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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 FFmpeg.
6  *
7  * FFmpeg 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  * FFmpeg 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 FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * MPEG Audio decoder
25  */
26 
27 #include "libavutil/avassert.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/libm.h"
31 #include "avcodec.h"
32 #include "get_bits.h"
33 #include "internal.h"
34 #include "mathops.h"
35 #include "mpegaudiodsp.h"
36 
37 /*
38  * TODO:
39  * - test lsf / mpeg25 extensively.
40  */
41 
42 #include "mpegaudio.h"
43 #include "mpegaudiodecheader.h"
44 
45 #define BACKSTEP_SIZE 512
46 #define EXTRABYTES 24
47 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
48 
49 /* layer 3 "granule" */
50 typedef struct GranuleDef {
58  int table_select[3];
59  int subblock_gain[3];
62  int region_size[3]; /* number of huffman codes in each region */
63  int preflag;
64  int short_start, long_end; /* long/short band indexes */
66  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
67 } GranuleDef;
68 
69 typedef struct MPADecodeContext {
73  /* next header (used in free format parsing) */
80  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
81  GranuleDef granules[2][2]; /* Used in Layer 3 */
82  int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
90 
91 #if CONFIG_FLOAT
92 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
93 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
94 # define FIXR(x) ((float)(x))
95 # define FIXHR(x) ((float)(x))
96 # define MULH3(x, y, s) ((s)*(y)*(x))
97 # define MULLx(x, y, s) ((y)*(x))
98 # define RENAME(a) a ## _float
99 # define OUT_FMT AV_SAMPLE_FMT_FLT
100 # define OUT_FMT_P AV_SAMPLE_FMT_FLTP
101 #else
102 # define SHR(a,b) ((a)>>(b))
103 /* WARNING: only correct for positive numbers */
104 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
105 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
106 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
107 # define MULH3(x, y, s) MULH((s)*(x), y)
108 # define MULLx(x, y, s) MULL(x,y,s)
109 # define RENAME(a) a ## _fixed
110 # define OUT_FMT AV_SAMPLE_FMT_S16
111 # define OUT_FMT_P AV_SAMPLE_FMT_S16P
112 #endif
113 
114 /****************/
115 
116 #define HEADER_SIZE 4
117 
118 #include "mpegaudiodata.h"
119 #include "mpegaudiodectab.h"
120 
121 /* vlc structure for decoding layer 3 huffman tables */
122 static VLC huff_vlc[16];
124  0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
125  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
126  ][2];
127 static const int huff_vlc_tables_sizes[16] = {
128  0, 128, 128, 128, 130, 128, 154, 166,
129  142, 204, 190, 170, 542, 460, 662, 414
130 };
131 static VLC huff_quad_vlc[2];
132 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
133 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
134 /* computed from band_size_long */
135 static uint16_t band_index_long[9][23];
136 #include "mpegaudio_tablegen.h"
137 /* intensity stereo coef table */
138 static INTFLOAT is_table[2][16];
139 static INTFLOAT is_table_lsf[2][2][16];
140 static INTFLOAT csa_table[8][4];
141 
142 static int16_t division_tab3[1<<6 ];
143 static int16_t division_tab5[1<<8 ];
144 static int16_t division_tab9[1<<11];
145 
146 static int16_t * const division_tabs[4] = {
148 };
149 
150 /* lower 2 bits: modulo 3, higher bits: shift */
151 static uint16_t scale_factor_modshift[64];
152 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
154 /* mult table for layer 2 group quantization */
155 
156 #define SCALE_GEN(v) \
157 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
158 
159 static const int32_t scale_factor_mult2[3][3] = {
160  SCALE_GEN(4.0 / 3.0), /* 3 steps */
161  SCALE_GEN(4.0 / 5.0), /* 5 steps */
162  SCALE_GEN(4.0 / 9.0), /* 9 steps */
163 };
164 
165 /**
166  * Convert region offsets to region sizes and truncate
167  * size to big_values.
168  */
170 {
171  int i, k, j = 0;
172  g->region_size[2] = 576 / 2;
173  for (i = 0; i < 3; i++) {
174  k = FFMIN(g->region_size[i], g->big_values);
175  g->region_size[i] = k - j;
176  j = k;
177  }
178 }
179 
181 {
182  if (g->block_type == 2) {
183  if (s->sample_rate_index != 8)
184  g->region_size[0] = (36 / 2);
185  else
186  g->region_size[0] = (72 / 2);
187  } else {
188  if (s->sample_rate_index <= 2)
189  g->region_size[0] = (36 / 2);
190  else if (s->sample_rate_index != 8)
191  g->region_size[0] = (54 / 2);
192  else
193  g->region_size[0] = (108 / 2);
194  }
195  g->region_size[1] = (576 / 2);
196 }
197 
198 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
199 {
200  int l;
201  g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
202  /* should not overflow */
203  l = FFMIN(ra1 + ra2 + 2, 22);
204  g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
205 }
206 
208 {
209  if (g->block_type == 2) {
210  if (g->switch_point) {
211  if(s->sample_rate_index == 8)
212  av_log_ask_for_sample(s->avctx, "switch point in 8khz\n");
213  /* if switched mode, we handle the 36 first samples as
214  long blocks. For 8000Hz, we handle the 72 first
215  exponents as long blocks */
216  if (s->sample_rate_index <= 2)
217  g->long_end = 8;
218  else
219  g->long_end = 6;
220 
221  g->short_start = 3;
222  } else {
223  g->long_end = 0;
224  g->short_start = 0;
225  }
226  } else {
227  g->short_start = 13;
228  g->long_end = 22;
229  }
230 }
231 
232 /* layer 1 unscaling */
233 /* n = number of bits of the mantissa minus 1 */
234 static inline int l1_unscale(int n, int mant, int scale_factor)
235 {
236  int shift, mod;
237  int64_t val;
238 
239  shift = scale_factor_modshift[scale_factor];
240  mod = shift & 3;
241  shift >>= 2;
242  val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
243  shift += n;
244  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
245  return (int)((val + (1LL << (shift - 1))) >> shift);
246 }
247 
248 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
249 {
250  int shift, mod, val;
251 
252  shift = scale_factor_modshift[scale_factor];
253  mod = shift & 3;
254  shift >>= 2;
255 
256  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
257  /* NOTE: at this point, 0 <= shift <= 21 */
258  if (shift > 0)
259  val = (val + (1 << (shift - 1))) >> shift;
260  return val;
261 }
262 
263 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
264 static inline int l3_unscale(int value, int exponent)
265 {
266  unsigned int m;
267  int e;
268 
269  e = table_4_3_exp [4 * value + (exponent & 3)];
270  m = table_4_3_value[4 * value + (exponent & 3)];
271  e -= exponent >> 2;
272 #ifdef DEBUG
273  if(e < 1)
274  av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
275 #endif
276  if (e > 31)
277  return 0;
278  m = (m + (1 << (e - 1))) >> e;
279 
280  return m;
281 }
282 
283 static av_cold void decode_init_static(void)
284 {
285  int i, j, k;
286  int offset;
287 
288  /* scale factors table for layer 1/2 */
289  for (i = 0; i < 64; i++) {
290  int shift, mod;
291  /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
292  shift = i / 3;
293  mod = i % 3;
294  scale_factor_modshift[i] = mod | (shift << 2);
295  }
296 
297  /* scale factor multiply for layer 1 */
298  for (i = 0; i < 15; i++) {
299  int n, norm;
300  n = i + 2;
301  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
302  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
303  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
304  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
305  av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
306  scale_factor_mult[i][0],
307  scale_factor_mult[i][1],
308  scale_factor_mult[i][2]);
309  }
310 
311  RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
312 
313  /* huffman decode tables */
314  offset = 0;
315  for (i = 1; i < 16; i++) {
316  const HuffTable *h = &mpa_huff_tables[i];
317  int xsize, x, y;
318  uint8_t tmp_bits [512] = { 0 };
319  uint16_t tmp_codes[512] = { 0 };
320 
321  xsize = h->xsize;
322 
323  j = 0;
324  for (x = 0; x < xsize; x++) {
325  for (y = 0; y < xsize; y++) {
326  tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
327  tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
328  }
329  }
330 
331  /* XXX: fail test */
332  huff_vlc[i].table = huff_vlc_tables+offset;
333  huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
334  init_vlc(&huff_vlc[i], 7, 512,
335  tmp_bits, 1, 1, tmp_codes, 2, 2,
337  offset += huff_vlc_tables_sizes[i];
338  }
340 
341  offset = 0;
342  for (i = 0; i < 2; i++) {
343  huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
344  huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
345  init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
346  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
348  offset += huff_quad_vlc_tables_sizes[i];
349  }
351 
352  for (i = 0; i < 9; i++) {
353  k = 0;
354  for (j = 0; j < 22; j++) {
355  band_index_long[i][j] = k;
356  k += band_size_long[i][j];
357  }
358  band_index_long[i][22] = k;
359  }
360 
361  /* compute n ^ (4/3) and store it in mantissa/exp format */
362 
364 
365  for (i = 0; i < 4; i++) {
366  if (ff_mpa_quant_bits[i] < 0) {
367  for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
368  int val1, val2, val3, steps;
369  int val = j;
370  steps = ff_mpa_quant_steps[i];
371  val1 = val % steps;
372  val /= steps;
373  val2 = val % steps;
374  val3 = val / steps;
375  division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
376  }
377  }
378  }
379 
380 
381  for (i = 0; i < 7; i++) {
382  float f;
383  INTFLOAT v;
384  if (i != 6) {
385  f = tan((double)i * M_PI / 12.0);
386  v = FIXR(f / (1.0 + f));
387  } else {
388  v = FIXR(1.0);
389  }
390  is_table[0][ i] = v;
391  is_table[1][6 - i] = v;
392  }
393  /* invalid values */
394  for (i = 7; i < 16; i++)
395  is_table[0][i] = is_table[1][i] = 0.0;
396 
397  for (i = 0; i < 16; i++) {
398  double f;
399  int e, k;
400 
401  for (j = 0; j < 2; j++) {
402  e = -(j + 1) * ((i + 1) >> 1);
403  f = exp2(e / 4.0);
404  k = i & 1;
405  is_table_lsf[j][k ^ 1][i] = FIXR(f);
406  is_table_lsf[j][k ][i] = FIXR(1.0);
407  av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
408  i, j, (float) is_table_lsf[j][0][i],
409  (float) is_table_lsf[j][1][i]);
410  }
411  }
412 
413  for (i = 0; i < 8; i++) {
414  float ci, cs, ca;
415  ci = ci_table[i];
416  cs = 1.0 / sqrt(1.0 + ci * ci);
417  ca = cs * ci;
418 #if !CONFIG_FLOAT
419  csa_table[i][0] = FIXHR(cs/4);
420  csa_table[i][1] = FIXHR(ca/4);
421  csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
422  csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
423 #else
424  csa_table[i][0] = cs;
425  csa_table[i][1] = ca;
426  csa_table[i][2] = ca + cs;
427  csa_table[i][3] = ca - cs;
428 #endif
429  }
430 }
431 
432 static av_cold int decode_init(AVCodecContext * avctx)
433 {
434  static int initialized_tables = 0;
435  MPADecodeContext *s = avctx->priv_data;
436 
437  if (!initialized_tables) {
439  initialized_tables = 1;
440  }
441 
442  s->avctx = avctx;
443 
445  ff_mpadsp_init(&s->mpadsp);
446 
447  if (avctx->request_sample_fmt == OUT_FMT &&
448  avctx->codec_id != AV_CODEC_ID_MP3ON4)
449  avctx->sample_fmt = OUT_FMT;
450  else
451  avctx->sample_fmt = OUT_FMT_P;
452  s->err_recognition = avctx->err_recognition;
453 
454  if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
455  s->adu_mode = 1;
456 
457  return 0;
458 }
459 
460 #define C3 FIXHR(0.86602540378443864676/2)
461 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
462 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
463 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
464 
465 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
466  cases. */
467 static void imdct12(INTFLOAT *out, INTFLOAT *in)
468 {
469  INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
470 
471  in0 = in[0*3];
472  in1 = in[1*3] + in[0*3];
473  in2 = in[2*3] + in[1*3];
474  in3 = in[3*3] + in[2*3];
475  in4 = in[4*3] + in[3*3];
476  in5 = in[5*3] + in[4*3];
477  in5 += in3;
478  in3 += in1;
479 
480  in2 = MULH3(in2, C3, 2);
481  in3 = MULH3(in3, C3, 4);
482 
483  t1 = in0 - in4;
484  t2 = MULH3(in1 - in5, C4, 2);
485 
486  out[ 7] =
487  out[10] = t1 + t2;
488  out[ 1] =
489  out[ 4] = t1 - t2;
490 
491  in0 += SHR(in4, 1);
492  in4 = in0 + in2;
493  in5 += 2*in1;
494  in1 = MULH3(in5 + in3, C5, 1);
495  out[ 8] =
496  out[ 9] = in4 + in1;
497  out[ 2] =
498  out[ 3] = in4 - in1;
499 
500  in0 -= in2;
501  in5 = MULH3(in5 - in3, C6, 2);
502  out[ 0] =
503  out[ 5] = in0 - in5;
504  out[ 6] =
505  out[11] = in0 + in5;
506 }
507 
508 /* return the number of decoded frames */
510 {
511  int bound, i, v, n, ch, j, mant;
512  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
513  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
514 
515  if (s->mode == MPA_JSTEREO)
516  bound = (s->mode_ext + 1) * 4;
517  else
518  bound = SBLIMIT;
519 
520  /* allocation bits */
521  for (i = 0; i < bound; i++) {
522  for (ch = 0; ch < s->nb_channels; ch++) {
523  allocation[ch][i] = get_bits(&s->gb, 4);
524  }
525  }
526  for (i = bound; i < SBLIMIT; i++)
527  allocation[0][i] = get_bits(&s->gb, 4);
528 
529  /* scale factors */
530  for (i = 0; i < bound; i++) {
531  for (ch = 0; ch < s->nb_channels; ch++) {
532  if (allocation[ch][i])
533  scale_factors[ch][i] = get_bits(&s->gb, 6);
534  }
535  }
536  for (i = bound; i < SBLIMIT; i++) {
537  if (allocation[0][i]) {
538  scale_factors[0][i] = get_bits(&s->gb, 6);
539  scale_factors[1][i] = get_bits(&s->gb, 6);
540  }
541  }
542 
543  /* compute samples */
544  for (j = 0; j < 12; j++) {
545  for (i = 0; i < bound; i++) {
546  for (ch = 0; ch < s->nb_channels; ch++) {
547  n = allocation[ch][i];
548  if (n) {
549  mant = get_bits(&s->gb, n + 1);
550  v = l1_unscale(n, mant, scale_factors[ch][i]);
551  } else {
552  v = 0;
553  }
554  s->sb_samples[ch][j][i] = v;
555  }
556  }
557  for (i = bound; i < SBLIMIT; i++) {
558  n = allocation[0][i];
559  if (n) {
560  mant = get_bits(&s->gb, n + 1);
561  v = l1_unscale(n, mant, scale_factors[0][i]);
562  s->sb_samples[0][j][i] = v;
563  v = l1_unscale(n, mant, scale_factors[1][i]);
564  s->sb_samples[1][j][i] = v;
565  } else {
566  s->sb_samples[0][j][i] = 0;
567  s->sb_samples[1][j][i] = 0;
568  }
569  }
570  }
571  return 12;
572 }
573 
575 {
576  int sblimit; /* number of used subbands */
577  const unsigned char *alloc_table;
578  int table, bit_alloc_bits, i, j, ch, bound, v;
579  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
580  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
581  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
582  int scale, qindex, bits, steps, k, l, m, b;
583 
584  /* select decoding table */
585  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
586  s->sample_rate, s->lsf);
587  sblimit = ff_mpa_sblimit_table[table];
588  alloc_table = ff_mpa_alloc_tables[table];
589 
590  if (s->mode == MPA_JSTEREO)
591  bound = (s->mode_ext + 1) * 4;
592  else
593  bound = sblimit;
594 
595  av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
596 
597  /* sanity check */
598  if (bound > sblimit)
599  bound = sblimit;
600 
601  /* parse bit allocation */
602  j = 0;
603  for (i = 0; i < bound; i++) {
604  bit_alloc_bits = alloc_table[j];
605  for (ch = 0; ch < s->nb_channels; ch++)
606  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
607  j += 1 << bit_alloc_bits;
608  }
609  for (i = bound; i < sblimit; i++) {
610  bit_alloc_bits = alloc_table[j];
611  v = get_bits(&s->gb, bit_alloc_bits);
612  bit_alloc[0][i] = v;
613  bit_alloc[1][i] = v;
614  j += 1 << bit_alloc_bits;
615  }
616 
617  /* scale codes */
618  for (i = 0; i < sblimit; i++) {
619  for (ch = 0; ch < s->nb_channels; ch++) {
620  if (bit_alloc[ch][i])
621  scale_code[ch][i] = get_bits(&s->gb, 2);
622  }
623  }
624 
625  /* scale factors */
626  for (i = 0; i < sblimit; i++) {
627  for (ch = 0; ch < s->nb_channels; ch++) {
628  if (bit_alloc[ch][i]) {
629  sf = scale_factors[ch][i];
630  switch (scale_code[ch][i]) {
631  default:
632  case 0:
633  sf[0] = get_bits(&s->gb, 6);
634  sf[1] = get_bits(&s->gb, 6);
635  sf[2] = get_bits(&s->gb, 6);
636  break;
637  case 2:
638  sf[0] = get_bits(&s->gb, 6);
639  sf[1] = sf[0];
640  sf[2] = sf[0];
641  break;
642  case 1:
643  sf[0] = get_bits(&s->gb, 6);
644  sf[2] = get_bits(&s->gb, 6);
645  sf[1] = sf[0];
646  break;
647  case 3:
648  sf[0] = get_bits(&s->gb, 6);
649  sf[2] = get_bits(&s->gb, 6);
650  sf[1] = sf[2];
651  break;
652  }
653  }
654  }
655  }
656 
657  /* samples */
658  for (k = 0; k < 3; k++) {
659  for (l = 0; l < 12; l += 3) {
660  j = 0;
661  for (i = 0; i < bound; i++) {
662  bit_alloc_bits = alloc_table[j];
663  for (ch = 0; ch < s->nb_channels; ch++) {
664  b = bit_alloc[ch][i];
665  if (b) {
666  scale = scale_factors[ch][i][k];
667  qindex = alloc_table[j+b];
668  bits = ff_mpa_quant_bits[qindex];
669  if (bits < 0) {
670  int v2;
671  /* 3 values at the same time */
672  v = get_bits(&s->gb, -bits);
673  v2 = division_tabs[qindex][v];
674  steps = ff_mpa_quant_steps[qindex];
675 
676  s->sb_samples[ch][k * 12 + l + 0][i] =
677  l2_unscale_group(steps, v2 & 15, scale);
678  s->sb_samples[ch][k * 12 + l + 1][i] =
679  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
680  s->sb_samples[ch][k * 12 + l + 2][i] =
681  l2_unscale_group(steps, v2 >> 8 , scale);
682  } else {
683  for (m = 0; m < 3; m++) {
684  v = get_bits(&s->gb, bits);
685  v = l1_unscale(bits - 1, v, scale);
686  s->sb_samples[ch][k * 12 + l + m][i] = v;
687  }
688  }
689  } else {
690  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
691  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
692  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
693  }
694  }
695  /* next subband in alloc table */
696  j += 1 << bit_alloc_bits;
697  }
698  /* XXX: find a way to avoid this duplication of code */
699  for (i = bound; i < sblimit; i++) {
700  bit_alloc_bits = alloc_table[j];
701  b = bit_alloc[0][i];
702  if (b) {
703  int mant, scale0, scale1;
704  scale0 = scale_factors[0][i][k];
705  scale1 = scale_factors[1][i][k];
706  qindex = alloc_table[j+b];
707  bits = ff_mpa_quant_bits[qindex];
708  if (bits < 0) {
709  /* 3 values at the same time */
710  v = get_bits(&s->gb, -bits);
711  steps = ff_mpa_quant_steps[qindex];
712  mant = v % steps;
713  v = v / steps;
714  s->sb_samples[0][k * 12 + l + 0][i] =
715  l2_unscale_group(steps, mant, scale0);
716  s->sb_samples[1][k * 12 + l + 0][i] =
717  l2_unscale_group(steps, mant, scale1);
718  mant = v % steps;
719  v = v / steps;
720  s->sb_samples[0][k * 12 + l + 1][i] =
721  l2_unscale_group(steps, mant, scale0);
722  s->sb_samples[1][k * 12 + l + 1][i] =
723  l2_unscale_group(steps, mant, scale1);
724  s->sb_samples[0][k * 12 + l + 2][i] =
725  l2_unscale_group(steps, v, scale0);
726  s->sb_samples[1][k * 12 + l + 2][i] =
727  l2_unscale_group(steps, v, scale1);
728  } else {
729  for (m = 0; m < 3; m++) {
730  mant = get_bits(&s->gb, bits);
731  s->sb_samples[0][k * 12 + l + m][i] =
732  l1_unscale(bits - 1, mant, scale0);
733  s->sb_samples[1][k * 12 + l + m][i] =
734  l1_unscale(bits - 1, mant, scale1);
735  }
736  }
737  } else {
738  s->sb_samples[0][k * 12 + l + 0][i] = 0;
739  s->sb_samples[0][k * 12 + l + 1][i] = 0;
740  s->sb_samples[0][k * 12 + l + 2][i] = 0;
741  s->sb_samples[1][k * 12 + l + 0][i] = 0;
742  s->sb_samples[1][k * 12 + l + 1][i] = 0;
743  s->sb_samples[1][k * 12 + l + 2][i] = 0;
744  }
745  /* next subband in alloc table */
746  j += 1 << bit_alloc_bits;
747  }
748  /* fill remaining samples to zero */
749  for (i = sblimit; i < SBLIMIT; i++) {
750  for (ch = 0; ch < s->nb_channels; ch++) {
751  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
752  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
753  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
754  }
755  }
756  }
757  }
758  return 3 * 12;
759 }
760 
761 #define SPLIT(dst,sf,n) \
762  if (n == 3) { \
763  int m = (sf * 171) >> 9; \
764  dst = sf - 3 * m; \
765  sf = m; \
766  } else if (n == 4) { \
767  dst = sf & 3; \
768  sf >>= 2; \
769  } else if (n == 5) { \
770  int m = (sf * 205) >> 10; \
771  dst = sf - 5 * m; \
772  sf = m; \
773  } else if (n == 6) { \
774  int m = (sf * 171) >> 10; \
775  dst = sf - 6 * m; \
776  sf = m; \
777  } else { \
778  dst = 0; \
779  }
780 
781 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
782  int n3)
783 {
784  SPLIT(slen[3], sf, n3)
785  SPLIT(slen[2], sf, n2)
786  SPLIT(slen[1], sf, n1)
787  slen[0] = sf;
788 }
789 
791  int16_t *exponents)
792 {
793  const uint8_t *bstab, *pretab;
794  int len, i, j, k, l, v0, shift, gain, gains[3];
795  int16_t *exp_ptr;
796 
797  exp_ptr = exponents;
798  gain = g->global_gain - 210;
799  shift = g->scalefac_scale + 1;
800 
801  bstab = band_size_long[s->sample_rate_index];
802  pretab = mpa_pretab[g->preflag];
803  for (i = 0; i < g->long_end; i++) {
804  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
805  len = bstab[i];
806  for (j = len; j > 0; j--)
807  *exp_ptr++ = v0;
808  }
809 
810  if (g->short_start < 13) {
811  bstab = band_size_short[s->sample_rate_index];
812  gains[0] = gain - (g->subblock_gain[0] << 3);
813  gains[1] = gain - (g->subblock_gain[1] << 3);
814  gains[2] = gain - (g->subblock_gain[2] << 3);
815  k = g->long_end;
816  for (i = g->short_start; i < 13; i++) {
817  len = bstab[i];
818  for (l = 0; l < 3; l++) {
819  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
820  for (j = len; j > 0; j--)
821  *exp_ptr++ = v0;
822  }
823  }
824  }
825 }
826 
827 /* handle n = 0 too */
828 static inline int get_bitsz(GetBitContext *s, int n)
829 {
830  return n ? get_bits(s, n) : 0;
831 }
832 
833 
834 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
835  int *end_pos2)
836 {
837  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
838  s->gb = s->in_gb;
839  s->in_gb.buffer = NULL;
840  av_assert2((get_bits_count(&s->gb) & 7) == 0);
841  skip_bits_long(&s->gb, *pos - *end_pos);
842  *end_pos2 =
843  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
844  *pos = get_bits_count(&s->gb);
845  }
846 }
847 
848 /* Following is a optimized code for
849  INTFLOAT v = *src
850  if(get_bits1(&s->gb))
851  v = -v;
852  *dst = v;
853 */
854 #if CONFIG_FLOAT
855 #define READ_FLIP_SIGN(dst,src) \
856  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
857  AV_WN32A(dst, v);
858 #else
859 #define READ_FLIP_SIGN(dst,src) \
860  v = -get_bits1(&s->gb); \
861  *(dst) = (*(src) ^ v) - v;
862 #endif
863 
865  int16_t *exponents, int end_pos2)
866 {
867  int s_index;
868  int i;
869  int last_pos, bits_left;
870  VLC *vlc;
871  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
872 
873  /* low frequencies (called big values) */
874  s_index = 0;
875  for (i = 0; i < 3; i++) {
876  int j, k, l, linbits;
877  j = g->region_size[i];
878  if (j == 0)
879  continue;
880  /* select vlc table */
881  k = g->table_select[i];
882  l = mpa_huff_data[k][0];
883  linbits = mpa_huff_data[k][1];
884  vlc = &huff_vlc[l];
885 
886  if (!l) {
887  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
888  s_index += 2 * j;
889  continue;
890  }
891 
892  /* read huffcode and compute each couple */
893  for (; j > 0; j--) {
894  int exponent, x, y;
895  int v;
896  int pos = get_bits_count(&s->gb);
897 
898  if (pos >= end_pos){
899  switch_buffer(s, &pos, &end_pos, &end_pos2);
900  if (pos >= end_pos)
901  break;
902  }
903  y = get_vlc2(&s->gb, vlc->table, 7, 3);
904 
905  if (!y) {
906  g->sb_hybrid[s_index ] =
907  g->sb_hybrid[s_index+1] = 0;
908  s_index += 2;
909  continue;
910  }
911 
912  exponent= exponents[s_index];
913 
914  av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
915  i, g->region_size[i] - j, x, y, exponent);
916  if (y & 16) {
917  x = y >> 5;
918  y = y & 0x0f;
919  if (x < 15) {
920  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
921  } else {
922  x += get_bitsz(&s->gb, linbits);
923  v = l3_unscale(x, exponent);
924  if (get_bits1(&s->gb))
925  v = -v;
926  g->sb_hybrid[s_index] = v;
927  }
928  if (y < 15) {
929  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
930  } else {
931  y += get_bitsz(&s->gb, linbits);
932  v = l3_unscale(y, exponent);
933  if (get_bits1(&s->gb))
934  v = -v;
935  g->sb_hybrid[s_index+1] = v;
936  }
937  } else {
938  x = y >> 5;
939  y = y & 0x0f;
940  x += y;
941  if (x < 15) {
942  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
943  } else {
944  x += get_bitsz(&s->gb, linbits);
945  v = l3_unscale(x, exponent);
946  if (get_bits1(&s->gb))
947  v = -v;
948  g->sb_hybrid[s_index+!!y] = v;
949  }
950  g->sb_hybrid[s_index + !y] = 0;
951  }
952  s_index += 2;
953  }
954  }
955 
956  /* high frequencies */
957  vlc = &huff_quad_vlc[g->count1table_select];
958  last_pos = 0;
959  while (s_index <= 572) {
960  int pos, code;
961  pos = get_bits_count(&s->gb);
962  if (pos >= end_pos) {
963  if (pos > end_pos2 && last_pos) {
964  /* some encoders generate an incorrect size for this
965  part. We must go back into the data */
966  s_index -= 4;
967  skip_bits_long(&s->gb, last_pos - pos);
968  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
970  s_index=0;
971  break;
972  }
973  switch_buffer(s, &pos, &end_pos, &end_pos2);
974  if (pos >= end_pos)
975  break;
976  }
977  last_pos = pos;
978 
979  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
980  av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
981  g->sb_hybrid[s_index+0] =
982  g->sb_hybrid[s_index+1] =
983  g->sb_hybrid[s_index+2] =
984  g->sb_hybrid[s_index+3] = 0;
985  while (code) {
986  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
987  int v;
988  int pos = s_index + idxtab[code];
989  code ^= 8 >> idxtab[code];
990  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
991  }
992  s_index += 4;
993  }
994  /* skip extension bits */
995  bits_left = end_pos2 - get_bits_count(&s->gb);
996  if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
997  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
998  s_index=0;
999  } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
1000  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1001  s_index = 0;
1002  }
1003  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
1004  skip_bits_long(&s->gb, bits_left);
1005 
1006  i = get_bits_count(&s->gb);
1007  switch_buffer(s, &i, &end_pos, &end_pos2);
1008 
1009  return 0;
1010 }
1011 
1012 /* Reorder short blocks from bitstream order to interleaved order. It
1013  would be faster to do it in parsing, but the code would be far more
1014  complicated */
1016 {
1017  int i, j, len;
1018  INTFLOAT *ptr, *dst, *ptr1;
1019  INTFLOAT tmp[576];
1020 
1021  if (g->block_type != 2)
1022  return;
1023 
1024  if (g->switch_point) {
1025  if (s->sample_rate_index != 8)
1026  ptr = g->sb_hybrid + 36;
1027  else
1028  ptr = g->sb_hybrid + 72;
1029  } else {
1030  ptr = g->sb_hybrid;
1031  }
1032 
1033  for (i = g->short_start; i < 13; i++) {
1034  len = band_size_short[s->sample_rate_index][i];
1035  ptr1 = ptr;
1036  dst = tmp;
1037  for (j = len; j > 0; j--) {
1038  *dst++ = ptr[0*len];
1039  *dst++ = ptr[1*len];
1040  *dst++ = ptr[2*len];
1041  ptr++;
1042  }
1043  ptr += 2 * len;
1044  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1045  }
1046 }
1047 
1048 #define ISQRT2 FIXR(0.70710678118654752440)
1049 
1051 {
1052  int i, j, k, l;
1053  int sf_max, sf, len, non_zero_found;
1054  INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1055  int non_zero_found_short[3];
1056 
1057  /* intensity stereo */
1058  if (s->mode_ext & MODE_EXT_I_STEREO) {
1059  if (!s->lsf) {
1060  is_tab = is_table;
1061  sf_max = 7;
1062  } else {
1063  is_tab = is_table_lsf[g1->scalefac_compress & 1];
1064  sf_max = 16;
1065  }
1066 
1067  tab0 = g0->sb_hybrid + 576;
1068  tab1 = g1->sb_hybrid + 576;
1069 
1070  non_zero_found_short[0] = 0;
1071  non_zero_found_short[1] = 0;
1072  non_zero_found_short[2] = 0;
1073  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1074  for (i = 12; i >= g1->short_start; i--) {
1075  /* for last band, use previous scale factor */
1076  if (i != 11)
1077  k -= 3;
1078  len = band_size_short[s->sample_rate_index][i];
1079  for (l = 2; l >= 0; l--) {
1080  tab0 -= len;
1081  tab1 -= len;
1082  if (!non_zero_found_short[l]) {
1083  /* test if non zero band. if so, stop doing i-stereo */
1084  for (j = 0; j < len; j++) {
1085  if (tab1[j] != 0) {
1086  non_zero_found_short[l] = 1;
1087  goto found1;
1088  }
1089  }
1090  sf = g1->scale_factors[k + l];
1091  if (sf >= sf_max)
1092  goto found1;
1093 
1094  v1 = is_tab[0][sf];
1095  v2 = is_tab[1][sf];
1096  for (j = 0; j < len; j++) {
1097  tmp0 = tab0[j];
1098  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1099  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1100  }
1101  } else {
1102 found1:
1103  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1104  /* lower part of the spectrum : do ms stereo
1105  if enabled */
1106  for (j = 0; j < len; j++) {
1107  tmp0 = tab0[j];
1108  tmp1 = tab1[j];
1109  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1110  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1111  }
1112  }
1113  }
1114  }
1115  }
1116 
1117  non_zero_found = non_zero_found_short[0] |
1118  non_zero_found_short[1] |
1119  non_zero_found_short[2];
1120 
1121  for (i = g1->long_end - 1;i >= 0;i--) {
1122  len = band_size_long[s->sample_rate_index][i];
1123  tab0 -= len;
1124  tab1 -= len;
1125  /* test if non zero band. if so, stop doing i-stereo */
1126  if (!non_zero_found) {
1127  for (j = 0; j < len; j++) {
1128  if (tab1[j] != 0) {
1129  non_zero_found = 1;
1130  goto found2;
1131  }
1132  }
1133  /* for last band, use previous scale factor */
1134  k = (i == 21) ? 20 : i;
1135  sf = g1->scale_factors[k];
1136  if (sf >= sf_max)
1137  goto found2;
1138  v1 = is_tab[0][sf];
1139  v2 = is_tab[1][sf];
1140  for (j = 0; j < len; j++) {
1141  tmp0 = tab0[j];
1142  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1143  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1144  }
1145  } else {
1146 found2:
1147  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1148  /* lower part of the spectrum : do ms stereo
1149  if enabled */
1150  for (j = 0; j < len; j++) {
1151  tmp0 = tab0[j];
1152  tmp1 = tab1[j];
1153  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1154  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1155  }
1156  }
1157  }
1158  }
1159  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1160  /* ms stereo ONLY */
1161  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1162  global gain */
1163 #if CONFIG_FLOAT
1164  s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1165 #else
1166  tab0 = g0->sb_hybrid;
1167  tab1 = g1->sb_hybrid;
1168  for (i = 0; i < 576; i++) {
1169  tmp0 = tab0[i];
1170  tmp1 = tab1[i];
1171  tab0[i] = tmp0 + tmp1;
1172  tab1[i] = tmp0 - tmp1;
1173  }
1174 #endif
1175  }
1176 }
1177 
1178 #if CONFIG_FLOAT
1179 #if HAVE_MIPSFPU
1181 #endif /* HAVE_MIPSFPU */
1182 #else
1183 #if HAVE_MIPSDSPR1
1185 #endif /* HAVE_MIPSDSPR1 */
1186 #endif /* CONFIG_FLOAT */
1187 
1188 #ifndef compute_antialias
1189 #if CONFIG_FLOAT
1190 #define AA(j) do { \
1191  float tmp0 = ptr[-1-j]; \
1192  float tmp1 = ptr[ j]; \
1193  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1194  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1195  } while (0)
1196 #else
1197 #define AA(j) do { \
1198  int tmp0 = ptr[-1-j]; \
1199  int tmp1 = ptr[ j]; \
1200  int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1201  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1202  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1203  } while (0)
1204 #endif
1205 
1207 {
1208  INTFLOAT *ptr;
1209  int n, i;
1210 
1211  /* we antialias only "long" bands */
1212  if (g->block_type == 2) {
1213  if (!g->switch_point)
1214  return;
1215  /* XXX: check this for 8000Hz case */
1216  n = 1;
1217  } else {
1218  n = SBLIMIT - 1;
1219  }
1220 
1221  ptr = g->sb_hybrid + 18;
1222  for (i = n; i > 0; i--) {
1223  AA(0);
1224  AA(1);
1225  AA(2);
1226  AA(3);
1227  AA(4);
1228  AA(5);
1229  AA(6);
1230  AA(7);
1231 
1232  ptr += 18;
1233  }
1234 }
1235 #endif /* compute_antialias */
1236 
1238  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1239 {
1240  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1241  INTFLOAT out2[12];
1242  int i, j, mdct_long_end, sblimit;
1243 
1244  /* find last non zero block */
1245  ptr = g->sb_hybrid + 576;
1246  ptr1 = g->sb_hybrid + 2 * 18;
1247  while (ptr >= ptr1) {
1248  int32_t *p;
1249  ptr -= 6;
1250  p = (int32_t*)ptr;
1251  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1252  break;
1253  }
1254  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1255 
1256  if (g->block_type == 2) {
1257  /* XXX: check for 8000 Hz */
1258  if (g->switch_point)
1259  mdct_long_end = 2;
1260  else
1261  mdct_long_end = 0;
1262  } else {
1263  mdct_long_end = sblimit;
1264  }
1265 
1266  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1267  mdct_long_end, g->switch_point,
1268  g->block_type);
1269 
1270  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1271  ptr = g->sb_hybrid + 18 * mdct_long_end;
1272 
1273  for (j = mdct_long_end; j < sblimit; j++) {
1274  /* select frequency inversion */
1275  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1276  out_ptr = sb_samples + j;
1277 
1278  for (i = 0; i < 6; i++) {
1279  *out_ptr = buf[4*i];
1280  out_ptr += SBLIMIT;
1281  }
1282  imdct12(out2, ptr + 0);
1283  for (i = 0; i < 6; i++) {
1284  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1285  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1286  out_ptr += SBLIMIT;
1287  }
1288  imdct12(out2, ptr + 1);
1289  for (i = 0; i < 6; i++) {
1290  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1291  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1292  out_ptr += SBLIMIT;
1293  }
1294  imdct12(out2, ptr + 2);
1295  for (i = 0; i < 6; i++) {
1296  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1297  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1298  buf[4*(i + 6*2)] = 0;
1299  }
1300  ptr += 18;
1301  buf += (j&3) != 3 ? 1 : (4*18-3);
1302  }
1303  /* zero bands */
1304  for (j = sblimit; j < SBLIMIT; j++) {
1305  /* overlap */
1306  out_ptr = sb_samples + j;
1307  for (i = 0; i < 18; i++) {
1308  *out_ptr = buf[4*i];
1309  buf[4*i] = 0;
1310  out_ptr += SBLIMIT;
1311  }
1312  buf += (j&3) != 3 ? 1 : (4*18-3);
1313  }
1314 }
1315 
1316 /* main layer3 decoding function */
1318 {
1319  int nb_granules, main_data_begin;
1320  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1321  GranuleDef *g;
1322  int16_t exponents[576]; //FIXME try INTFLOAT
1323 
1324  /* read side info */
1325  if (s->lsf) {
1326  main_data_begin = get_bits(&s->gb, 8);
1327  skip_bits(&s->gb, s->nb_channels);
1328  nb_granules = 1;
1329  } else {
1330  main_data_begin = get_bits(&s->gb, 9);
1331  if (s->nb_channels == 2)
1332  skip_bits(&s->gb, 3);
1333  else
1334  skip_bits(&s->gb, 5);
1335  nb_granules = 2;
1336  for (ch = 0; ch < s->nb_channels; ch++) {
1337  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1338  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1339  }
1340  }
1341 
1342  for (gr = 0; gr < nb_granules; gr++) {
1343  for (ch = 0; ch < s->nb_channels; ch++) {
1344  av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1345  g = &s->granules[ch][gr];
1346  g->part2_3_length = get_bits(&s->gb, 12);
1347  g->big_values = get_bits(&s->gb, 9);
1348  if (g->big_values > 288) {
1349  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1350  return AVERROR_INVALIDDATA;
1351  }
1352 
1353  g->global_gain = get_bits(&s->gb, 8);
1354  /* if MS stereo only is selected, we precompute the
1355  1/sqrt(2) renormalization factor */
1356  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1358  g->global_gain -= 2;
1359  if (s->lsf)
1360  g->scalefac_compress = get_bits(&s->gb, 9);
1361  else
1362  g->scalefac_compress = get_bits(&s->gb, 4);
1363  blocksplit_flag = get_bits1(&s->gb);
1364  if (blocksplit_flag) {
1365  g->block_type = get_bits(&s->gb, 2);
1366  if (g->block_type == 0) {
1367  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1368  return AVERROR_INVALIDDATA;
1369  }
1370  g->switch_point = get_bits1(&s->gb);
1371  for (i = 0; i < 2; i++)
1372  g->table_select[i] = get_bits(&s->gb, 5);
1373  for (i = 0; i < 3; i++)
1374  g->subblock_gain[i] = get_bits(&s->gb, 3);
1375  ff_init_short_region(s, g);
1376  } else {
1377  int region_address1, region_address2;
1378  g->block_type = 0;
1379  g->switch_point = 0;
1380  for (i = 0; i < 3; i++)
1381  g->table_select[i] = get_bits(&s->gb, 5);
1382  /* compute huffman coded region sizes */
1383  region_address1 = get_bits(&s->gb, 4);
1384  region_address2 = get_bits(&s->gb, 3);
1385  av_dlog(s->avctx, "region1=%d region2=%d\n",
1386  region_address1, region_address2);
1387  ff_init_long_region(s, g, region_address1, region_address2);
1388  }
1391 
1392  g->preflag = 0;
1393  if (!s->lsf)
1394  g->preflag = get_bits1(&s->gb);
1395  g->scalefac_scale = get_bits1(&s->gb);
1396  g->count1table_select = get_bits1(&s->gb);
1397  av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1398  g->block_type, g->switch_point);
1399  }
1400  }
1401 
1402  if (!s->adu_mode) {
1403  int skip;
1404  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1405  int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES);
1406  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1407  /* now we get bits from the main_data_begin offset */
1408  av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1409  main_data_begin, s->last_buf_size);
1410 
1411  memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
1412  s->in_gb = s->gb;
1413  init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1414 #if !UNCHECKED_BITSTREAM_READER
1415  s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8;
1416 #endif
1417  s->last_buf_size <<= 3;
1418  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1419  for (ch = 0; ch < s->nb_channels; ch++) {
1420  g = &s->granules[ch][gr];
1421  s->last_buf_size += g->part2_3_length;
1422  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1423  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1424  }
1425  }
1426  skip = s->last_buf_size - 8 * main_data_begin;
1427  if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
1428  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
1429  s->gb = s->in_gb;
1430  s->in_gb.buffer = NULL;
1431  } else {
1432  skip_bits_long(&s->gb, skip);
1433  }
1434  } else {
1435  gr = 0;
1436  }
1437 
1438  for (; gr < nb_granules; gr++) {
1439  for (ch = 0; ch < s->nb_channels; ch++) {
1440  g = &s->granules[ch][gr];
1441  bits_pos = get_bits_count(&s->gb);
1442 
1443  if (!s->lsf) {
1444  uint8_t *sc;
1445  int slen, slen1, slen2;
1446 
1447  /* MPEG1 scale factors */
1448  slen1 = slen_table[0][g->scalefac_compress];
1449  slen2 = slen_table[1][g->scalefac_compress];
1450  av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1451  if (g->block_type == 2) {
1452  n = g->switch_point ? 17 : 18;
1453  j = 0;
1454  if (slen1) {
1455  for (i = 0; i < n; i++)
1456  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1457  } else {
1458  for (i = 0; i < n; i++)
1459  g->scale_factors[j++] = 0;
1460  }
1461  if (slen2) {
1462  for (i = 0; i < 18; i++)
1463  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1464  for (i = 0; i < 3; i++)
1465  g->scale_factors[j++] = 0;
1466  } else {
1467  for (i = 0; i < 21; i++)
1468  g->scale_factors[j++] = 0;
1469  }
1470  } else {
1471  sc = s->granules[ch][0].scale_factors;
1472  j = 0;
1473  for (k = 0; k < 4; k++) {
1474  n = k == 0 ? 6 : 5;
1475  if ((g->scfsi & (0x8 >> k)) == 0) {
1476  slen = (k < 2) ? slen1 : slen2;
1477  if (slen) {
1478  for (i = 0; i < n; i++)
1479  g->scale_factors[j++] = get_bits(&s->gb, slen);
1480  } else {
1481  for (i = 0; i < n; i++)
1482  g->scale_factors[j++] = 0;
1483  }
1484  } else {
1485  /* simply copy from last granule */
1486  for (i = 0; i < n; i++) {
1487  g->scale_factors[j] = sc[j];
1488  j++;
1489  }
1490  }
1491  }
1492  g->scale_factors[j++] = 0;
1493  }
1494  } else {
1495  int tindex, tindex2, slen[4], sl, sf;
1496 
1497  /* LSF scale factors */
1498  if (g->block_type == 2)
1499  tindex = g->switch_point ? 2 : 1;
1500  else
1501  tindex = 0;
1502 
1503  sf = g->scalefac_compress;
1504  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1505  /* intensity stereo case */
1506  sf >>= 1;
1507  if (sf < 180) {
1508  lsf_sf_expand(slen, sf, 6, 6, 0);
1509  tindex2 = 3;
1510  } else if (sf < 244) {
1511  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1512  tindex2 = 4;
1513  } else {
1514  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1515  tindex2 = 5;
1516  }
1517  } else {
1518  /* normal case */
1519  if (sf < 400) {
1520  lsf_sf_expand(slen, sf, 5, 4, 4);
1521  tindex2 = 0;
1522  } else if (sf < 500) {
1523  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1524  tindex2 = 1;
1525  } else {
1526  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1527  tindex2 = 2;
1528  g->preflag = 1;
1529  }
1530  }
1531 
1532  j = 0;
1533  for (k = 0; k < 4; k++) {
1534  n = lsf_nsf_table[tindex2][tindex][k];
1535  sl = slen[k];
1536  if (sl) {
1537  for (i = 0; i < n; i++)
1538  g->scale_factors[j++] = get_bits(&s->gb, sl);
1539  } else {
1540  for (i = 0; i < n; i++)
1541  g->scale_factors[j++] = 0;
1542  }
1543  }
1544  /* XXX: should compute exact size */
1545  for (; j < 40; j++)
1546  g->scale_factors[j] = 0;
1547  }
1548 
1549  exponents_from_scale_factors(s, g, exponents);
1550 
1551  /* read Huffman coded residue */
1552  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1553  } /* ch */
1554 
1555  if (s->mode == MPA_JSTEREO)
1556  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1557 
1558  for (ch = 0; ch < s->nb_channels; ch++) {
1559  g = &s->granules[ch][gr];
1560 
1561  reorder_block(s, g);
1562  compute_antialias(s, g);
1563  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1564  }
1565  } /* gr */
1566  if (get_bits_count(&s->gb) < 0)
1567  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1568  return nb_granules * 18;
1569 }
1570 
1572  const uint8_t *buf, int buf_size)
1573 {
1574  int i, nb_frames, ch, ret;
1575  OUT_INT *samples_ptr;
1576 
1577  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1578 
1579  /* skip error protection field */
1580  if (s->error_protection)
1581  skip_bits(&s->gb, 16);
1582 
1583  switch(s->layer) {
1584  case 1:
1585  s->avctx->frame_size = 384;
1586  nb_frames = mp_decode_layer1(s);
1587  break;
1588  case 2:
1589  s->avctx->frame_size = 1152;
1590  nb_frames = mp_decode_layer2(s);
1591  break;
1592  case 3:
1593  s->avctx->frame_size = s->lsf ? 576 : 1152;
1594  default:
1595  nb_frames = mp_decode_layer3(s);
1596 
1597  s->last_buf_size=0;
1598  if (s->in_gb.buffer) {
1599  align_get_bits(&s->gb);
1600  i = get_bits_left(&s->gb)>>3;
1601  if (i >= 0 && i <= BACKSTEP_SIZE) {
1602  memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1603  s->last_buf_size=i;
1604  } else
1605  av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1606  s->gb = s->in_gb;
1607  s->in_gb.buffer = NULL;
1608  }
1609 
1610  align_get_bits(&s->gb);
1611  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1612  i = get_bits_left(&s->gb) >> 3;
1613 
1614  if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1615  if (i < 0)
1616  av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1617  i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1618  }
1619  av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
1620  memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1621  s->last_buf_size += i;
1622  }
1623 
1624  if(nb_frames < 0)
1625  return nb_frames;
1626 
1627  /* get output buffer */
1628  if (!samples) {
1629  av_assert0(s->frame != NULL);
1630  s->frame->nb_samples = s->avctx->frame_size;
1631  if ((ret = ff_get_buffer(s->avctx, s->frame)) < 0) {
1632  av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1633  return ret;
1634  }
1635  samples = (OUT_INT **)s->frame->extended_data;
1636  }
1637 
1638  /* apply the synthesis filter */
1639  for (ch = 0; ch < s->nb_channels; ch++) {
1640  int sample_stride;
1641  if (s->avctx->sample_fmt == OUT_FMT_P) {
1642  samples_ptr = samples[ch];
1643  sample_stride = 1;
1644  } else {
1645  samples_ptr = samples[0] + ch;
1646  sample_stride = s->nb_channels;
1647  }
1648  for (i = 0; i < nb_frames; i++) {
1649  RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
1650  &(s->synth_buf_offset[ch]),
1651  RENAME(ff_mpa_synth_window),
1652  &s->dither_state, samples_ptr,
1653  sample_stride, s->sb_samples[ch][i]);
1654  samples_ptr += 32 * sample_stride;
1655  }
1656  }
1657 
1658  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1659 }
1660 
1661 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1662  AVPacket *avpkt)
1663 {
1664  const uint8_t *buf = avpkt->data;
1665  int buf_size = avpkt->size;
1666  MPADecodeContext *s = avctx->priv_data;
1667  uint32_t header;
1668  int ret;
1669 
1670  while(buf_size && !*buf){
1671  buf++;
1672  buf_size--;
1673  }
1674 
1675  if (buf_size < HEADER_SIZE)
1676  return AVERROR_INVALIDDATA;
1677 
1678  header = AV_RB32(buf);
1679  if (header>>8 == AV_RB32("TAG")>>8) {
1680  av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
1681  return buf_size;
1682  }
1683  if (ff_mpa_check_header(header) < 0) {
1684  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1685  return AVERROR_INVALIDDATA;
1686  }
1687 
1688  if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1689  /* free format: prepare to compute frame size */
1690  s->frame_size = -1;
1691  return AVERROR_INVALIDDATA;
1692  }
1693  /* update codec info */
1694  avctx->channels = s->nb_channels;
1695  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1696  if (!avctx->bit_rate)
1697  avctx->bit_rate = s->bit_rate;
1698 
1699  if (s->frame_size <= 0 || s->frame_size > buf_size) {
1700  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1701  return AVERROR_INVALIDDATA;
1702  } else if (s->frame_size < buf_size) {
1703  av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
1704  buf_size= s->frame_size;
1705  }
1706 
1707  s->frame = data;
1708 
1709  ret = mp_decode_frame(s, NULL, buf, buf_size);
1710  if (ret >= 0) {
1711  s->frame->nb_samples = avctx->frame_size;
1712  *got_frame_ptr = 1;
1713  avctx->sample_rate = s->sample_rate;
1714  //FIXME maybe move the other codec info stuff from above here too
1715  } else {
1716  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1717  /* Only return an error if the bad frame makes up the whole packet or
1718  * the error is related to buffer management.
1719  * If there is more data in the packet, just consume the bad frame
1720  * instead of returning an error, which would discard the whole
1721  * packet. */
1722  *got_frame_ptr = 0;
1723  if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1724  return ret;
1725  }
1726  s->frame_size = 0;
1727  return buf_size;
1728 }
1729 
1730 static void mp_flush(MPADecodeContext *ctx)
1731 {
1732  memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1733  ctx->last_buf_size = 0;
1734 }
1735 
1736 static void flush(AVCodecContext *avctx)
1737 {
1738  mp_flush(avctx->priv_data);
1739 }
1740 
1741 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1742 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1743  int *got_frame_ptr, AVPacket *avpkt)
1744 {
1745  const uint8_t *buf = avpkt->data;
1746  int buf_size = avpkt->size;
1747  MPADecodeContext *s = avctx->priv_data;
1748  uint32_t header;
1749  int len, ret;
1750  int av_unused out_size;
1751 
1752  len = buf_size;
1753 
1754  // Discard too short frames
1755  if (buf_size < HEADER_SIZE) {
1756  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1757  return AVERROR_INVALIDDATA;
1758  }
1759 
1760 
1761  if (len > MPA_MAX_CODED_FRAME_SIZE)
1763 
1764  // Get header and restore sync word
1765  header = AV_RB32(buf) | 0xffe00000;
1766 
1767  if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
1768  av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1769  return AVERROR_INVALIDDATA;
1770  }
1771 
1773  /* update codec info */
1774  avctx->sample_rate = s->sample_rate;
1775  avctx->channels = s->nb_channels;
1776  if (!avctx->bit_rate)
1777  avctx->bit_rate = s->bit_rate;
1778 
1779  s->frame_size = len;
1780 
1781  s->frame = data;
1782 
1783  ret = mp_decode_frame(s, NULL, buf, buf_size);
1784  if (ret < 0) {
1785  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1786  return ret;
1787  }
1788 
1789  *got_frame_ptr = 1;
1790 
1791  return buf_size;
1792 }
1793 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1794 
1795 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1796 
1797 /**
1798  * Context for MP3On4 decoder
1799  */
1800 typedef struct MP3On4DecodeContext {
1801  int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
1802  int syncword; ///< syncword patch
1803  const uint8_t *coff; ///< channel offsets in output buffer
1804  MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
1805 } MP3On4DecodeContext;
1806 
1807 #include "mpeg4audio.h"
1808 
1809 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1810 
1811 /* number of mp3 decoder instances */
1812 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1813 
1814 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1815 static const uint8_t chan_offset[8][5] = {
1816  { 0 },
1817  { 0 }, // C
1818  { 0 }, // FLR
1819  { 2, 0 }, // C FLR
1820  { 2, 0, 3 }, // C FLR BS
1821  { 2, 0, 3 }, // C FLR BLRS
1822  { 2, 0, 4, 3 }, // C FLR BLRS LFE
1823  { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1824 };
1825 
1826 /* mp3on4 channel layouts */
1827 static const int16_t chan_layout[8] = {
1828  0,
1836 };
1837 
1838 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1839 {
1840  MP3On4DecodeContext *s = avctx->priv_data;
1841  int i;
1842 
1843  for (i = 0; i < s->frames; i++)
1844  av_free(s->mp3decctx[i]);
1845 
1846  return 0;
1847 }
1848 
1849 
1850 static int decode_init_mp3on4(AVCodecContext * avctx)
1851 {
1852  MP3On4DecodeContext *s = avctx->priv_data;
1853  MPEG4AudioConfig cfg;
1854  int i;
1855 
1856  if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
1857  av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1858  return AVERROR_INVALIDDATA;
1859  }
1860 
1862  avctx->extradata_size * 8, 1);
1863  if (!cfg.chan_config || cfg.chan_config > 7) {
1864  av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1865  return AVERROR_INVALIDDATA;
1866  }
1867  s->frames = mp3Frames[cfg.chan_config];
1868  s->coff = chan_offset[cfg.chan_config];
1870  avctx->channel_layout = chan_layout[cfg.chan_config];
1871 
1872  if (cfg.sample_rate < 16000)
1873  s->syncword = 0xffe00000;
1874  else
1875  s->syncword = 0xfff00000;
1876 
1877  /* Init the first mp3 decoder in standard way, so that all tables get builded
1878  * We replace avctx->priv_data with the context of the first decoder so that
1879  * decode_init() does not have to be changed.
1880  * Other decoders will be initialized here copying data from the first context
1881  */
1882  // Allocate zeroed memory for the first decoder context
1883  s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1884  if (!s->mp3decctx[0])
1885  goto alloc_fail;
1886  // Put decoder context in place to make init_decode() happy
1887  avctx->priv_data = s->mp3decctx[0];
1888  decode_init(avctx);
1889  // Restore mp3on4 context pointer
1890  avctx->priv_data = s;
1891  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1892 
1893  /* Create a separate codec/context for each frame (first is already ok).
1894  * Each frame is 1 or 2 channels - up to 5 frames allowed
1895  */
1896  for (i = 1; i < s->frames; i++) {
1897  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1898  if (!s->mp3decctx[i])
1899  goto alloc_fail;
1900  s->mp3decctx[i]->adu_mode = 1;
1901  s->mp3decctx[i]->avctx = avctx;
1902  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1903  }
1904 
1905  return 0;
1906 alloc_fail:
1907  decode_close_mp3on4(avctx);
1908  return AVERROR(ENOMEM);
1909 }
1910 
1911 
1912 static void flush_mp3on4(AVCodecContext *avctx)
1913 {
1914  int i;
1915  MP3On4DecodeContext *s = avctx->priv_data;
1916 
1917  for (i = 0; i < s->frames; i++)
1918  mp_flush(s->mp3decctx[i]);
1919 }
1920 
1921 
1922 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1923  int *got_frame_ptr, AVPacket *avpkt)
1924 {
1925  AVFrame *frame = data;
1926  const uint8_t *buf = avpkt->data;
1927  int buf_size = avpkt->size;
1928  MP3On4DecodeContext *s = avctx->priv_data;
1930  int fsize, len = buf_size, out_size = 0;
1931  uint32_t header;
1932  OUT_INT **out_samples;
1933  OUT_INT *outptr[2];
1934  int fr, ch, ret;
1935 
1936  /* get output buffer */
1937  frame->nb_samples = MPA_FRAME_SIZE;
1938  if ((ret = ff_get_buffer(avctx, frame)) < 0) {
1939  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1940  return ret;
1941  }
1942  out_samples = (OUT_INT **)frame->extended_data;
1943 
1944  // Discard too short frames
1945  if (buf_size < HEADER_SIZE)
1946  return AVERROR_INVALIDDATA;
1947 
1948  avctx->bit_rate = 0;
1949 
1950  ch = 0;
1951  for (fr = 0; fr < s->frames; fr++) {
1952  fsize = AV_RB16(buf) >> 4;
1953  fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1954  m = s->mp3decctx[fr];
1955  av_assert1(m);
1956 
1957  if (fsize < HEADER_SIZE) {
1958  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1959  return AVERROR_INVALIDDATA;
1960  }
1961  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1962 
1963  if (ff_mpa_check_header(header) < 0) // Bad header, discard block
1964  break;
1965 
1967 
1968  if (ch + m->nb_channels > avctx->channels || s->coff[fr] + m->nb_channels > avctx->channels) {
1969  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1970  "channel count\n");
1971  return AVERROR_INVALIDDATA;
1972  }
1973  ch += m->nb_channels;
1974 
1975  outptr[0] = out_samples[s->coff[fr]];
1976  if (m->nb_channels > 1)
1977  outptr[1] = out_samples[s->coff[fr] + 1];
1978 
1979  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
1980  return ret;
1981 
1982  out_size += ret;
1983  buf += fsize;
1984  len -= fsize;
1985 
1986  avctx->bit_rate += m->bit_rate;
1987  }
1988 
1989  /* update codec info */
1990  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1991 
1992  frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1993  *got_frame_ptr = 1;
1994 
1995  return buf_size;
1996 }
1997 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
1998 
1999 #if !CONFIG_FLOAT
2000 #if CONFIG_MP1_DECODER
2001 AVCodec ff_mp1_decoder = {
2002  .name = "mp1",
2003  .type = AVMEDIA_TYPE_AUDIO,
2004  .id = AV_CODEC_ID_MP1,
2005  .priv_data_size = sizeof(MPADecodeContext),
2006  .init = decode_init,
2007  .decode = decode_frame,
2008  .capabilities = CODEC_CAP_DR1,
2009  .flush = flush,
2010  .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
2011  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2014 };
2015 #endif
2016 #if CONFIG_MP2_DECODER
2017 AVCodec ff_mp2_decoder = {
2018  .name = "mp2",
2019  .type = AVMEDIA_TYPE_AUDIO,
2020  .id = AV_CODEC_ID_MP2,
2021  .priv_data_size = sizeof(MPADecodeContext),
2022  .init = decode_init,
2023  .decode = decode_frame,
2024  .capabilities = CODEC_CAP_DR1,
2025  .flush = flush,
2026  .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2027  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2030 };
2031 #endif
2032 #if CONFIG_MP3_DECODER
2033 AVCodec ff_mp3_decoder = {
2034  .name = "mp3",
2035  .type = AVMEDIA_TYPE_AUDIO,
2036  .id = AV_CODEC_ID_MP3,
2037  .priv_data_size = sizeof(MPADecodeContext),
2038  .init = decode_init,
2039  .decode = decode_frame,
2040  .capabilities = CODEC_CAP_DR1,
2041  .flush = flush,
2042  .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2043  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2046 };
2047 #endif
2048 #if CONFIG_MP3ADU_DECODER
2049 AVCodec ff_mp3adu_decoder = {
2050  .name = "mp3adu",
2051  .type = AVMEDIA_TYPE_AUDIO,
2052  .id = AV_CODEC_ID_MP3ADU,
2053  .priv_data_size = sizeof(MPADecodeContext),
2054  .init = decode_init,
2055  .decode = decode_frame_adu,
2056  .capabilities = CODEC_CAP_DR1,
2057  .flush = flush,
2058  .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2059  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2062 };
2063 #endif
2064 #if CONFIG_MP3ON4_DECODER
2065 AVCodec ff_mp3on4_decoder = {
2066  .name = "mp3on4",
2067  .type = AVMEDIA_TYPE_AUDIO,
2068  .id = AV_CODEC_ID_MP3ON4,
2069  .priv_data_size = sizeof(MP3On4DecodeContext),
2070  .init = decode_init_mp3on4,
2071  .close = decode_close_mp3on4,
2072  .decode = decode_frame_mp3on4,
2073  .capabilities = CODEC_CAP_DR1,
2074  .flush = flush_mp3on4,
2075  .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
2076  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2078 };
2079 #endif
2080 #endif