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 
28 #include "libavutil/float_dsp.h"
29 #include "avcodec.h"
30 #include "get_bits.h"
31 #include "internal.h"
32 #include "mathops.h"
33 #include "mpegaudiodsp.h"
34 #include "dsputil.h"
35 
36 /*
37  * TODO:
38  * - test lsf / mpeg25 extensively.
39  */
40 
41 #include "mpegaudio.h"
42 #include "mpegaudiodecheader.h"
43 
44 #define BACKSTEP_SIZE 512
45 #define EXTRABYTES 24
46 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
47 
48 /* layer 3 "granule" */
49 typedef struct GranuleDef {
57  int table_select[3];
58  int subblock_gain[3];
61  int region_size[3]; /* number of huffman codes in each region */
62  int preflag;
63  int short_start, long_end; /* long/short band indexes */
65  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
66 } GranuleDef;
67 
68 typedef struct MPADecodeContext {
72  /* next header (used in free format parsing) */
79  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
80  GranuleDef granules[2][2]; /* Used in Layer 3 */
81  int adu_mode;
89 
90 #if CONFIG_FLOAT
91 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
92 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
93 # define FIXR(x) ((float)(x))
94 # define FIXHR(x) ((float)(x))
95 # define MULH3(x, y, s) ((s)*(y)*(x))
96 # define MULLx(x, y, s) ((y)*(x))
97 # define RENAME(a) a ## _float
98 # define OUT_FMT AV_SAMPLE_FMT_FLT
99 # define OUT_FMT_P AV_SAMPLE_FMT_FLTP
100 #else
101 # define SHR(a,b) ((a)>>(b))
102 /* WARNING: only correct for positive numbers */
103 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
104 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
105 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
106 # define MULH3(x, y, s) MULH((s)*(x), y)
107 # define MULLx(x, y, s) MULL(x,y,s)
108 # define RENAME(a) a ## _fixed
109 # define OUT_FMT AV_SAMPLE_FMT_S16
110 # define OUT_FMT_P AV_SAMPLE_FMT_S16P
111 #endif
112 
113 /****************/
114 
115 #define HEADER_SIZE 4
116 
117 #include "mpegaudiodata.h"
118 #include "mpegaudiodectab.h"
119 
120 /* vlc structure for decoding layer 3 huffman tables */
121 static VLC huff_vlc[16];
123  0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
124  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
125  ][2];
126 static const int huff_vlc_tables_sizes[16] = {
127  0, 128, 128, 128, 130, 128, 154, 166,
128  142, 204, 190, 170, 542, 460, 662, 414
129 };
130 static VLC huff_quad_vlc[2];
131 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
132 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
133 /* computed from band_size_long */
134 static uint16_t band_index_long[9][23];
135 #include "mpegaudio_tablegen.h"
136 /* intensity stereo coef table */
137 static INTFLOAT is_table[2][16];
138 static INTFLOAT is_table_lsf[2][2][16];
139 static INTFLOAT csa_table[8][4];
140 
141 static int16_t division_tab3[1<<6 ];
142 static int16_t division_tab5[1<<8 ];
143 static int16_t division_tab9[1<<11];
144 
145 static int16_t * const division_tabs[4] = {
147 };
148 
149 /* lower 2 bits: modulo 3, higher bits: shift */
150 static uint16_t scale_factor_modshift[64];
151 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
153 /* mult table for layer 2 group quantization */
154 
155 #define SCALE_GEN(v) \
156 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
157 
158 static const int32_t scale_factor_mult2[3][3] = {
159  SCALE_GEN(4.0 / 3.0), /* 3 steps */
160  SCALE_GEN(4.0 / 5.0), /* 5 steps */
161  SCALE_GEN(4.0 / 9.0), /* 9 steps */
162 };
163 
169 {
170  int i, k, j = 0;
171  g->region_size[2] = 576 / 2;
172  for (i = 0; i < 3; i++) {
173  k = FFMIN(g->region_size[i], g->big_values);
174  g->region_size[i] = k - j;
175  j = k;
176  }
177 }
178 
180 {
181  if (g->block_type == 2) {
182  if (s->sample_rate_index != 8)
183  g->region_size[0] = (36 / 2);
184  else
185  g->region_size[0] = (72 / 2);
186  } else {
187  if (s->sample_rate_index <= 2)
188  g->region_size[0] = (36 / 2);
189  else if (s->sample_rate_index != 8)
190  g->region_size[0] = (54 / 2);
191  else
192  g->region_size[0] = (108 / 2);
193  }
194  g->region_size[1] = (576 / 2);
195 }
196 
197 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
198 {
199  int l;
200  g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
201  /* should not overflow */
202  l = FFMIN(ra1 + ra2 + 2, 22);
203  g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
204 }
205 
207 {
208  if (g->block_type == 2) {
209  if (g->switch_point) {
210  /* if switched mode, we handle the 36 first samples as
211  long blocks. For 8000Hz, we handle the 72 first
212  exponents as long blocks */
213  if (s->sample_rate_index <= 2)
214  g->long_end = 8;
215  else
216  g->long_end = 6;
217 
218  g->short_start = 3;
219  } else {
220  g->long_end = 0;
221  g->short_start = 0;
222  }
223  } else {
224  g->short_start = 13;
225  g->long_end = 22;
226  }
227 }
228 
229 /* layer 1 unscaling */
230 /* n = number of bits of the mantissa minus 1 */
231 static inline int l1_unscale(int n, int mant, int scale_factor)
232 {
233  int shift, mod;
234  int64_t val;
235 
236  shift = scale_factor_modshift[scale_factor];
237  mod = shift & 3;
238  shift >>= 2;
239  val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
240  shift += n;
241  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
242  return (int)((val + (1LL << (shift - 1))) >> shift);
243 }
244 
245 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
246 {
247  int shift, mod, val;
248 
249  shift = scale_factor_modshift[scale_factor];
250  mod = shift & 3;
251  shift >>= 2;
252 
253  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
254  /* NOTE: at this point, 0 <= shift <= 21 */
255  if (shift > 0)
256  val = (val + (1 << (shift - 1))) >> shift;
257  return val;
258 }
259 
260 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
261 static inline int l3_unscale(int value, int exponent)
262 {
263  unsigned int m;
264  int e;
265 
266  e = table_4_3_exp [4 * value + (exponent & 3)];
267  m = table_4_3_value[4 * value + (exponent & 3)];
268  e -= exponent >> 2;
269  assert(e >= 1);
270  if (e > 31)
271  return 0;
272  m = (m + (1 << (e - 1))) >> e;
273 
274  return m;
275 }
276 
277 static av_cold void decode_init_static(void)
278 {
279  int i, j, k;
280  int offset;
281 
282  /* scale factors table for layer 1/2 */
283  for (i = 0; i < 64; i++) {
284  int shift, mod;
285  /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
286  shift = i / 3;
287  mod = i % 3;
288  scale_factor_modshift[i] = mod | (shift << 2);
289  }
290 
291  /* scale factor multiply for layer 1 */
292  for (i = 0; i < 15; i++) {
293  int n, norm;
294  n = i + 2;
295  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
296  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
297  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
298  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
299  av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
300  scale_factor_mult[i][0],
301  scale_factor_mult[i][1],
302  scale_factor_mult[i][2]);
303  }
304 
305  RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
306 
307  /* huffman decode tables */
308  offset = 0;
309  for (i = 1; i < 16; i++) {
310  const HuffTable *h = &mpa_huff_tables[i];
311  int xsize, x, y;
312  uint8_t tmp_bits [512] = { 0 };
313  uint16_t tmp_codes[512] = { 0 };
314 
315  xsize = h->xsize;
316 
317  j = 0;
318  for (x = 0; x < xsize; x++) {
319  for (y = 0; y < xsize; y++) {
320  tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
321  tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
322  }
323  }
324 
325  /* XXX: fail test */
326  huff_vlc[i].table = huff_vlc_tables+offset;
327  huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
328  init_vlc(&huff_vlc[i], 7, 512,
329  tmp_bits, 1, 1, tmp_codes, 2, 2,
331  offset += huff_vlc_tables_sizes[i];
332  }
333  assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
334 
335  offset = 0;
336  for (i = 0; i < 2; i++) {
337  huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
338  huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
339  init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
340  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
342  offset += huff_quad_vlc_tables_sizes[i];
343  }
344  assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
345 
346  for (i = 0; i < 9; i++) {
347  k = 0;
348  for (j = 0; j < 22; j++) {
349  band_index_long[i][j] = k;
350  k += band_size_long[i][j];
351  }
352  band_index_long[i][22] = k;
353  }
354 
355  /* compute n ^ (4/3) and store it in mantissa/exp format */
356 
358 
359  for (i = 0; i < 4; i++) {
360  if (ff_mpa_quant_bits[i] < 0) {
361  for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
362  int val1, val2, val3, steps;
363  int val = j;
364  steps = ff_mpa_quant_steps[i];
365  val1 = val % steps;
366  val /= steps;
367  val2 = val % steps;
368  val3 = val / steps;
369  division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
370  }
371  }
372  }
373 
374 
375  for (i = 0; i < 7; i++) {
376  float f;
377  INTFLOAT v;
378  if (i != 6) {
379  f = tan((double)i * M_PI / 12.0);
380  v = FIXR(f / (1.0 + f));
381  } else {
382  v = FIXR(1.0);
383  }
384  is_table[0][ i] = v;
385  is_table[1][6 - i] = v;
386  }
387  /* invalid values */
388  for (i = 7; i < 16; i++)
389  is_table[0][i] = is_table[1][i] = 0.0;
390 
391  for (i = 0; i < 16; i++) {
392  double f;
393  int e, k;
394 
395  for (j = 0; j < 2; j++) {
396  e = -(j + 1) * ((i + 1) >> 1);
397  f = pow(2.0, e / 4.0);
398  k = i & 1;
399  is_table_lsf[j][k ^ 1][i] = FIXR(f);
400  is_table_lsf[j][k ][i] = FIXR(1.0);
401  av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
402  i, j, (float) is_table_lsf[j][0][i],
403  (float) is_table_lsf[j][1][i]);
404  }
405  }
406 
407  for (i = 0; i < 8; i++) {
408  float ci, cs, ca;
409  ci = ci_table[i];
410  cs = 1.0 / sqrt(1.0 + ci * ci);
411  ca = cs * ci;
412 #if !CONFIG_FLOAT
413  csa_table[i][0] = FIXHR(cs/4);
414  csa_table[i][1] = FIXHR(ca/4);
415  csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
416  csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
417 #else
418  csa_table[i][0] = cs;
419  csa_table[i][1] = ca;
420  csa_table[i][2] = ca + cs;
421  csa_table[i][3] = ca - cs;
422 #endif
423  }
424 }
425 
426 static av_cold int decode_init(AVCodecContext * avctx)
427 {
428  static int initialized_tables = 0;
429  MPADecodeContext *s = avctx->priv_data;
430 
431  if (!initialized_tables) {
433  initialized_tables = 1;
434  }
435 
436  s->avctx = avctx;
437 
439  ff_mpadsp_init(&s->mpadsp);
440 
441  if (avctx->request_sample_fmt == OUT_FMT &&
442  avctx->codec_id != AV_CODEC_ID_MP3ON4)
443  avctx->sample_fmt = OUT_FMT;
444  else
445  avctx->sample_fmt = OUT_FMT_P;
446  s->err_recognition = avctx->err_recognition;
447 
448  if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
449  s->adu_mode = 1;
450 
452  avctx->coded_frame = &s->frame;
453 
454  return 0;
455 }
456 
457 #define C3 FIXHR(0.86602540378443864676/2)
458 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
459 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
460 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
461 
462 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
463  cases. */
464 static void imdct12(INTFLOAT *out, INTFLOAT *in)
465 {
466  INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
467 
468  in0 = in[0*3];
469  in1 = in[1*3] + in[0*3];
470  in2 = in[2*3] + in[1*3];
471  in3 = in[3*3] + in[2*3];
472  in4 = in[4*3] + in[3*3];
473  in5 = in[5*3] + in[4*3];
474  in5 += in3;
475  in3 += in1;
476 
477  in2 = MULH3(in2, C3, 2);
478  in3 = MULH3(in3, C3, 4);
479 
480  t1 = in0 - in4;
481  t2 = MULH3(in1 - in5, C4, 2);
482 
483  out[ 7] =
484  out[10] = t1 + t2;
485  out[ 1] =
486  out[ 4] = t1 - t2;
487 
488  in0 += SHR(in4, 1);
489  in4 = in0 + in2;
490  in5 += 2*in1;
491  in1 = MULH3(in5 + in3, C5, 1);
492  out[ 8] =
493  out[ 9] = in4 + in1;
494  out[ 2] =
495  out[ 3] = in4 - in1;
496 
497  in0 -= in2;
498  in5 = MULH3(in5 - in3, C6, 2);
499  out[ 0] =
500  out[ 5] = in0 - in5;
501  out[ 6] =
502  out[11] = in0 + in5;
503 }
504 
505 /* return the number of decoded frames */
507 {
508  int bound, i, v, n, ch, j, mant;
509  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
510  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
511 
512  if (s->mode == MPA_JSTEREO)
513  bound = (s->mode_ext + 1) * 4;
514  else
515  bound = SBLIMIT;
516 
517  /* allocation bits */
518  for (i = 0; i < bound; i++) {
519  for (ch = 0; ch < s->nb_channels; ch++) {
520  allocation[ch][i] = get_bits(&s->gb, 4);
521  }
522  }
523  for (i = bound; i < SBLIMIT; i++)
524  allocation[0][i] = get_bits(&s->gb, 4);
525 
526  /* scale factors */
527  for (i = 0; i < bound; i++) {
528  for (ch = 0; ch < s->nb_channels; ch++) {
529  if (allocation[ch][i])
530  scale_factors[ch][i] = get_bits(&s->gb, 6);
531  }
532  }
533  for (i = bound; i < SBLIMIT; i++) {
534  if (allocation[0][i]) {
535  scale_factors[0][i] = get_bits(&s->gb, 6);
536  scale_factors[1][i] = get_bits(&s->gb, 6);
537  }
538  }
539 
540  /* compute samples */
541  for (j = 0; j < 12; j++) {
542  for (i = 0; i < bound; i++) {
543  for (ch = 0; ch < s->nb_channels; ch++) {
544  n = allocation[ch][i];
545  if (n) {
546  mant = get_bits(&s->gb, n + 1);
547  v = l1_unscale(n, mant, scale_factors[ch][i]);
548  } else {
549  v = 0;
550  }
551  s->sb_samples[ch][j][i] = v;
552  }
553  }
554  for (i = bound; i < SBLIMIT; i++) {
555  n = allocation[0][i];
556  if (n) {
557  mant = get_bits(&s->gb, n + 1);
558  v = l1_unscale(n, mant, scale_factors[0][i]);
559  s->sb_samples[0][j][i] = v;
560  v = l1_unscale(n, mant, scale_factors[1][i]);
561  s->sb_samples[1][j][i] = v;
562  } else {
563  s->sb_samples[0][j][i] = 0;
564  s->sb_samples[1][j][i] = 0;
565  }
566  }
567  }
568  return 12;
569 }
570 
572 {
573  int sblimit; /* number of used subbands */
574  const unsigned char *alloc_table;
575  int table, bit_alloc_bits, i, j, ch, bound, v;
576  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
577  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
578  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
579  int scale, qindex, bits, steps, k, l, m, b;
580 
581  /* select decoding table */
582  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
583  s->sample_rate, s->lsf);
584  sblimit = ff_mpa_sblimit_table[table];
585  alloc_table = ff_mpa_alloc_tables[table];
586 
587  if (s->mode == MPA_JSTEREO)
588  bound = (s->mode_ext + 1) * 4;
589  else
590  bound = sblimit;
591 
592  av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
593 
594  /* sanity check */
595  if (bound > sblimit)
596  bound = sblimit;
597 
598  /* parse bit allocation */
599  j = 0;
600  for (i = 0; i < bound; i++) {
601  bit_alloc_bits = alloc_table[j];
602  for (ch = 0; ch < s->nb_channels; ch++)
603  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
604  j += 1 << bit_alloc_bits;
605  }
606  for (i = bound; i < sblimit; i++) {
607  bit_alloc_bits = alloc_table[j];
608  v = get_bits(&s->gb, bit_alloc_bits);
609  bit_alloc[0][i] = v;
610  bit_alloc[1][i] = v;
611  j += 1 << bit_alloc_bits;
612  }
613 
614  /* scale codes */
615  for (i = 0; i < sblimit; i++) {
616  for (ch = 0; ch < s->nb_channels; ch++) {
617  if (bit_alloc[ch][i])
618  scale_code[ch][i] = get_bits(&s->gb, 2);
619  }
620  }
621 
622  /* scale factors */
623  for (i = 0; i < sblimit; i++) {
624  for (ch = 0; ch < s->nb_channels; ch++) {
625  if (bit_alloc[ch][i]) {
626  sf = scale_factors[ch][i];
627  switch (scale_code[ch][i]) {
628  default:
629  case 0:
630  sf[0] = get_bits(&s->gb, 6);
631  sf[1] = get_bits(&s->gb, 6);
632  sf[2] = get_bits(&s->gb, 6);
633  break;
634  case 2:
635  sf[0] = get_bits(&s->gb, 6);
636  sf[1] = sf[0];
637  sf[2] = sf[0];
638  break;
639  case 1:
640  sf[0] = get_bits(&s->gb, 6);
641  sf[2] = get_bits(&s->gb, 6);
642  sf[1] = sf[0];
643  break;
644  case 3:
645  sf[0] = get_bits(&s->gb, 6);
646  sf[2] = get_bits(&s->gb, 6);
647  sf[1] = sf[2];
648  break;
649  }
650  }
651  }
652  }
653 
654  /* samples */
655  for (k = 0; k < 3; k++) {
656  for (l = 0; l < 12; l += 3) {
657  j = 0;
658  for (i = 0; i < bound; i++) {
659  bit_alloc_bits = alloc_table[j];
660  for (ch = 0; ch < s->nb_channels; ch++) {
661  b = bit_alloc[ch][i];
662  if (b) {
663  scale = scale_factors[ch][i][k];
664  qindex = alloc_table[j+b];
665  bits = ff_mpa_quant_bits[qindex];
666  if (bits < 0) {
667  int v2;
668  /* 3 values at the same time */
669  v = get_bits(&s->gb, -bits);
670  v2 = division_tabs[qindex][v];
671  steps = ff_mpa_quant_steps[qindex];
672 
673  s->sb_samples[ch][k * 12 + l + 0][i] =
674  l2_unscale_group(steps, v2 & 15, scale);
675  s->sb_samples[ch][k * 12 + l + 1][i] =
676  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
677  s->sb_samples[ch][k * 12 + l + 2][i] =
678  l2_unscale_group(steps, v2 >> 8 , scale);
679  } else {
680  for (m = 0; m < 3; m++) {
681  v = get_bits(&s->gb, bits);
682  v = l1_unscale(bits - 1, v, scale);
683  s->sb_samples[ch][k * 12 + l + m][i] = v;
684  }
685  }
686  } else {
687  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
688  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
689  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
690  }
691  }
692  /* next subband in alloc table */
693  j += 1 << bit_alloc_bits;
694  }
695  /* XXX: find a way to avoid this duplication of code */
696  for (i = bound; i < sblimit; i++) {
697  bit_alloc_bits = alloc_table[j];
698  b = bit_alloc[0][i];
699  if (b) {
700  int mant, scale0, scale1;
701  scale0 = scale_factors[0][i][k];
702  scale1 = scale_factors[1][i][k];
703  qindex = alloc_table[j+b];
704  bits = ff_mpa_quant_bits[qindex];
705  if (bits < 0) {
706  /* 3 values at the same time */
707  v = get_bits(&s->gb, -bits);
708  steps = ff_mpa_quant_steps[qindex];
709  mant = v % steps;
710  v = v / steps;
711  s->sb_samples[0][k * 12 + l + 0][i] =
712  l2_unscale_group(steps, mant, scale0);
713  s->sb_samples[1][k * 12 + l + 0][i] =
714  l2_unscale_group(steps, mant, scale1);
715  mant = v % steps;
716  v = v / steps;
717  s->sb_samples[0][k * 12 + l + 1][i] =
718  l2_unscale_group(steps, mant, scale0);
719  s->sb_samples[1][k * 12 + l + 1][i] =
720  l2_unscale_group(steps, mant, scale1);
721  s->sb_samples[0][k * 12 + l + 2][i] =
722  l2_unscale_group(steps, v, scale0);
723  s->sb_samples[1][k * 12 + l + 2][i] =
724  l2_unscale_group(steps, v, scale1);
725  } else {
726  for (m = 0; m < 3; m++) {
727  mant = get_bits(&s->gb, bits);
728  s->sb_samples[0][k * 12 + l + m][i] =
729  l1_unscale(bits - 1, mant, scale0);
730  s->sb_samples[1][k * 12 + l + m][i] =
731  l1_unscale(bits - 1, mant, scale1);
732  }
733  }
734  } else {
735  s->sb_samples[0][k * 12 + l + 0][i] = 0;
736  s->sb_samples[0][k * 12 + l + 1][i] = 0;
737  s->sb_samples[0][k * 12 + l + 2][i] = 0;
738  s->sb_samples[1][k * 12 + l + 0][i] = 0;
739  s->sb_samples[1][k * 12 + l + 1][i] = 0;
740  s->sb_samples[1][k * 12 + l + 2][i] = 0;
741  }
742  /* next subband in alloc table */
743  j += 1 << bit_alloc_bits;
744  }
745  /* fill remaining samples to zero */
746  for (i = sblimit; i < SBLIMIT; i++) {
747  for (ch = 0; ch < s->nb_channels; ch++) {
748  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
749  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
750  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
751  }
752  }
753  }
754  }
755  return 3 * 12;
756 }
757 
758 #define SPLIT(dst,sf,n) \
759  if (n == 3) { \
760  int m = (sf * 171) >> 9; \
761  dst = sf - 3 * m; \
762  sf = m; \
763  } else if (n == 4) { \
764  dst = sf & 3; \
765  sf >>= 2; \
766  } else if (n == 5) { \
767  int m = (sf * 205) >> 10; \
768  dst = sf - 5 * m; \
769  sf = m; \
770  } else if (n == 6) { \
771  int m = (sf * 171) >> 10; \
772  dst = sf - 6 * m; \
773  sf = m; \
774  } else { \
775  dst = 0; \
776  }
777 
778 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
779  int n3)
780 {
781  SPLIT(slen[3], sf, n3)
782  SPLIT(slen[2], sf, n2)
783  SPLIT(slen[1], sf, n1)
784  slen[0] = sf;
785 }
786 
788  int16_t *exponents)
789 {
790  const uint8_t *bstab, *pretab;
791  int len, i, j, k, l, v0, shift, gain, gains[3];
792  int16_t *exp_ptr;
793 
794  exp_ptr = exponents;
795  gain = g->global_gain - 210;
796  shift = g->scalefac_scale + 1;
797 
798  bstab = band_size_long[s->sample_rate_index];
799  pretab = mpa_pretab[g->preflag];
800  for (i = 0; i < g->long_end; i++) {
801  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
802  len = bstab[i];
803  for (j = len; j > 0; j--)
804  *exp_ptr++ = v0;
805  }
806 
807  if (g->short_start < 13) {
808  bstab = band_size_short[s->sample_rate_index];
809  gains[0] = gain - (g->subblock_gain[0] << 3);
810  gains[1] = gain - (g->subblock_gain[1] << 3);
811  gains[2] = gain - (g->subblock_gain[2] << 3);
812  k = g->long_end;
813  for (i = g->short_start; i < 13; i++) {
814  len = bstab[i];
815  for (l = 0; l < 3; l++) {
816  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
817  for (j = len; j > 0; j--)
818  *exp_ptr++ = v0;
819  }
820  }
821  }
822 }
823 
824 /* handle n = 0 too */
825 static inline int get_bitsz(GetBitContext *s, int n)
826 {
827  return n ? get_bits(s, n) : 0;
828 }
829 
830 
831 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
832  int *end_pos2)
833 {
834  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
835  s->gb = s->in_gb;
836  s->in_gb.buffer = NULL;
837  assert((get_bits_count(&s->gb) & 7) == 0);
838  skip_bits_long(&s->gb, *pos - *end_pos);
839  *end_pos2 =
840  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
841  *pos = get_bits_count(&s->gb);
842  }
843 }
844 
845 /* Following is a optimized code for
846  INTFLOAT v = *src
847  if(get_bits1(&s->gb))
848  v = -v;
849  *dst = v;
850 */
851 #if CONFIG_FLOAT
852 #define READ_FLIP_SIGN(dst,src) \
853  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
854  AV_WN32A(dst, v);
855 #else
856 #define READ_FLIP_SIGN(dst,src) \
857  v = -get_bits1(&s->gb); \
858  *(dst) = (*(src) ^ v) - v;
859 #endif
860 
862  int16_t *exponents, int end_pos2)
863 {
864  int s_index;
865  int i;
866  int last_pos, bits_left;
867  VLC *vlc;
868  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
869 
870  /* low frequencies (called big values) */
871  s_index = 0;
872  for (i = 0; i < 3; i++) {
873  int j, k, l, linbits;
874  j = g->region_size[i];
875  if (j == 0)
876  continue;
877  /* select vlc table */
878  k = g->table_select[i];
879  l = mpa_huff_data[k][0];
880  linbits = mpa_huff_data[k][1];
881  vlc = &huff_vlc[l];
882 
883  if (!l) {
884  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
885  s_index += 2 * j;
886  continue;
887  }
888 
889  /* read huffcode and compute each couple */
890  for (; j > 0; j--) {
891  int exponent, x, y;
892  int v;
893  int pos = get_bits_count(&s->gb);
894 
895  if (pos >= end_pos){
896  switch_buffer(s, &pos, &end_pos, &end_pos2);
897  if (pos >= end_pos)
898  break;
899  }
900  y = get_vlc2(&s->gb, vlc->table, 7, 3);
901 
902  if (!y) {
903  g->sb_hybrid[s_index ] =
904  g->sb_hybrid[s_index+1] = 0;
905  s_index += 2;
906  continue;
907  }
908 
909  exponent= exponents[s_index];
910 
911  av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
912  i, g->region_size[i] - j, x, y, exponent);
913  if (y & 16) {
914  x = y >> 5;
915  y = y & 0x0f;
916  if (x < 15) {
917  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
918  } else {
919  x += get_bitsz(&s->gb, linbits);
920  v = l3_unscale(x, exponent);
921  if (get_bits1(&s->gb))
922  v = -v;
923  g->sb_hybrid[s_index] = v;
924  }
925  if (y < 15) {
926  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
927  } else {
928  y += get_bitsz(&s->gb, linbits);
929  v = l3_unscale(y, exponent);
930  if (get_bits1(&s->gb))
931  v = -v;
932  g->sb_hybrid[s_index+1] = v;
933  }
934  } else {
935  x = y >> 5;
936  y = y & 0x0f;
937  x += y;
938  if (x < 15) {
939  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
940  } else {
941  x += get_bitsz(&s->gb, linbits);
942  v = l3_unscale(x, exponent);
943  if (get_bits1(&s->gb))
944  v = -v;
945  g->sb_hybrid[s_index+!!y] = v;
946  }
947  g->sb_hybrid[s_index + !y] = 0;
948  }
949  s_index += 2;
950  }
951  }
952 
953  /* high frequencies */
954  vlc = &huff_quad_vlc[g->count1table_select];
955  last_pos = 0;
956  while (s_index <= 572) {
957  int pos, code;
958  pos = get_bits_count(&s->gb);
959  if (pos >= end_pos) {
960  if (pos > end_pos2 && last_pos) {
961  /* some encoders generate an incorrect size for this
962  part. We must go back into the data */
963  s_index -= 4;
964  skip_bits_long(&s->gb, last_pos - pos);
965  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
967  s_index=0;
968  break;
969  }
970  switch_buffer(s, &pos, &end_pos, &end_pos2);
971  if (pos >= end_pos)
972  break;
973  }
974  last_pos = pos;
975 
976  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
977  av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
978  g->sb_hybrid[s_index+0] =
979  g->sb_hybrid[s_index+1] =
980  g->sb_hybrid[s_index+2] =
981  g->sb_hybrid[s_index+3] = 0;
982  while (code) {
983  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
984  int v;
985  int pos = s_index + idxtab[code];
986  code ^= 8 >> idxtab[code];
987  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
988  }
989  s_index += 4;
990  }
991  /* skip extension bits */
992  bits_left = end_pos2 - get_bits_count(&s->gb);
993  if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {
994  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
995  s_index=0;
996  } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
997  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
998  s_index = 0;
999  }
1000  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
1001  skip_bits_long(&s->gb, bits_left);
1002 
1003  i = get_bits_count(&s->gb);
1004  switch_buffer(s, &i, &end_pos, &end_pos2);
1005 
1006  return 0;
1007 }
1008 
1009 /* Reorder short blocks from bitstream order to interleaved order. It
1010  would be faster to do it in parsing, but the code would be far more
1011  complicated */
1013 {
1014  int i, j, len;
1015  INTFLOAT *ptr, *dst, *ptr1;
1016  INTFLOAT tmp[576];
1017 
1018  if (g->block_type != 2)
1019  return;
1020 
1021  if (g->switch_point) {
1022  if (s->sample_rate_index != 8)
1023  ptr = g->sb_hybrid + 36;
1024  else
1025  ptr = g->sb_hybrid + 72;
1026  } else {
1027  ptr = g->sb_hybrid;
1028  }
1029 
1030  for (i = g->short_start; i < 13; i++) {
1031  len = band_size_short[s->sample_rate_index][i];
1032  ptr1 = ptr;
1033  dst = tmp;
1034  for (j = len; j > 0; j--) {
1035  *dst++ = ptr[0*len];
1036  *dst++ = ptr[1*len];
1037  *dst++ = ptr[2*len];
1038  ptr++;
1039  }
1040  ptr += 2 * len;
1041  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1042  }
1043 }
1044 
1045 #define ISQRT2 FIXR(0.70710678118654752440)
1046 
1048 {
1049  int i, j, k, l;
1050  int sf_max, sf, len, non_zero_found;
1051  INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1052  int non_zero_found_short[3];
1053 
1054  /* intensity stereo */
1055  if (s->mode_ext & MODE_EXT_I_STEREO) {
1056  if (!s->lsf) {
1057  is_tab = is_table;
1058  sf_max = 7;
1059  } else {
1060  is_tab = is_table_lsf[g1->scalefac_compress & 1];
1061  sf_max = 16;
1062  }
1063 
1064  tab0 = g0->sb_hybrid + 576;
1065  tab1 = g1->sb_hybrid + 576;
1066 
1067  non_zero_found_short[0] = 0;
1068  non_zero_found_short[1] = 0;
1069  non_zero_found_short[2] = 0;
1070  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1071  for (i = 12; i >= g1->short_start; i--) {
1072  /* for last band, use previous scale factor */
1073  if (i != 11)
1074  k -= 3;
1075  len = band_size_short[s->sample_rate_index][i];
1076  for (l = 2; l >= 0; l--) {
1077  tab0 -= len;
1078  tab1 -= len;
1079  if (!non_zero_found_short[l]) {
1080  /* test if non zero band. if so, stop doing i-stereo */
1081  for (j = 0; j < len; j++) {
1082  if (tab1[j] != 0) {
1083  non_zero_found_short[l] = 1;
1084  goto found1;
1085  }
1086  }
1087  sf = g1->scale_factors[k + l];
1088  if (sf >= sf_max)
1089  goto found1;
1090 
1091  v1 = is_tab[0][sf];
1092  v2 = is_tab[1][sf];
1093  for (j = 0; j < len; j++) {
1094  tmp0 = tab0[j];
1095  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1096  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1097  }
1098  } else {
1099 found1:
1100  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1101  /* lower part of the spectrum : do ms stereo
1102  if enabled */
1103  for (j = 0; j < len; j++) {
1104  tmp0 = tab0[j];
1105  tmp1 = tab1[j];
1106  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1107  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1108  }
1109  }
1110  }
1111  }
1112  }
1113 
1114  non_zero_found = non_zero_found_short[0] |
1115  non_zero_found_short[1] |
1116  non_zero_found_short[2];
1117 
1118  for (i = g1->long_end - 1;i >= 0;i--) {
1119  len = band_size_long[s->sample_rate_index][i];
1120  tab0 -= len;
1121  tab1 -= len;
1122  /* test if non zero band. if so, stop doing i-stereo */
1123  if (!non_zero_found) {
1124  for (j = 0; j < len; j++) {
1125  if (tab1[j] != 0) {
1126  non_zero_found = 1;
1127  goto found2;
1128  }
1129  }
1130  /* for last band, use previous scale factor */
1131  k = (i == 21) ? 20 : i;
1132  sf = g1->scale_factors[k];
1133  if (sf >= sf_max)
1134  goto found2;
1135  v1 = is_tab[0][sf];
1136  v2 = is_tab[1][sf];
1137  for (j = 0; j < len; j++) {
1138  tmp0 = tab0[j];
1139  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1140  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1141  }
1142  } else {
1143 found2:
1144  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1145  /* lower part of the spectrum : do ms stereo
1146  if enabled */
1147  for (j = 0; j < len; j++) {
1148  tmp0 = tab0[j];
1149  tmp1 = tab1[j];
1150  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1151  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1152  }
1153  }
1154  }
1155  }
1156  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1157  /* ms stereo ONLY */
1158  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1159  global gain */
1160 #if CONFIG_FLOAT
1161  s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1162 #else
1163  tab0 = g0->sb_hybrid;
1164  tab1 = g1->sb_hybrid;
1165  for (i = 0; i < 576; i++) {
1166  tmp0 = tab0[i];
1167  tmp1 = tab1[i];
1168  tab0[i] = tmp0 + tmp1;
1169  tab1[i] = tmp0 - tmp1;
1170  }
1171 #endif
1172  }
1173 }
1174 
1175 #if CONFIG_FLOAT
1176 #define AA(j) do { \
1177  float tmp0 = ptr[-1-j]; \
1178  float tmp1 = ptr[ j]; \
1179  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1180  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1181  } while (0)
1182 #else
1183 #define AA(j) do { \
1184  int tmp0 = ptr[-1-j]; \
1185  int tmp1 = ptr[ j]; \
1186  int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1187  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1188  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1189  } while (0)
1190 #endif
1191 
1193 {
1194  INTFLOAT *ptr;
1195  int n, i;
1196 
1197  /* we antialias only "long" bands */
1198  if (g->block_type == 2) {
1199  if (!g->switch_point)
1200  return;
1201  /* XXX: check this for 8000Hz case */
1202  n = 1;
1203  } else {
1204  n = SBLIMIT - 1;
1205  }
1206 
1207  ptr = g->sb_hybrid + 18;
1208  for (i = n; i > 0; i--) {
1209  AA(0);
1210  AA(1);
1211  AA(2);
1212  AA(3);
1213  AA(4);
1214  AA(5);
1215  AA(6);
1216  AA(7);
1217 
1218  ptr += 18;
1219  }
1220 }
1221 
1223  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1224 {
1225  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1226  INTFLOAT out2[12];
1227  int i, j, mdct_long_end, sblimit;
1228 
1229  /* find last non zero block */
1230  ptr = g->sb_hybrid + 576;
1231  ptr1 = g->sb_hybrid + 2 * 18;
1232  while (ptr >= ptr1) {
1233  int32_t *p;
1234  ptr -= 6;
1235  p = (int32_t*)ptr;
1236  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1237  break;
1238  }
1239  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1240 
1241  if (g->block_type == 2) {
1242  /* XXX: check for 8000 Hz */
1243  if (g->switch_point)
1244  mdct_long_end = 2;
1245  else
1246  mdct_long_end = 0;
1247  } else {
1248  mdct_long_end = sblimit;
1249  }
1250 
1251  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1252  mdct_long_end, g->switch_point,
1253  g->block_type);
1254 
1255  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1256  ptr = g->sb_hybrid + 18 * mdct_long_end;
1257 
1258  for (j = mdct_long_end; j < sblimit; j++) {
1259  /* select frequency inversion */
1260  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1261  out_ptr = sb_samples + j;
1262 
1263  for (i = 0; i < 6; i++) {
1264  *out_ptr = buf[4*i];
1265  out_ptr += SBLIMIT;
1266  }
1267  imdct12(out2, ptr + 0);
1268  for (i = 0; i < 6; i++) {
1269  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1270  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1271  out_ptr += SBLIMIT;
1272  }
1273  imdct12(out2, ptr + 1);
1274  for (i = 0; i < 6; i++) {
1275  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1276  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1277  out_ptr += SBLIMIT;
1278  }
1279  imdct12(out2, ptr + 2);
1280  for (i = 0; i < 6; i++) {
1281  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1282  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1283  buf[4*(i + 6*2)] = 0;
1284  }
1285  ptr += 18;
1286  buf += (j&3) != 3 ? 1 : (4*18-3);
1287  }
1288  /* zero bands */
1289  for (j = sblimit; j < SBLIMIT; j++) {
1290  /* overlap */
1291  out_ptr = sb_samples + j;
1292  for (i = 0; i < 18; i++) {
1293  *out_ptr = buf[4*i];
1294  buf[4*i] = 0;
1295  out_ptr += SBLIMIT;
1296  }
1297  buf += (j&3) != 3 ? 1 : (4*18-3);
1298  }
1299 }
1300 
1301 /* main layer3 decoding function */
1303 {
1304  int nb_granules, main_data_begin;
1305  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1306  GranuleDef *g;
1307  int16_t exponents[576]; //FIXME try INTFLOAT
1308 
1309  /* read side info */
1310  if (s->lsf) {
1311  main_data_begin = get_bits(&s->gb, 8);
1312  skip_bits(&s->gb, s->nb_channels);
1313  nb_granules = 1;
1314  } else {
1315  main_data_begin = get_bits(&s->gb, 9);
1316  if (s->nb_channels == 2)
1317  skip_bits(&s->gb, 3);
1318  else
1319  skip_bits(&s->gb, 5);
1320  nb_granules = 2;
1321  for (ch = 0; ch < s->nb_channels; ch++) {
1322  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1323  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1324  }
1325  }
1326 
1327  for (gr = 0; gr < nb_granules; gr++) {
1328  for (ch = 0; ch < s->nb_channels; ch++) {
1329  av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1330  g = &s->granules[ch][gr];
1331  g->part2_3_length = get_bits(&s->gb, 12);
1332  g->big_values = get_bits(&s->gb, 9);
1333  if (g->big_values > 288) {
1334  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1335  return AVERROR_INVALIDDATA;
1336  }
1337 
1338  g->global_gain = get_bits(&s->gb, 8);
1339  /* if MS stereo only is selected, we precompute the
1340  1/sqrt(2) renormalization factor */
1341  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1343  g->global_gain -= 2;
1344  if (s->lsf)
1345  g->scalefac_compress = get_bits(&s->gb, 9);
1346  else
1347  g->scalefac_compress = get_bits(&s->gb, 4);
1348  blocksplit_flag = get_bits1(&s->gb);
1349  if (blocksplit_flag) {
1350  g->block_type = get_bits(&s->gb, 2);
1351  if (g->block_type == 0) {
1352  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1353  return AVERROR_INVALIDDATA;
1354  }
1355  g->switch_point = get_bits1(&s->gb);
1356  for (i = 0; i < 2; i++)
1357  g->table_select[i] = get_bits(&s->gb, 5);
1358  for (i = 0; i < 3; i++)
1359  g->subblock_gain[i] = get_bits(&s->gb, 3);
1360  ff_init_short_region(s, g);
1361  } else {
1362  int region_address1, region_address2;
1363  g->block_type = 0;
1364  g->switch_point = 0;
1365  for (i = 0; i < 3; i++)
1366  g->table_select[i] = get_bits(&s->gb, 5);
1367  /* compute huffman coded region sizes */
1368  region_address1 = get_bits(&s->gb, 4);
1369  region_address2 = get_bits(&s->gb, 3);
1370  av_dlog(s->avctx, "region1=%d region2=%d\n",
1371  region_address1, region_address2);
1372  ff_init_long_region(s, g, region_address1, region_address2);
1373  }
1376 
1377  g->preflag = 0;
1378  if (!s->lsf)
1379  g->preflag = get_bits1(&s->gb);
1380  g->scalefac_scale = get_bits1(&s->gb);
1381  g->count1table_select = get_bits1(&s->gb);
1382  av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1383  g->block_type, g->switch_point);
1384  }
1385  }
1386 
1387  if (!s->adu_mode) {
1388  int skip;
1389  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1390  int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0,
1391  FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
1392  assert((get_bits_count(&s->gb) & 7) == 0);
1393  /* now we get bits from the main_data_begin offset */
1394  av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1395  main_data_begin, s->last_buf_size);
1396 
1397  memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
1398  s->in_gb = s->gb;
1399  init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1400 #if !UNCHECKED_BITSTREAM_READER
1401  s->gb.size_in_bits_plus8 += extrasize * 8;
1402 #endif
1403  s->last_buf_size <<= 3;
1404  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1405  for (ch = 0; ch < s->nb_channels; ch++) {
1406  g = &s->granules[ch][gr];
1407  s->last_buf_size += g->part2_3_length;
1408  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1409  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1410  }
1411  }
1412  skip = s->last_buf_size - 8 * main_data_begin;
1413  if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
1414  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
1415  s->gb = s->in_gb;
1416  s->in_gb.buffer = NULL;
1417  } else {
1418  skip_bits_long(&s->gb, skip);
1419  }
1420  } else {
1421  gr = 0;
1422  }
1423 
1424  for (; gr < nb_granules; gr++) {
1425  for (ch = 0; ch < s->nb_channels; ch++) {
1426  g = &s->granules[ch][gr];
1427  bits_pos = get_bits_count(&s->gb);
1428 
1429  if (!s->lsf) {
1430  uint8_t *sc;
1431  int slen, slen1, slen2;
1432 
1433  /* MPEG1 scale factors */
1434  slen1 = slen_table[0][g->scalefac_compress];
1435  slen2 = slen_table[1][g->scalefac_compress];
1436  av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1437  if (g->block_type == 2) {
1438  n = g->switch_point ? 17 : 18;
1439  j = 0;
1440  if (slen1) {
1441  for (i = 0; i < n; i++)
1442  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1443  } else {
1444  for (i = 0; i < n; i++)
1445  g->scale_factors[j++] = 0;
1446  }
1447  if (slen2) {
1448  for (i = 0; i < 18; i++)
1449  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1450  for (i = 0; i < 3; i++)
1451  g->scale_factors[j++] = 0;
1452  } else {
1453  for (i = 0; i < 21; i++)
1454  g->scale_factors[j++] = 0;
1455  }
1456  } else {
1457  sc = s->granules[ch][0].scale_factors;
1458  j = 0;
1459  for (k = 0; k < 4; k++) {
1460  n = k == 0 ? 6 : 5;
1461  if ((g->scfsi & (0x8 >> k)) == 0) {
1462  slen = (k < 2) ? slen1 : slen2;
1463  if (slen) {
1464  for (i = 0; i < n; i++)
1465  g->scale_factors[j++] = get_bits(&s->gb, slen);
1466  } else {
1467  for (i = 0; i < n; i++)
1468  g->scale_factors[j++] = 0;
1469  }
1470  } else {
1471  /* simply copy from last granule */
1472  for (i = 0; i < n; i++) {
1473  g->scale_factors[j] = sc[j];
1474  j++;
1475  }
1476  }
1477  }
1478  g->scale_factors[j++] = 0;
1479  }
1480  } else {
1481  int tindex, tindex2, slen[4], sl, sf;
1482 
1483  /* LSF scale factors */
1484  if (g->block_type == 2)
1485  tindex = g->switch_point ? 2 : 1;
1486  else
1487  tindex = 0;
1488 
1489  sf = g->scalefac_compress;
1490  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1491  /* intensity stereo case */
1492  sf >>= 1;
1493  if (sf < 180) {
1494  lsf_sf_expand(slen, sf, 6, 6, 0);
1495  tindex2 = 3;
1496  } else if (sf < 244) {
1497  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1498  tindex2 = 4;
1499  } else {
1500  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1501  tindex2 = 5;
1502  }
1503  } else {
1504  /* normal case */
1505  if (sf < 400) {
1506  lsf_sf_expand(slen, sf, 5, 4, 4);
1507  tindex2 = 0;
1508  } else if (sf < 500) {
1509  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1510  tindex2 = 1;
1511  } else {
1512  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1513  tindex2 = 2;
1514  g->preflag = 1;
1515  }
1516  }
1517 
1518  j = 0;
1519  for (k = 0; k < 4; k++) {
1520  n = lsf_nsf_table[tindex2][tindex][k];
1521  sl = slen[k];
1522  if (sl) {
1523  for (i = 0; i < n; i++)
1524  g->scale_factors[j++] = get_bits(&s->gb, sl);
1525  } else {
1526  for (i = 0; i < n; i++)
1527  g->scale_factors[j++] = 0;
1528  }
1529  }
1530  /* XXX: should compute exact size */
1531  for (; j < 40; j++)
1532  g->scale_factors[j] = 0;
1533  }
1534 
1535  exponents_from_scale_factors(s, g, exponents);
1536 
1537  /* read Huffman coded residue */
1538  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1539  } /* ch */
1540 
1541  if (s->mode == MPA_JSTEREO)
1542  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1543 
1544  for (ch = 0; ch < s->nb_channels; ch++) {
1545  g = &s->granules[ch][gr];
1546 
1547  reorder_block(s, g);
1548  compute_antialias(s, g);
1549  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1550  }
1551  } /* gr */
1552  if (get_bits_count(&s->gb) < 0)
1553  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1554  return nb_granules * 18;
1555 }
1556 
1558  const uint8_t *buf, int buf_size)
1559 {
1560  int i, nb_frames, ch, ret;
1561  OUT_INT *samples_ptr;
1562 
1563  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1564 
1565  /* skip error protection field */
1566  if (s->error_protection)
1567  skip_bits(&s->gb, 16);
1568 
1569  switch(s->layer) {
1570  case 1:
1571  s->avctx->frame_size = 384;
1572  nb_frames = mp_decode_layer1(s);
1573  break;
1574  case 2:
1575  s->avctx->frame_size = 1152;
1576  nb_frames = mp_decode_layer2(s);
1577  break;
1578  case 3:
1579  s->avctx->frame_size = s->lsf ? 576 : 1152;
1580  default:
1581  nb_frames = mp_decode_layer3(s);
1582 
1583  if (nb_frames < 0)
1584  return nb_frames;
1585 
1586  s->last_buf_size=0;
1587  if (s->in_gb.buffer) {
1588  align_get_bits(&s->gb);
1589  i = get_bits_left(&s->gb)>>3;
1590  if (i >= 0 && i <= BACKSTEP_SIZE) {
1591  memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1592  s->last_buf_size=i;
1593  } else
1594  av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1595  s->gb = s->in_gb;
1596  s->in_gb.buffer = NULL;
1597  }
1598 
1599  align_get_bits(&s->gb);
1600  assert((get_bits_count(&s->gb) & 7) == 0);
1601  i = get_bits_left(&s->gb) >> 3;
1602 
1603  if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1604  if (i < 0)
1605  av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1606  i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1607  }
1608  assert(i <= buf_size - HEADER_SIZE && i >= 0);
1609  memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1610  s->last_buf_size += i;
1611  }
1612 
1613  /* get output buffer */
1614  if (!samples) {
1615  s->frame.nb_samples = s->avctx->frame_size;
1616  if ((ret = ff_get_buffer(s->avctx, &s->frame)) < 0) {
1617  av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1618  return ret;
1619  }
1620  samples = (OUT_INT **)s->frame.extended_data;
1621  }
1622 
1623  /* apply the synthesis filter */
1624  for (ch = 0; ch < s->nb_channels; ch++) {
1625  int sample_stride;
1626  if (s->avctx->sample_fmt == OUT_FMT_P) {
1627  samples_ptr = samples[ch];
1628  sample_stride = 1;
1629  } else {
1630  samples_ptr = samples[0] + ch;
1631  sample_stride = s->nb_channels;
1632  }
1633  for (i = 0; i < nb_frames; i++) {
1635  &(s->synth_buf_offset[ch]),
1636  RENAME(ff_mpa_synth_window),
1637  &s->dither_state, samples_ptr,
1638  sample_stride, s->sb_samples[ch][i]);
1639  samples_ptr += 32 * sample_stride;
1640  }
1641  }
1642 
1643  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1644 }
1645 
1646 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1647  AVPacket *avpkt)
1648 {
1649  const uint8_t *buf = avpkt->data;
1650  int buf_size = avpkt->size;
1651  MPADecodeContext *s = avctx->priv_data;
1652  uint32_t header;
1653  int ret;
1654 
1655  if (buf_size < HEADER_SIZE)
1656  return AVERROR_INVALIDDATA;
1657 
1658  header = AV_RB32(buf);
1659  if (ff_mpa_check_header(header) < 0) {
1660  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1661  return AVERROR_INVALIDDATA;
1662  }
1663 
1664  if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1665  /* free format: prepare to compute frame size */
1666  s->frame_size = -1;
1667  return AVERROR_INVALIDDATA;
1668  }
1669  /* update codec info */
1670  avctx->channels = s->nb_channels;
1671  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1672  if (!avctx->bit_rate)
1673  avctx->bit_rate = s->bit_rate;
1674 
1675  if (s->frame_size <= 0 || s->frame_size > buf_size) {
1676  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1677  return AVERROR_INVALIDDATA;
1678  } else if (s->frame_size < buf_size) {
1679  buf_size= s->frame_size;
1680  }
1681 
1682  ret = mp_decode_frame(s, NULL, buf, buf_size);
1683  if (ret >= 0) {
1684  *got_frame_ptr = 1;
1685  *(AVFrame *)data = s->frame;
1686  avctx->sample_rate = s->sample_rate;
1687  //FIXME maybe move the other codec info stuff from above here too
1688  } else {
1689  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1690  /* Only return an error if the bad frame makes up the whole packet or
1691  * the error is related to buffer management.
1692  * If there is more data in the packet, just consume the bad frame
1693  * instead of returning an error, which would discard the whole
1694  * packet. */
1695  *got_frame_ptr = 0;
1696  if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1697  return ret;
1698  }
1699  s->frame_size = 0;
1700  return buf_size;
1701 }
1702 
1703 static void mp_flush(MPADecodeContext *ctx)
1704 {
1705  memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1706  ctx->last_buf_size = 0;
1707 }
1708 
1709 static void flush(AVCodecContext *avctx)
1710 {
1711  mp_flush(avctx->priv_data);
1712 }
1713 
1714 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1715 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1716  int *got_frame_ptr, AVPacket *avpkt)
1717 {
1718  const uint8_t *buf = avpkt->data;
1719  int buf_size = avpkt->size;
1720  MPADecodeContext *s = avctx->priv_data;
1721  uint32_t header;
1722  int len, ret;
1723 
1724  len = buf_size;
1725 
1726  // Discard too short frames
1727  if (buf_size < HEADER_SIZE) {
1728  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1729  return AVERROR_INVALIDDATA;
1730  }
1731 
1732 
1733  if (len > MPA_MAX_CODED_FRAME_SIZE)
1735 
1736  // Get header and restore sync word
1737  header = AV_RB32(buf) | 0xffe00000;
1738 
1739  if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
1740  av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1741  return AVERROR_INVALIDDATA;
1742  }
1743 
1745  /* update codec info */
1746  avctx->sample_rate = s->sample_rate;
1747  avctx->channels = s->nb_channels;
1748  if (!avctx->bit_rate)
1749  avctx->bit_rate = s->bit_rate;
1750 
1751  s->frame_size = len;
1752 
1753  ret = mp_decode_frame(s, NULL, buf, buf_size);
1754  if (ret < 0) {
1755  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1756  return ret;
1757  }
1758 
1759  *got_frame_ptr = 1;
1760  *(AVFrame *)data = s->frame;
1761 
1762  return buf_size;
1763 }
1764 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1765 
1766 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1767 
1771 typedef struct MP3On4DecodeContext {
1772  AVFrame *frame;
1773  int frames;
1774  int syncword;
1775  const uint8_t *coff;
1776  MPADecodeContext *mp3decctx[5];
1777 } MP3On4DecodeContext;
1778 
1779 #include "mpeg4audio.h"
1780 
1781 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1782 
1783 /* number of mp3 decoder instances */
1784 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1785 
1786 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1787 static const uint8_t chan_offset[8][5] = {
1788  { 0 },
1789  { 0 }, // C
1790  { 0 }, // FLR
1791  { 2, 0 }, // C FLR
1792  { 2, 0, 3 }, // C FLR BS
1793  { 2, 0, 3 }, // C FLR BLRS
1794  { 2, 0, 4, 3 }, // C FLR BLRS LFE
1795  { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1796 };
1797 
1798 /* mp3on4 channel layouts */
1799 static const int16_t chan_layout[8] = {
1800  0,
1808 };
1809 
1810 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1811 {
1812  MP3On4DecodeContext *s = avctx->priv_data;
1813  int i;
1814 
1815  for (i = 0; i < s->frames; i++)
1816  av_free(s->mp3decctx[i]);
1817 
1818  return 0;
1819 }
1820 
1821 
1822 static int decode_init_mp3on4(AVCodecContext * avctx)
1823 {
1824  MP3On4DecodeContext *s = avctx->priv_data;
1825  MPEG4AudioConfig cfg;
1826  int i;
1827 
1828  if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
1829  av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1830  return AVERROR_INVALIDDATA;
1831  }
1832 
1834  avctx->extradata_size * 8, 1);
1835  if (!cfg.chan_config || cfg.chan_config > 7) {
1836  av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1837  return AVERROR_INVALIDDATA;
1838  }
1839  s->frames = mp3Frames[cfg.chan_config];
1840  s->coff = chan_offset[cfg.chan_config];
1842  avctx->channel_layout = chan_layout[cfg.chan_config];
1843 
1844  if (cfg.sample_rate < 16000)
1845  s->syncword = 0xffe00000;
1846  else
1847  s->syncword = 0xfff00000;
1848 
1849  /* Init the first mp3 decoder in standard way, so that all tables get builded
1850  * We replace avctx->priv_data with the context of the first decoder so that
1851  * decode_init() does not have to be changed.
1852  * Other decoders will be initialized here copying data from the first context
1853  */
1854  // Allocate zeroed memory for the first decoder context
1855  s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1856  if (!s->mp3decctx[0])
1857  goto alloc_fail;
1858  // Put decoder context in place to make init_decode() happy
1859  avctx->priv_data = s->mp3decctx[0];
1860  decode_init(avctx);
1861  s->frame = avctx->coded_frame;
1862  // Restore mp3on4 context pointer
1863  avctx->priv_data = s;
1864  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1865 
1866  /* Create a separate codec/context for each frame (first is already ok).
1867  * Each frame is 1 or 2 channels - up to 5 frames allowed
1868  */
1869  for (i = 1; i < s->frames; i++) {
1870  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1871  if (!s->mp3decctx[i])
1872  goto alloc_fail;
1873  s->mp3decctx[i]->adu_mode = 1;
1874  s->mp3decctx[i]->avctx = avctx;
1875  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
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  mp_flush(s->mp3decctx[i]);
1892 }
1893 
1894 
1895 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1896  int *got_frame_ptr, AVPacket *avpkt)
1897 {
1898  const uint8_t *buf = avpkt->data;
1899  int buf_size = avpkt->size;
1900  MP3On4DecodeContext *s = avctx->priv_data;
1901  MPADecodeContext *m;
1902  int fsize, len = buf_size, out_size = 0;
1903  uint32_t header;
1904  OUT_INT **out_samples;
1905  OUT_INT *outptr[2];
1906  int fr, ch, ret;
1907 
1908  /* get output buffer */
1909  s->frame->nb_samples = MPA_FRAME_SIZE;
1910  if ((ret = ff_get_buffer(avctx, s->frame)) < 0) {
1911  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1912  return ret;
1913  }
1914  out_samples = (OUT_INT **)s->frame->extended_data;
1915 
1916  // Discard too short frames
1917  if (buf_size < HEADER_SIZE)
1918  return AVERROR_INVALIDDATA;
1919 
1920  avctx->bit_rate = 0;
1921 
1922  ch = 0;
1923  for (fr = 0; fr < s->frames; fr++) {
1924  fsize = AV_RB16(buf) >> 4;
1925  fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1926  m = s->mp3decctx[fr];
1927  assert(m != NULL);
1928 
1929  if (fsize < HEADER_SIZE) {
1930  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1931  return AVERROR_INVALIDDATA;
1932  }
1933  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1934 
1935  if (ff_mpa_check_header(header) < 0) // Bad header, discard block
1936  break;
1937 
1939 
1940  if (ch + m->nb_channels > avctx->channels ||
1941  s->coff[fr] + m->nb_channels > avctx->channels) {
1942  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1943  "channel count\n");
1944  return AVERROR_INVALIDDATA;
1945  }
1946  ch += m->nb_channels;
1947 
1948  outptr[0] = out_samples[s->coff[fr]];
1949  if (m->nb_channels > 1)
1950  outptr[1] = out_samples[s->coff[fr] + 1];
1951 
1952  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
1953  return ret;
1954 
1955  out_size += ret;
1956  buf += fsize;
1957  len -= fsize;
1958 
1959  avctx->bit_rate += m->bit_rate;
1960  }
1961 
1962  /* update codec info */
1963  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1964 
1965  s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1966  *got_frame_ptr = 1;
1967  *(AVFrame *)data = *s->frame;
1968 
1969  return buf_size;
1970 }
1971 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
1972 
1973 #if !CONFIG_FLOAT
1974 #if CONFIG_MP1_DECODER
1975 AVCodec ff_mp1_decoder = {
1976  .name = "mp1",
1977  .type = AVMEDIA_TYPE_AUDIO,
1978  .id = AV_CODEC_ID_MP1,
1979  .priv_data_size = sizeof(MPADecodeContext),
1980  .init = decode_init,
1981  .decode = decode_frame,
1982  .capabilities = CODEC_CAP_DR1,
1983  .flush = flush,
1984  .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
1985  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
1988 };
1989 #endif
1990 #if CONFIG_MP2_DECODER
1991 AVCodec ff_mp2_decoder = {
1992  .name = "mp2",
1993  .type = AVMEDIA_TYPE_AUDIO,
1994  .id = AV_CODEC_ID_MP2,
1995  .priv_data_size = sizeof(MPADecodeContext),
1996  .init = decode_init,
1997  .decode = decode_frame,
1998  .capabilities = CODEC_CAP_DR1,
1999  .flush = flush,
2000  .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2001  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2004 };
2005 #endif
2006 #if CONFIG_MP3_DECODER
2007 AVCodec ff_mp3_decoder = {
2008  .name = "mp3",
2009  .type = AVMEDIA_TYPE_AUDIO,
2010  .id = AV_CODEC_ID_MP3,
2011  .priv_data_size = sizeof(MPADecodeContext),
2012  .init = decode_init,
2013  .decode = decode_frame,
2014  .capabilities = CODEC_CAP_DR1,
2015  .flush = flush,
2016  .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2017  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2020 };
2021 #endif
2022 #if CONFIG_MP3ADU_DECODER
2023 AVCodec ff_mp3adu_decoder = {
2024  .name = "mp3adu",
2025  .type = AVMEDIA_TYPE_AUDIO,
2026  .id = AV_CODEC_ID_MP3ADU,
2027  .priv_data_size = sizeof(MPADecodeContext),
2028  .init = decode_init,
2029  .decode = decode_frame_adu,
2030  .capabilities = CODEC_CAP_DR1,
2031  .flush = flush,
2032  .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2033  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2036 };
2037 #endif
2038 #if CONFIG_MP3ON4_DECODER
2039 AVCodec ff_mp3on4_decoder = {
2040  .name = "mp3on4",
2041  .type = AVMEDIA_TYPE_AUDIO,
2042  .id = AV_CODEC_ID_MP3ON4,
2043  .priv_data_size = sizeof(MP3On4DecodeContext),
2044  .init = decode_init_mp3on4,
2045  .close = decode_close_mp3on4,
2046  .decode = decode_frame_mp3on4,
2047  .capabilities = CODEC_CAP_DR1,
2048  .flush = flush_mp3on4,
2049  .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
2050  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2052 };
2053 #endif
2054 #endif