(&s->gb, linbits);
935  v = l3_unscale(x, exponent);
936  if (get_bits1(&s->gb))
937  v = -v;
938  g->sb_hybrid[s_index+!!y] = v;
939  }
940  g->sb_hybrid[s_index + !y] = 0;
941  }
942  s_index += 2;
943  }
944  }
945 
946  /* high frequencies */
947  vlc = &huff_quad_vlc[g->count1table_select];
948  last_pos = 0;
949  while (s_index <= 572) {
950  int pos, code;
951  pos = get_bits_count(&s->gb);
952  if (pos >= end_pos) {
953  if (pos > end_pos2 && last_pos) {
954  /* some encoders generate an incorrect size for this
955  part. We must go back into the data */
956  s_index -= 4;
957  skip_bits_long(&s->gb, last_pos - pos);
958  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
960  s_index=0;
961  break;
962  }
963 // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
964  switch_buffer(s, &pos, &end_pos, &end_pos2);
965 // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
966  if (pos >= end_pos)
967  break;
968  }
969  last_pos = pos;
970 
971  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
972  av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
973  g->sb_hybrid[s_index+0] =
974  g->sb_hybrid[s_index+1] =
975  g->sb_hybrid[s_index+2] =
976  g->sb_hybrid[s_index+3] = 0;
977  while (code) {
978  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
979  int v;
980  int pos = s_index + idxtab[code];
981  code ^= 8 >> idxtab[code];
982  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
983  }
984  s_index += 4;
985  }
986  /* skip extension bits */
987  bits_left = end_pos2 - get_bits_count(&s->gb);
988 //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
989  if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {
990  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
991  s_index=0;
992  } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
993  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
994  s_index = 0;
995  }
996  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
997  skip_bits_long(&s->gb, bits_left);
998 
999  i = get_bits_count(&s->gb);
1000  switch_buffer(s, &i, &end_pos, &end_pos2);
1001 
1002  return 0;
1003 }
1004 
1005 /* Reorder short blocks from bitstream order to interleaved order. It
1006  would be faster to do it in parsing, but the code would be far more
1007  complicated */
1009 {
1010  int i, j, len;
1011  INTFLOAT *ptr, *dst, *ptr1;
1012  INTFLOAT tmp[576];
1013 
1014  if (g->block_type != 2)
1015  return;
1016 
1017  if (g->switch_point) {
1018  if (s->sample_rate_index != 8)
1019  ptr = g->sb_hybrid + 36;
1020  else
1021  ptr = g->sb_hybrid + 48;
1022  } else {
1023  ptr = g->sb_hybrid;
1024  }
1025 
1026  for (i = g->short_start; i < 13; i++) {
1027  len = band_size_short[s->sample_rate_index][i];
1028  ptr1 = ptr;
1029  dst = tmp;
1030  for (j = len; j > 0; j--) {
1031  *dst++ = ptr[0*len];
1032  *dst++ = ptr[1*len];
1033  *dst++ = ptr[2*len];
1034  ptr++;
1035  }
1036  ptr += 2 * len;
1037  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1038  }
1039 }
1040 
1041 #define ISQRT2 FIXR(0.70710678118654752440)
1042 
1044 {
1045  int i, j, k, l;
1046  int sf_max, sf, len, non_zero_found;
1047  INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1048  int non_zero_found_short[3];
1049 
1050  /* intensity stereo */
1051  if (s->mode_ext & MODE_EXT_I_STEREO) {
1052  if (!s->lsf) {
1053  is_tab = is_table;
1054  sf_max = 7;
1055  } else {
1056  is_tab = is_table_lsf[g1->scalefac_compress & 1];
1057  sf_max = 16;
1058  }
1059 
1060  tab0 = g0->sb_hybrid + 576;
1061  tab1 = g1->sb_hybrid + 576;
1062 
1063  non_zero_found_short[0] = 0;
1064  non_zero_found_short[1] = 0;
1065  non_zero_found_short[2] = 0;
1066  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1067  for (i = 12; i >= g1->short_start; i--) {
1068  /* for last band, use previous scale factor */
1069  if (i != 11)
1070  k -= 3;
1071  len = band_size_short[s->sample_rate_index][i];
1072  for (l = 2; l >= 0; l--) {
1073  tab0 -= len;
1074  tab1 -= len;
1075  if (!non_zero_found_short[l]) {
1076  /* test if non zero band. if so, stop doing i-stereo */
1077  for (j = 0; j < len; j++) {
1078  if (tab1[j] != 0) {
1079  non_zero_found_short[l] = 1;
1080  goto found1;
1081  }
1082  }
1083  sf = g1->scale_factors[k + l];
1084  if (sf >= sf_max)
1085  goto found1;
1086 
1087  v1 = is_tab[0][sf];
1088  v2 = is_tab[1][sf];
1089  for (j = 0; j < len; j++) {
1090  tmp0 = tab0[j];
1091  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1092  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1093  }
1094  } else {
1095 found1:
1096  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1097  /* lower part of the spectrum : do ms stereo
1098  if enabled */
1099  for (j = 0; j < len; j++) {
1100  tmp0 = tab0[j];
1101  tmp1 = tab1[j];
1102  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1103  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1104  }
1105  }
1106  }
1107  }
1108  }
1109 
1110  non_zero_found = non_zero_found_short[0] |
1111  non_zero_found_short[1] |
1112  non_zero_found_short[2];
1113 
1114  for (i = g1->long_end - 1;i >= 0;i--) {
1115  len = band_size_long[s->sample_rate_index][i];
1116  tab0 -= len;
1117  tab1 -= len;
1118  /* test if non zero band. if so, stop doing i-stereo */
1119  if (!non_zero_found) {
1120  for (j = 0; j < len; j++) {
1121  if (tab1[j] != 0) {
1122  non_zero_found = 1;
1123  goto found2;
1124  }
1125  }
1126  /* for last band, use previous scale factor */
1127  k = (i == 21) ? 20 : i;
1128  sf = g1->scale_factors[k];
1129  if (sf >= sf_max)
1130  goto found2;
1131  v1 = is_tab[0][sf];
1132  v2 = is_tab[1][sf];
1133  for (j = 0; j < len; j++) {
1134  tmp0 = tab0[j];
1135  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1136  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1137  }
1138  } else {
1139 found2:
1140  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1141  /* lower part of the spectrum : do ms stereo
1142  if enabled */
1143  for (j = 0; j < len; j++) {
1144  tmp0 = tab0[j];
1145  tmp1 = tab1[j];
1146  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1147  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1148  }
1149  }
1150  }
1151  }
1152  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1153  /* ms stereo ONLY */
1154  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1155  global gain */
1156  tab0 = g0->sb_hybrid;
1157  tab1 = g1->sb_hybrid;
1158  for (i = 0; i < 576; i++) {
1159  tmp0 = tab0[i];
1160  tmp1 = tab1[i];
1161  tab0[i] = tmp0 + tmp1;
1162  tab1[i] = tmp0 - tmp1;
1163  }
1164  }
1165 }
1166 
1167 #if CONFIG_FLOAT
1168 #define AA(j) do { \
1169  float tmp0 = ptr[-1-j]; \
1170  float tmp1 = ptr[ j]; \
1171  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1172  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1173  } while (0)
1174 #else
1175 #define AA(j) do { \
1176  int tmp0 = ptr[-1-j]; \
1177  int tmp1 = ptr[ j]; \
1178  int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1179  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1180  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1181  } while (0)
1182 #endif
1183 
1185 {
1186  INTFLOAT *ptr;
1187  int n, i;
1188 
1189  /* we antialias only "long" bands */
1190  if (g->block_type == 2) {
1191  if (!g->switch_point)
1192  return;
1193  /* XXX: check this for 8000Hz case */
1194  n = 1;
1195  } else {
1196  n = SBLIMIT - 1;
1197  }
1198 
1199  ptr = g->sb_hybrid + 18;
1200  for (i = n; i > 0; i--) {
1201  AA(0);
1202  AA(1);
1203  AA(2);
1204  AA(3);
1205  AA(4);
1206  AA(5);
1207  AA(6);
1208  AA(7);
1209 
1210  ptr += 18;
1211  }
1212 }
1213 
1215  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1216 {
1217  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1218  INTFLOAT out2[12];
1219  int i, j, mdct_long_end, sblimit;
1220 
1221  /* find last non zero block */
1222  ptr = g->sb_hybrid + 576;
1223  ptr1 = g->sb_hybrid + 2 * 18;
1224  while (ptr >= ptr1) {
1225  int32_t *p;
1226  ptr -= 6;
1227  p = (int32_t*)ptr;
1228  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1229  break;
1230  }
1231  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1232 
1233  if (g->block_type == 2) {
1234  /* XXX: check for 8000 Hz */
1235  if (g->switch_point)
1236  mdct_long_end = 2;
1237  else
1238  mdct_long_end = 0;
1239  } else {
1240  mdct_long_end = sblimit;
1241  }
1242 
1243  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1244  mdct_long_end, g->switch_point,
1245  g->block_type);
1246 
1247  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1248  ptr = g->sb_hybrid + 18 * mdct_long_end;
1249 
1250  for (j = mdct_long_end; j < sblimit; j++) {
1251  /* select frequency inversion */
1252  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1253  out_ptr = sb_samples + j;
1254 
1255  for (i = 0; i < 6; i++) {
1256  *out_ptr = buf[4*i];
1257  out_ptr += SBLIMIT;
1258  }
1259  imdct12(out2, ptr + 0);
1260  for (i = 0; i < 6; i++) {
1261  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1262  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1263  out_ptr += SBLIMIT;
1264  }
1265  imdct12(out2, ptr + 1);
1266  for (i = 0; i < 6; i++) {
1267  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1268  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1269  out_ptr += SBLIMIT;
1270  }
1271  imdct12(out2, ptr + 2);
1272  for (i = 0; i < 6; i++) {
1273  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1274  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1275  buf[4*(i + 6*2)] = 0;
1276  }
1277  ptr += 18;
1278  buf += (j&3) != 3 ? 1 : (4*18-3);
1279  }
1280  /* zero bands */
1281  for (j = sblimit; j < SBLIMIT; j++) {
1282  /* overlap */
1283  out_ptr = sb_samples + j;
1284  for (i = 0; i < 18; i++) {
1285  *out_ptr = buf[4*i];
1286  buf[4*i] = 0;
1287  out_ptr += SBLIMIT;
1288  }
1289  buf += (j&3) != 3 ? 1 : (4*18-3);
1290  }
1291 }
1292 
1293 /* main layer3 decoding function */
1295 {
1296  int nb_granules, main_data_begin;
1297  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1298  GranuleDef *g;
1299  int16_t exponents[576]; //FIXME try INTFLOAT
1300 
1301  /* read side info */
1302  if (s->lsf) {
1303  main_data_begin = get_bits(&s->gb, 8);
1304  skip_bits(&s->gb, s->nb_channels);
1305  nb_granules = 1;
1306  } else {
1307  main_data_begin = get_bits(&s->gb, 9);
1308  if (s->nb_channels == 2)
1309  skip_bits(&s->gb, 3);
1310  else
1311  skip_bits(&s->gb, 5);
1312  nb_granules = 2;
1313  for (ch = 0; ch < s->nb_channels; ch++) {
1314  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1315  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1316  }
1317  }
1318 
1319  for (gr = 0; gr < nb_granules; gr++) {
1320  for (ch = 0; ch < s->nb_channels; ch++) {
1321  av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1322  g = &s->granules[ch][gr];
1323  g->part2_3_length = get_bits(&s->gb, 12);
1324  g->big_values = get_bits(&s->gb, 9);
1325  if (g->big_values > 288) {
1326  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1327  return AVERROR_INVALIDDATA;
1328  }
1329 
1330  g->global_gain = get_bits(&s->gb, 8);
1331  /* if MS stereo only is selected, we precompute the
1332  1/sqrt(2) renormalization factor */
1333  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1335  g->global_gain -= 2;
1336  if (s->lsf)
1337  g->scalefac_compress = get_bits(&s->gb, 9);
1338  else
1339  g->scalefac_compress = get_bits(&s->gb, 4);
1340  blocksplit_flag = get_bits1(&s->gb);
1341  if (blocksplit_flag) {
1342  g->block_type = get_bits(&s->gb, 2);
1343  if (g->block_type == 0) {
1344  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1345  return AVERROR_INVALIDDATA;
1346  }
1347  g->switch_point = get_bits1(&s->gb);
1348  for (i = 0; i < 2; i++)
1349  g->table_select[i] = get_bits(&s->gb, 5);
1350  for (i = 0; i < 3; i++)
1351  g->subblock_gain[i] = get_bits(&s->gb, 3);
1352  ff_init_short_region(s, g);
1353  } else {
1354  int region_address1, region_address2;
1355  g->block_type = 0;
1356  g->switch_point = 0;
1357  for (i = 0; i < 3; i++)
1358  g->table_select[i] = get_bits(&s->gb, 5);
1359  /* compute huffman coded region sizes */
1360  region_address1 = get_bits(&s->gb, 4);
1361  region_address2 = get_bits(&s->gb, 3);
1362  av_dlog(s->avctx, "region1=%d region2=%d\n",
1363  region_address1, region_address2);
1364  ff_init_long_region(s, g, region_address1, region_address2);
1365  }
1368 
1369  g->preflag = 0;
1370  if (!s->lsf)
1371  g->preflag = get_bits1(&s->gb);
1372  g->scalefac_scale = get_bits1(&s->gb);
1373  g->count1table_select = get_bits1(&s->gb);
1374  av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1375  g->block_type, g->switch_point);
1376  }
1377  }
1378 
1379  if (!s->adu_mode) {
1380  int skip;
1381  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1382  int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0,
1383  FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
1384  assert((get_bits_count(&s->gb) & 7) == 0);
1385  /* now we get bits from the main_data_begin offset */
1386  av_dlog(s->avctx, "seekback: %d\n", main_data_begin);
1387  //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
1388 
1389  memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
1390  s->in_gb = s->gb;
1391  init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1392 #if !UNCHECKED_BITSTREAM_READER
1393  s->gb.size_in_bits_plus8 += extrasize * 8;
1394 #endif
1395  s->last_buf_size <<= 3;
1396  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1397  for (ch = 0; ch < s->nb_channels; ch++) {
1398  g = &s->granules[ch][gr];
1399  s->last_buf_size += g->part2_3_length;
1400  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1401  }
1402  }
1403  skip = s->last_buf_size - 8 * main_data_begin;
1404  if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
1405  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
1406  s->gb = s->in_gb;
1407  s->in_gb.buffer = NULL;
1408  } else {
1409  skip_bits_long(&s->gb, skip);
1410  }
1411  } else {
1412  gr = 0;
1413  }
1414 
1415  for (; gr < nb_granules; gr++) {
1416  for (ch = 0; ch < s->nb_channels; ch++) {
1417  g = &s->granules[ch][gr];
1418  bits_pos = get_bits_count(&s->gb);
1419 
1420  if (!s->lsf) {
1421  uint8_t *sc;
1422  int slen, slen1, slen2;
1423 
1424  /* MPEG1 scale factors */
1425  slen1 = slen_table[0][g->scalefac_compress];
1426  slen2 = slen_table[1][g->scalefac_compress];
1427  av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1428  if (g->block_type == 2) {
1429  n = g->switch_point ? 17 : 18;
1430  j = 0;
1431  if (slen1) {
1432  for (i = 0; i < n; i++)
1433  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1434  } else {
1435  for (i = 0; i < n; i++)
1436  g->scale_factors[j++] = 0;
1437  }
1438  if (slen2) {
1439  for (i = 0; i < 18; i++)
1440  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1441  for (i = 0; i < 3; i++)
1442  g->scale_factors[j++] = 0;
1443  } else {
1444  for (i = 0; i < 21; i++)
1445  g->scale_factors[j++] = 0;
1446  }
1447  } else {
1448  sc = s->granules[ch][0].scale_factors;
1449  j = 0;
1450  for (k = 0; k < 4; k++) {
1451  n = k == 0 ? 6 : 5;
1452  if ((g->scfsi & (0x8 >> k)) == 0) {
1453  slen = (k < 2) ? slen1 : slen2;
1454  if (slen) {
1455  for (i = 0; i < n; i++)
1456  g->scale_factors[j++] = get_bits(&s->gb, slen);
1457  } else {
1458  for (i = 0; i < n; i++)
1459  g->scale_factors[j++] = 0;
1460  }
1461  } else {
1462  /* simply copy from last granule */
1463  for (i = 0; i < n; i++) {
1464  g->scale_factors[j] = sc[j];
1465  j++;
1466  }
1467  }
1468  }
1469  g->scale_factors[j++] = 0;
1470  }
1471  } else {
1472  int tindex, tindex2, slen[4], sl, sf;
1473 
1474  /* LSF scale factors */
1475  if (g->block_type == 2)
1476  tindex = g->switch_point ? 2 : 1;
1477  else
1478  tindex = 0;
1479 
1480  sf = g->scalefac_compress;
1481  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1482  /* intensity stereo case */
1483  sf >>= 1;
1484  if (sf < 180) {
1485  lsf_sf_expand(slen, sf, 6, 6, 0);
1486  tindex2 = 3;
1487  } else if (sf < 244) {
1488  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1489  tindex2 = 4;
1490  } else {
1491  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1492  tindex2 = 5;
1493  }
1494  } else {
1495  /* normal case */
1496  if (sf < 400) {
1497  lsf_sf_expand(slen, sf, 5, 4, 4);
1498  tindex2 = 0;
1499  } else if (sf < 500) {
1500  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1501  tindex2 = 1;
1502  } else {
1503  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1504  tindex2 = 2;
1505  g->preflag = 1;
1506  }
1507  }
1508 
1509  j = 0;
1510  for (k = 0; k < 4; k++) {
1511  n = lsf_nsf_table[tindex2][tindex][k];
1512  sl = slen[k];
1513  if (sl) {
1514  for (i = 0; i < n; i++)
1515  g->scale_factors[j++] = get_bits(&s->gb, sl);
1516  } else {
1517  for (i = 0; i < n; i++)
1518  g->scale_factors[j++] = 0;
1519  }
1520  }
1521  /* XXX: should compute exact size */
1522  for (; j < 40; j++)
1523  g->scale_factors[j] = 0;
1524  }
1525 
1526  exponents_from_scale_factors(s, g, exponents);
1527 
1528  /* read Huffman coded residue */
1529  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1530  } /* ch */
1531 
1532  if (s->nb_channels == 2)
1533  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1534 
1535  for (ch = 0; ch < s->nb_channels; ch++) {
1536  g = &s->granules[ch][gr];
1537 
1538  reorder_block(s, g);
1539  compute_antialias(s, g);
1540  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1541  }
1542  } /* gr */
1543  if (get_bits_count(&s->gb) < 0)
1544  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1545  return nb_granules * 18;
1546 }
1547 
1549  const uint8_t *buf, int buf_size)
1550 {
1551  int i, nb_frames, ch, ret;
1552  OUT_INT *samples_ptr;
1553 
1554  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1555 
1556  /* skip error protection field */
1557  if (s->error_protection)
1558  skip_bits(&s->gb, 16);
1559 
1560  switch(s->layer) {
1561  case 1:
1562  s->avctx->frame_size = 384;
1563  nb_frames = mp_decode_layer1(s);
1564  break;
1565  case 2:
1566  s->avctx->frame_size = 1152;
1567  nb_frames = mp_decode_layer2(s);
1568  break;
1569  case 3:
1570  s->avctx->frame_size = s->lsf ? 576 : 1152;
1571  default:
1572  nb_frames = mp_decode_layer3(s);
1573 
1574  if (nb_frames < 0)
1575  return nb_frames;
1576 
1577  s->last_buf_size=0;
1578  if (s->in_gb.buffer) {
1579  align_get_bits(&s->gb);
1580  i = get_bits_left(&s->gb)>>3;
1581  if (i >= 0 && i <= BACKSTEP_SIZE) {
1582  memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1583  s->last_buf_size=i;
1584  } else
1585  av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1586  s->gb = s->in_gb;
1587  s->in_gb.buffer = NULL;
1588  }
1589 
1590  align_get_bits(&s->gb);
1591  assert((get_bits_count(&s->gb) & 7) == 0);
1592  i = get_bits_left(&s->gb) >> 3;
1593 
1594  if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1595  if (i < 0)
1596  av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1597  i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1598  }
1599  assert(i <= buf_size - HEADER_SIZE && i >= 0);
1600  memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1601  s->last_buf_size += i;
1602  }
1603 
1604  /* get output buffer */
1605  if (!samples) {
1606  s->frame.nb_samples = s->avctx->frame_size;
1607  if ((ret = s->avctx->get_buffer(s->avctx, &s->frame)) < 0) {
1608  av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1609  return ret;
1610  }
1611  samples = (OUT_INT *)s->frame.data[0];
1612  }
1613 
1614  /* apply the synthesis filter */
1615  for (ch = 0; ch < s->nb_channels; ch++) {
1616  samples_ptr = samples + ch;
1617  for (i = 0; i < nb_frames; i++) {
1618  RENAME(ff_mpa_synth_filter)(
1619  &s->mpadsp,
1620  s->synth_buf[ch], &(s->synth_buf_offset[ch]),
1621  RENAME(ff_mpa_synth_window), &s->dither_state,
1622  samples_ptr, s->nb_channels,
1623  s->sb_samples[ch][i]);
1624  samples_ptr += 32 * s->nb_channels;
1625  }
1626  }
1627 
1628  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1629 }
1630 
1631 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1632  AVPacket *avpkt)
1633 {
1634  const uint8_t *buf = avpkt->data;
1635  int buf_size = avpkt->size;
1636  MPADecodeContext *s = avctx->priv_data;
1637  uint32_t header;
1638  int ret;
1639 
1640  if (buf_size < HEADER_SIZE)
1641  return AVERROR_INVALIDDATA;
1642 
1643  header = AV_RB32(buf);
1644  if (ff_mpa_check_header(header) < 0) {
1645  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1646  return AVERROR_INVALIDDATA;
1647  }
1648 
1649  if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1650  /* free format: prepare to compute frame size */
1651  s->frame_size = -1;
1652  return AVERROR_INVALIDDATA;
1653  }
1654  /* update codec info */
1655  avctx->channels = s->nb_channels;
1656  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1657  if (!avctx->bit_rate)
1658  avctx->bit_rate = s->bit_rate;
1659  avctx->sub_id = s->layer;
1660 
1661  if (s->frame_size <= 0 || s->frame_size > buf_size) {
1662  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1663  return AVERROR_INVALIDDATA;
1664  } else if (s->frame_size < buf_size) {
1665  av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
1666  buf_size= s->frame_size;
1667  }
1668 
1669  ret = mp_decode_frame(s, NULL, buf, buf_size);
1670  if (ret >= 0) {
1671  *got_frame_ptr = 1;
1672  *(AVFrame *)data = s->frame;
1673  avctx->sample_rate = s->sample_rate;
1674  //FIXME maybe move the other codec info stuff from above here too
1675  } else {
1676  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1677  /* Only return an error if the bad frame makes up the whole packet or
1678  * the error is related to buffer management.
1679  * If there is more data in the packet, just consume the bad frame
1680  * instead of returning an error, which would discard the whole
1681  * packet. */
1682  *got_frame_ptr = 0;
1683  if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1684  return ret;
1685  }
1686  s->frame_size = 0;
1687  return buf_size;
1688 }
1689 
1690 static void flush(AVCodecContext *avctx)
1691 {
1692  MPADecodeContext *s = avctx->priv_data;
1693  memset(s->synth_buf, 0, sizeof(s->synth_buf));
1694  s->last_buf_size = 0;
1695 }
1696 
1697 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1698 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1699  int *got_frame_ptr, AVPacket *avpkt)
1700 {
1701  const uint8_t *buf = avpkt->data;
1702  int buf_size = avpkt->size;
1703  MPADecodeContext *s = avctx->priv_data;
1704  uint32_t header;
1705  int len, out_size, ret = 0;
1706 
1707  len = buf_size;
1708 
1709  // Discard too short frames
1710  if (buf_size < HEADER_SIZE) {
1711  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1712  return AVERROR_INVALIDDATA;
1713  }
1714 
1715 
1716  if (len > MPA_MAX_CODED_FRAME_SIZE)
1718 
1719  // Get header and restore sync word
1720  header = AV_RB32(buf) | 0xffe00000;
1721 
1722  if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
1723  av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1724  return AVERROR_INVALIDDATA;
1725  }
1726 
1728  /* update codec info */
1729  avctx->sample_rate = s->sample_rate;
1730  avctx->channels = s->nb_channels;
1731  if (!avctx->bit_rate)
1732  avctx->bit_rate = s->bit_rate;
1733  avctx->sub_id = s->layer;
1734 
1735  s->frame_size = len;
1736 
1737 #if FF_API_PARSE_FRAME
1738  if (avctx->parse_only)
1739  out_size = buf_size;
1740  else
1741 #endif
1742  ret = mp_decode_frame(s, NULL, buf, buf_size);
1743  if (ret < 0) {
1744  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1745  return ret;
1746  }
1747 
1748  *got_frame_ptr = 1;
1749  *(AVFrame *)data = s->frame;
1750 
1751  return buf_size;
1752 }
1753 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1754 
1755 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1756 
1760 typedef struct MP3On4DecodeContext {
1761  AVFrame *frame;
1762  int frames;
1763  int syncword;
1764  const uint8_t *coff;
1765  MPADecodeContext *mp3decctx[5];
1766  OUT_INT *decoded_buf;
1767 } MP3On4DecodeContext;
1768 
1769 #include "mpeg4audio.h"
1770 
1771 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1772 
1773 /* number of mp3 decoder instances */
1774 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1775 
1776 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1777 static const uint8_t chan_offset[8][5] = {
1778  { 0 },
1779  { 0 }, // C
1780  { 0 }, // FLR
1781  { 2, 0 }, // C FLR
1782  { 2, 0, 3 }, // C FLR BS
1783  { 2, 0, 3 }, // C FLR BLRS
1784  { 2, 0, 4, 3 }, // C FLR BLRS LFE
1785  { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1786 };
1787 
1788 /* mp3on4 channel layouts */
1789 static const int16_t chan_layout[8] = {
1790  0,
1798 };
1799 
1800 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1801 {
1802  MP3On4DecodeContext *s = avctx->priv_data;
1803  int i;
1804 
1805  for (i = 0; i < s->frames; i++)
1806  av_free(s->mp3decctx[i]);
1807 
1808  av_freep(&s->decoded_buf);
1809 
1810  return 0;
1811 }
1812 
1813 
1814 static int decode_init_mp3on4(AVCodecContext * avctx)
1815 {
1816  MP3On4DecodeContext *s = avctx->priv_data;
1817  MPEG4AudioConfig cfg;
1818  int i;
1819 
1820  if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
1821  av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1822  return AVERROR_INVALIDDATA;
1823  }
1824 
1826  avctx->extradata_size * 8, 1);
1827  if (!cfg.chan_config || cfg.chan_config > 7) {
1828  av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1829  return AVERROR_INVALIDDATA;
1830  }
1831  s->frames = mp3Frames[cfg.chan_config];
1832  s->coff = chan_offset[cfg.chan_config];
1834  avctx->channel_layout = chan_layout[cfg.chan_config];
1835 
1836  if (cfg.sample_rate < 16000)
1837  s->syncword = 0xffe00000;
1838  else
1839  s->syncword = 0xfff00000;
1840 
1841  /* Init the first mp3 decoder in standard way, so that all tables get builded
1842  * We replace avctx->priv_data with the context of the first decoder so that
1843  * decode_init() does not have to be changed.
1844  * Other decoders will be initialized here copying data from the first context
1845  */
1846  // Allocate zeroed memory for the first decoder context
1847  s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1848  if (!s->mp3decctx[0])
1849  goto alloc_fail;
1850  // Put decoder context in place to make init_decode() happy
1851  avctx->priv_data = s->mp3decctx[0];
1852  decode_init(avctx);
1853  s->frame = avctx->coded_frame;
1854  // Restore mp3on4 context pointer
1855  avctx->priv_data = s;
1856  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1857 
1858  /* Create a separate codec/context for each frame (first is already ok).
1859  * Each frame is 1 or 2 channels - up to 5 frames allowed
1860  */
1861  for (i = 1; i < s->frames; i++) {
1862  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1863  if (!s->mp3decctx[i])
1864  goto alloc_fail;
1865  s->mp3decctx[i]->adu_mode = 1;
1866  s->mp3decctx[i]->avctx = avctx;
1867  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1868  }
1869 
1870  /* Allocate buffer for multi-channel output if needed */
1871  if (s->frames > 1) {
1872  s->decoded_buf = av_malloc(MPA_FRAME_SIZE * MPA_MAX_CHANNELS *
1873  sizeof(*s->decoded_buf));
1874  if (!s->decoded_buf)
1875  goto alloc_fail;
1876  }
1877 
1878  return 0;
1879 alloc_fail:
1880  decode_close_mp3on4(avctx);
1881  return AVERROR(ENOMEM);
1882 }
1883 
1884 
1885 static void flush_mp3on4(AVCodecContext *avctx)
1886 {
1887  int i;
1888  MP3On4DecodeContext *s = avctx->priv_data;
1889 
1890  for (i = 0; i < s->frames; i++) {
1891  MPADecodeContext *m = s->mp3decctx[i];
1892  memset(m->synth_buf, 0, sizeof(m->synth_buf));
1893  m->last_buf_size = 0;
1894  }
1895 }
1896 
1897 
1898 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1899  int *got_frame_ptr, AVPacket *avpkt)
1900 {
1901  const uint8_t *buf = avpkt->data;
1902  int buf_size = avpkt->size;
1903  MP3On4DecodeContext *s = avctx->priv_data;
1904  MPADecodeContext *m;
1905  int fsize, len = buf_size, out_size = 0;
1906  uint32_t header;
1907  OUT_INT *out_samples;
1908  OUT_INT *outptr, *bp;
1909  int fr, j, n, ch, ret;
1910 
1911  /* get output buffer */
1912  s->frame->nb_samples = MPA_FRAME_SIZE;
1913  if ((ret = avctx->get_buffer(avctx, s->frame)) < 0) {
1914  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1915  return ret;
1916  }
1917  out_samples = (OUT_INT *)s->frame->data[0];
1918 
1919  // Discard too short frames
1920  if (buf_size < HEADER_SIZE)
1921  return AVERROR_INVALIDDATA;
1922 
1923  // If only one decoder interleave is not needed
1924  outptr = s->frames == 1 ? out_samples : s->decoded_buf;
1925 
1926  avctx->bit_rate = 0;
1927 
1928  ch = 0;
1929  for (fr = 0; fr < s->frames; fr++) {
1930  fsize = AV_RB16(buf) >> 4;
1931  fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1932  m = s->mp3decctx[fr];
1933  assert(m != NULL);
1934 
1935  if (fsize < HEADER_SIZE) {
1936  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1937  return AVERROR_INVALIDDATA;
1938  }
1939  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1940 
1941  if (ff_mpa_check_header(header) < 0) // Bad header, discard block
1942  break;
1943 
1945 
1946  if (ch + m->nb_channels > avctx->channels) {
1947  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1948  "channel count\n");
1949  return AVERROR_INVALIDDATA;
1950  }
1951  ch += m->nb_channels;
1952 
1953  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
1954  return ret;
1955 
1956  out_size += ret;
1957  buf += fsize;
1958  len -= fsize;
1959 
1960  if (s->frames > 1) {
1961  n = m->avctx->frame_size*m->nb_channels;
1962  /* interleave output data */
1963  bp = out_samples + s->coff[fr];
1964  if (m->nb_channels == 1) {
1965  for (j = 0; j < n; j++) {
1966  *bp = s->decoded_buf[j];
1967  bp += avctx->channels;
1968  }
1969  } else {
1970  for (j = 0; j < n; j++) {
1971  bp[0] = s->decoded_buf[j++];
1972  bp[1] = s->decoded_buf[j];
1973  bp += avctx->channels;
1974  }
1975  }
1976  }
1977  avctx->bit_rate += m->bit_rate;
1978  }
1979 
1980  /* update codec info */
1981  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1982 
1983  s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1984  *got_frame_ptr = 1;
1985  *(AVFrame *)data = *s->frame;
1986 
1987  return buf_size;
1988 }
1989 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
1990 
1991 #if !CONFIG_FLOAT
1992 #if CONFIG_MP1_DECODER
1993 AVCodec ff_mp1_decoder = {
1994  .name = "mp1",
1995  .type = AVMEDIA_TYPE_AUDIO,
1996  .id = CODEC_ID_MP1,
1997  .priv_data_size = sizeof(MPADecodeContext),
1998  .init = decode_init,
1999  .decode = decode_frame,
2001  .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
2002 #else
2003  .capabilities = CODEC_CAP_DR1,
2004 #endif
2005  .flush = flush,
2006  .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
2007 };
2008 #endif
2009 #if CONFIG_MP2_DECODER
2010 AVCodec ff_mp2_decoder = {
2011  .name = "mp2",
2012  .type = AVMEDIA_TYPE_AUDIO,
2013  .id = CODEC_ID_MP2,
2014  .priv_data_size = sizeof(MPADecodeContext),
2015  .init = decode_init,
2016  .decode = decode_frame,
2018  .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
2019 #else
2020  .capabilities = CODEC_CAP_DR1,
2021 #endif
2022  .flush = flush,
2023  .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2024 };
2025 #endif
2026 #if CONFIG_MP3_DECODER
2027 AVCodec ff_mp3_decoder = {
2028  .name = "mp3",
2029  .type = AVMEDIA_TYPE_AUDIO,
2030  .id = CODEC_ID_MP3,
2031  .priv_data_size = sizeof(MPADecodeContext),
2032  .init = decode_init,
2033  .decode = decode_frame,
2035  .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
2036 #else
2037  .capabilities = CODEC_CAP_DR1,
2038 #endif
2039  .flush = flush,
2040  .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2041 };
2042 #endif
2043 #if CONFIG_MP3ADU_DECODER
2044 AVCodec ff_mp3adu_decoder = {
2045  .name = "mp3adu",
2046  .type = AVMEDIA_TYPE_AUDIO,
2047  .id = CODEC_ID_MP3ADU,
2048  .priv_data_size = sizeof(MPADecodeContext),
2049  .init = decode_init,
2050  .decode = decode_frame_adu,
2052  .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
2053 #else
2054  .capabilities = CODEC_CAP_DR1,
2055 #endif
2056  .flush = flush,
2057  .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2058 };
2059 #endif
2060 #if CONFIG_MP3ON4_DECODER
2061 AVCodec ff_mp3on4_decoder = {
2062  .name = "mp3on4",
2063  .type = AVMEDIA_TYPE_AUDIO,
2064  .id = CODEC_ID_MP3ON4,
2065  .priv_data_size = sizeof(MP3On4DecodeContext),
2066  .init = decode_init_mp3on4,
2067  .close = decode_close_mp3on4,
2068  .decode = decode_frame_mp3on4,
2069  .capabilities = CODEC_CAP_DR1,
2070  .flush = flush_mp3on4,
2071  .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
2072 };
2073 #endif
2074 #endif