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mpegaudiodec_template.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/attributes.h"
28 #include "libavutil/avassert.h"
30 #include "libavutil/float_dsp.h"
31 #include "libavutil/libm.h"
32 #include "avcodec.h"
33 #include "get_bits.h"
34 #include "internal.h"
35 #include "mathops.h"
36 #include "mpegaudiodsp.h"
37 
38 /*
39  * TODO:
40  * - test lsf / mpeg25 extensively.
41  */
42 
43 #include "mpegaudio.h"
44 #include "mpegaudiodecheader.h"
45 
46 #define BACKSTEP_SIZE 512
47 #define EXTRABYTES 24
48 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
49 
50 /* layer 3 "granule" */
51 typedef struct GranuleDef {
59  int table_select[3];
60  int subblock_gain[3];
63  int region_size[3]; /* number of huffman codes in each region */
64  int preflag;
65  int short_start, long_end; /* long/short band indexes */
67  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
68 } GranuleDef;
69 
70 typedef struct MPADecodeContext {
74  int extrasize;
75  /* next header (used in free format parsing) */
82  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
83  GranuleDef granules[2][2]; /* Used in Layer 3 */
84  int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
92 
93 #define HEADER_SIZE 4
94 
95 #include "mpegaudiodata.h"
96 #include "mpegaudiodectab.h"
97 
98 /* vlc structure for decoding layer 3 huffman tables */
99 static VLC huff_vlc[16];
101  0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
102  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
103  ][2];
104 static const int huff_vlc_tables_sizes[16] = {
105  0, 128, 128, 128, 130, 128, 154, 166,
106  142, 204, 190, 170, 542, 460, 662, 414
107 };
108 static VLC huff_quad_vlc[2];
109 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
110 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
111 /* computed from band_size_long */
112 static uint16_t band_index_long[9][23];
113 #include "mpegaudio_tablegen.h"
114 /* intensity stereo coef table */
115 static INTFLOAT is_table[2][16];
116 static INTFLOAT is_table_lsf[2][2][16];
117 static INTFLOAT csa_table[8][4];
118 
119 static int16_t division_tab3[1<<6 ];
120 static int16_t division_tab5[1<<8 ];
121 static int16_t division_tab9[1<<11];
122 
123 static int16_t * const division_tabs[4] = {
125 };
126 
127 /* lower 2 bits: modulo 3, higher bits: shift */
128 static uint16_t scale_factor_modshift[64];
129 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
131 /* mult table for layer 2 group quantization */
132 
133 #define SCALE_GEN(v) \
134 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
135 
136 static const int32_t scale_factor_mult2[3][3] = {
137  SCALE_GEN(4.0 / 3.0), /* 3 steps */
138  SCALE_GEN(4.0 / 5.0), /* 5 steps */
139  SCALE_GEN(4.0 / 9.0), /* 9 steps */
140 };
141 
142 /**
143  * Convert region offsets to region sizes and truncate
144  * size to big_values.
145  */
147 {
148  int i, k, j = 0;
149  g->region_size[2] = 576 / 2;
150  for (i = 0; i < 3; i++) {
151  k = FFMIN(g->region_size[i], g->big_values);
152  g->region_size[i] = k - j;
153  j = k;
154  }
155 }
156 
158 {
159  if (g->block_type == 2) {
160  if (s->sample_rate_index != 8)
161  g->region_size[0] = (36 / 2);
162  else
163  g->region_size[0] = (72 / 2);
164  } else {
165  if (s->sample_rate_index <= 2)
166  g->region_size[0] = (36 / 2);
167  else if (s->sample_rate_index != 8)
168  g->region_size[0] = (54 / 2);
169  else
170  g->region_size[0] = (108 / 2);
171  }
172  g->region_size[1] = (576 / 2);
173 }
174 
176  int ra1, int ra2)
177 {
178  int l;
179  g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
180  /* should not overflow */
181  l = FFMIN(ra1 + ra2 + 2, 22);
182  g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
183 }
184 
186 {
187  if (g->block_type == 2) {
188  if (g->switch_point) {
189  if(s->sample_rate_index == 8)
190  avpriv_request_sample(s->avctx, "switch point in 8khz");
191  /* if switched mode, we handle the 36 first samples as
192  long blocks. For 8000Hz, we handle the 72 first
193  exponents as long blocks */
194  if (s->sample_rate_index <= 2)
195  g->long_end = 8;
196  else
197  g->long_end = 6;
198 
199  g->short_start = 3;
200  } else {
201  g->long_end = 0;
202  g->short_start = 0;
203  }
204  } else {
205  g->short_start = 13;
206  g->long_end = 22;
207  }
208 }
209 
210 /* layer 1 unscaling */
211 /* n = number of bits of the mantissa minus 1 */
212 static inline int l1_unscale(int n, int mant, int scale_factor)
213 {
214  int shift, mod;
215  int64_t val;
216 
217  shift = scale_factor_modshift[scale_factor];
218  mod = shift & 3;
219  shift >>= 2;
220  val = MUL64((int)(mant + (-1U << n) + 1), scale_factor_mult[n-1][mod]);
221  shift += n;
222  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
223  return (int)((val + (1LL << (shift - 1))) >> shift);
224 }
225 
226 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
227 {
228  int shift, mod, val;
229 
230  shift = scale_factor_modshift[scale_factor];
231  mod = shift & 3;
232  shift >>= 2;
233 
234  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
235  /* NOTE: at this point, 0 <= shift <= 21 */
236  if (shift > 0)
237  val = (val + (1 << (shift - 1))) >> shift;
238  return val;
239 }
240 
241 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
242 static inline int l3_unscale(int value, int exponent)
243 {
244  unsigned int m;
245  int e;
246 
247  e = table_4_3_exp [4 * value + (exponent & 3)];
248  m = table_4_3_value[4 * value + (exponent & 3)];
249  e -= exponent >> 2;
250 #ifdef DEBUG
251  if(e < 1)
252  av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
253 #endif
254  if (e > (SUINT)31)
255  return 0;
256  m = (m + ((1U << e)>>1)) >> e;
257 
258  return m;
259 }
260 
261 static av_cold void decode_init_static(void)
262 {
263  int i, j, k;
264  int offset;
265 
266  /* scale factors table for layer 1/2 */
267  for (i = 0; i < 64; i++) {
268  int shift, mod;
269  /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
270  shift = i / 3;
271  mod = i % 3;
272  scale_factor_modshift[i] = mod | (shift << 2);
273  }
274 
275  /* scale factor multiply for layer 1 */
276  for (i = 0; i < 15; i++) {
277  int n, norm;
278  n = i + 2;
279  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
280  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
281  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
282  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
283  ff_dlog(NULL, "%d: norm=%x s=%"PRIx32" %"PRIx32" %"PRIx32"\n", i,
284  (unsigned)norm,
285  scale_factor_mult[i][0],
286  scale_factor_mult[i][1],
287  scale_factor_mult[i][2]);
288  }
289 
290  RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
291 
292  /* huffman decode tables */
293  offset = 0;
294  for (i = 1; i < 16; i++) {
295  const HuffTable *h = &mpa_huff_tables[i];
296  int xsize, x, y;
297  uint8_t tmp_bits [512] = { 0 };
298  uint16_t tmp_codes[512] = { 0 };
299 
300  xsize = h->xsize;
301 
302  j = 0;
303  for (x = 0; x < xsize; x++) {
304  for (y = 0; y < xsize; y++) {
305  tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
306  tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
307  }
308  }
309 
310  /* XXX: fail test */
311  huff_vlc[i].table = huff_vlc_tables+offset;
312  huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
313  init_vlc(&huff_vlc[i], 7, 512,
314  tmp_bits, 1, 1, tmp_codes, 2, 2,
316  offset += huff_vlc_tables_sizes[i];
317  }
319 
320  offset = 0;
321  for (i = 0; i < 2; i++) {
322  huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
323  huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
324  init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
325  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
327  offset += huff_quad_vlc_tables_sizes[i];
328  }
330 
331  for (i = 0; i < 9; i++) {
332  k = 0;
333  for (j = 0; j < 22; j++) {
334  band_index_long[i][j] = k;
335  k += band_size_long[i][j];
336  }
337  band_index_long[i][22] = k;
338  }
339 
340  /* compute n ^ (4/3) and store it in mantissa/exp format */
341 
343 
344  for (i = 0; i < 4; i++) {
345  if (ff_mpa_quant_bits[i] < 0) {
346  for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
347  int val1, val2, val3, steps;
348  int val = j;
349  steps = ff_mpa_quant_steps[i];
350  val1 = val % steps;
351  val /= steps;
352  val2 = val % steps;
353  val3 = val / steps;
354  division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
355  }
356  }
357  }
358 
359 
360  for (i = 0; i < 7; i++) {
361  float f;
362  INTFLOAT v;
363  if (i != 6) {
364  f = tan((double)i * M_PI / 12.0);
365  v = FIXR(f / (1.0 + f));
366  } else {
367  v = FIXR(1.0);
368  }
369  is_table[0][ i] = v;
370  is_table[1][6 - i] = v;
371  }
372  /* invalid values */
373  for (i = 7; i < 16; i++)
374  is_table[0][i] = is_table[1][i] = 0.0;
375 
376  for (i = 0; i < 16; i++) {
377  double f;
378  int e, k;
379 
380  for (j = 0; j < 2; j++) {
381  e = -(j + 1) * ((i + 1) >> 1);
382  f = exp2(e / 4.0);
383  k = i & 1;
384  is_table_lsf[j][k ^ 1][i] = FIXR(f);
385  is_table_lsf[j][k ][i] = FIXR(1.0);
386  ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
387  i, j, (float) is_table_lsf[j][0][i],
388  (float) is_table_lsf[j][1][i]);
389  }
390  }
391 
392  for (i = 0; i < 8; i++) {
393  double ci, cs, ca;
394  ci = ci_table[i];
395  cs = 1.0 / sqrt(1.0 + ci * ci);
396  ca = cs * ci;
397 #if !USE_FLOATS
398  csa_table[i][0] = FIXHR(cs/4);
399  csa_table[i][1] = FIXHR(ca/4);
400  csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
401  csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
402 #else
403  csa_table[i][0] = cs;
404  csa_table[i][1] = ca;
405  csa_table[i][2] = ca + cs;
406  csa_table[i][3] = ca - cs;
407 #endif
408  }
409 }
410 
411 #if USE_FLOATS
412 static av_cold int decode_close(AVCodecContext * avctx)
413 {
414  MPADecodeContext *s = avctx->priv_data;
415  av_freep(&s->fdsp);
416 
417  return 0;
418 }
419 #endif
420 
421 static av_cold int decode_init(AVCodecContext * avctx)
422 {
423  static int initialized_tables = 0;
424  MPADecodeContext *s = avctx->priv_data;
425 
426  if (!initialized_tables) {
428  initialized_tables = 1;
429  }
430 
431  s->avctx = avctx;
432 
433 #if USE_FLOATS
435  if (!s->fdsp)
436  return AVERROR(ENOMEM);
437 #endif
438 
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 
451  return 0;
452 }
453 
454 #define C3 FIXHR(0.86602540378443864676/2)
455 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
456 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
457 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
458 
459 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
460  cases. */
462 {
463  SUINTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
464 
465  in0 = in[0*3];
466  in1 = in[1*3] + in[0*3];
467  in2 = in[2*3] + in[1*3];
468  in3 = in[3*3] + in[2*3];
469  in4 = in[4*3] + in[3*3];
470  in5 = in[5*3] + in[4*3];
471  in5 += in3;
472  in3 += in1;
473 
474  in2 = MULH3(in2, C3, 2);
475  in3 = MULH3(in3, C3, 4);
476 
477  t1 = in0 - in4;
478  t2 = MULH3(in1 - in5, C4, 2);
479 
480  out[ 7] =
481  out[10] = t1 + t2;
482  out[ 1] =
483  out[ 4] = t1 - t2;
484 
485  in0 += SHR(in4, 1);
486  in4 = in0 + in2;
487  in5 += 2*in1;
488  in1 = MULH3(in5 + in3, C5, 1);
489  out[ 8] =
490  out[ 9] = in4 + in1;
491  out[ 2] =
492  out[ 3] = in4 - in1;
493 
494  in0 -= in2;
495  in5 = MULH3(in5 - in3, C6, 2);
496  out[ 0] =
497  out[ 5] = in0 - in5;
498  out[ 6] =
499  out[11] = in0 + in5;
500 }
501 
502 /* return the number of decoded frames */
504 {
505  int bound, i, v, n, ch, j, mant;
506  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
507  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
508 
509  if (s->mode == MPA_JSTEREO)
510  bound = (s->mode_ext + 1) * 4;
511  else
512  bound = SBLIMIT;
513 
514  /* allocation bits */
515  for (i = 0; i < bound; i++) {
516  for (ch = 0; ch < s->nb_channels; ch++) {
517  allocation[ch][i] = get_bits(&s->gb, 4);
518  }
519  }
520  for (i = bound; i < SBLIMIT; i++)
521  allocation[0][i] = get_bits(&s->gb, 4);
522 
523  /* scale factors */
524  for (i = 0; i < bound; i++) {
525  for (ch = 0; ch < s->nb_channels; ch++) {
526  if (allocation[ch][i])
527  scale_factors[ch][i] = get_bits(&s->gb, 6);
528  }
529  }
530  for (i = bound; i < SBLIMIT; i++) {
531  if (allocation[0][i]) {
532  scale_factors[0][i] = get_bits(&s->gb, 6);
533  scale_factors[1][i] = get_bits(&s->gb, 6);
534  }
535  }
536 
537  /* compute samples */
538  for (j = 0; j < 12; j++) {
539  for (i = 0; i < bound; i++) {
540  for (ch = 0; ch < s->nb_channels; ch++) {
541  n = allocation[ch][i];
542  if (n) {
543  mant = get_bits(&s->gb, n + 1);
544  v = l1_unscale(n, mant, scale_factors[ch][i]);
545  } else {
546  v = 0;
547  }
548  s->sb_samples[ch][j][i] = v;
549  }
550  }
551  for (i = bound; i < SBLIMIT; i++) {
552  n = allocation[0][i];
553  if (n) {
554  mant = get_bits(&s->gb, n + 1);
555  v = l1_unscale(n, mant, scale_factors[0][i]);
556  s->sb_samples[0][j][i] = v;
557  v = l1_unscale(n, mant, scale_factors[1][i]);
558  s->sb_samples[1][j][i] = v;
559  } else {
560  s->sb_samples[0][j][i] = 0;
561  s->sb_samples[1][j][i] = 0;
562  }
563  }
564  }
565  return 12;
566 }
567 
569 {
570  int sblimit; /* number of used subbands */
571  const unsigned char *alloc_table;
572  int table, bit_alloc_bits, i, j, ch, bound, v;
573  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
574  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
575  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
576  int scale, qindex, bits, steps, k, l, m, b;
577 
578  /* select decoding table */
579  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
580  s->sample_rate, s->lsf);
581  sblimit = ff_mpa_sblimit_table[table];
582  alloc_table = ff_mpa_alloc_tables[table];
583 
584  if (s->mode == MPA_JSTEREO)
585  bound = (s->mode_ext + 1) * 4;
586  else
587  bound = sblimit;
588 
589  ff_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
590 
591  /* sanity check */
592  if (bound > sblimit)
593  bound = sblimit;
594 
595  /* parse bit allocation */
596  j = 0;
597  for (i = 0; i < bound; i++) {
598  bit_alloc_bits = alloc_table[j];
599  for (ch = 0; ch < s->nb_channels; ch++)
600  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
601  j += 1 << bit_alloc_bits;
602  }
603  for (i = bound; i < sblimit; i++) {
604  bit_alloc_bits = alloc_table[j];
605  v = get_bits(&s->gb, bit_alloc_bits);
606  bit_alloc[0][i] = v;
607  bit_alloc[1][i] = v;
608  j += 1 << bit_alloc_bits;
609  }
610 
611  /* scale codes */
612  for (i = 0; i < sblimit; i++) {
613  for (ch = 0; ch < s->nb_channels; ch++) {
614  if (bit_alloc[ch][i])
615  scale_code[ch][i] = get_bits(&s->gb, 2);
616  }
617  }
618 
619  /* scale factors */
620  for (i = 0; i < sblimit; i++) {
621  for (ch = 0; ch < s->nb_channels; ch++) {
622  if (bit_alloc[ch][i]) {
623  sf = scale_factors[ch][i];
624  switch (scale_code[ch][i]) {
625  default:
626  case 0:
627  sf[0] = get_bits(&s->gb, 6);
628  sf[1] = get_bits(&s->gb, 6);
629  sf[2] = get_bits(&s->gb, 6);
630  break;
631  case 2:
632  sf[0] = get_bits(&s->gb, 6);
633  sf[1] = sf[0];
634  sf[2] = sf[0];
635  break;
636  case 1:
637  sf[0] = get_bits(&s->gb, 6);
638  sf[2] = get_bits(&s->gb, 6);
639  sf[1] = sf[0];
640  break;
641  case 3:
642  sf[0] = get_bits(&s->gb, 6);
643  sf[2] = get_bits(&s->gb, 6);
644  sf[1] = sf[2];
645  break;
646  }
647  }
648  }
649  }
650 
651  /* samples */
652  for (k = 0; k < 3; k++) {
653  for (l = 0; l < 12; l += 3) {
654  j = 0;
655  for (i = 0; i < bound; i++) {
656  bit_alloc_bits = alloc_table[j];
657  for (ch = 0; ch < s->nb_channels; ch++) {
658  b = bit_alloc[ch][i];
659  if (b) {
660  scale = scale_factors[ch][i][k];
661  qindex = alloc_table[j+b];
662  bits = ff_mpa_quant_bits[qindex];
663  if (bits < 0) {
664  int v2;
665  /* 3 values at the same time */
666  v = get_bits(&s->gb, -bits);
667  v2 = division_tabs[qindex][v];
668  steps = ff_mpa_quant_steps[qindex];
669 
670  s->sb_samples[ch][k * 12 + l + 0][i] =
671  l2_unscale_group(steps, v2 & 15, scale);
672  s->sb_samples[ch][k * 12 + l + 1][i] =
673  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
674  s->sb_samples[ch][k * 12 + l + 2][i] =
675  l2_unscale_group(steps, v2 >> 8 , scale);
676  } else {
677  for (m = 0; m < 3; m++) {
678  v = get_bits(&s->gb, bits);
679  v = l1_unscale(bits - 1, v, scale);
680  s->sb_samples[ch][k * 12 + l + m][i] = v;
681  }
682  }
683  } else {
684  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
685  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
686  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
687  }
688  }
689  /* next subband in alloc table */
690  j += 1 << bit_alloc_bits;
691  }
692  /* XXX: find a way to avoid this duplication of code */
693  for (i = bound; i < sblimit; i++) {
694  bit_alloc_bits = alloc_table[j];
695  b = bit_alloc[0][i];
696  if (b) {
697  int mant, scale0, scale1;
698  scale0 = scale_factors[0][i][k];
699  scale1 = scale_factors[1][i][k];
700  qindex = alloc_table[j+b];
701  bits = ff_mpa_quant_bits[qindex];
702  if (bits < 0) {
703  /* 3 values at the same time */
704  v = get_bits(&s->gb, -bits);
705  steps = ff_mpa_quant_steps[qindex];
706  mant = v % steps;
707  v = v / steps;
708  s->sb_samples[0][k * 12 + l + 0][i] =
709  l2_unscale_group(steps, mant, scale0);
710  s->sb_samples[1][k * 12 + l + 0][i] =
711  l2_unscale_group(steps, mant, scale1);
712  mant = v % steps;
713  v = v / steps;
714  s->sb_samples[0][k * 12 + l + 1][i] =
715  l2_unscale_group(steps, mant, scale0);
716  s->sb_samples[1][k * 12 + l + 1][i] =
717  l2_unscale_group(steps, mant, scale1);
718  s->sb_samples[0][k * 12 + l + 2][i] =
719  l2_unscale_group(steps, v, scale0);
720  s->sb_samples[1][k * 12 + l + 2][i] =
721  l2_unscale_group(steps, v, scale1);
722  } else {
723  for (m = 0; m < 3; m++) {
724  mant = get_bits(&s->gb, bits);
725  s->sb_samples[0][k * 12 + l + m][i] =
726  l1_unscale(bits - 1, mant, scale0);
727  s->sb_samples[1][k * 12 + l + m][i] =
728  l1_unscale(bits - 1, mant, scale1);
729  }
730  }
731  } else {
732  s->sb_samples[0][k * 12 + l + 0][i] = 0;
733  s->sb_samples[0][k * 12 + l + 1][i] = 0;
734  s->sb_samples[0][k * 12 + l + 2][i] = 0;
735  s->sb_samples[1][k * 12 + l + 0][i] = 0;
736  s->sb_samples[1][k * 12 + l + 1][i] = 0;
737  s->sb_samples[1][k * 12 + l + 2][i] = 0;
738  }
739  /* next subband in alloc table */
740  j += 1 << bit_alloc_bits;
741  }
742  /* fill remaining samples to zero */
743  for (i = sblimit; i < SBLIMIT; i++) {
744  for (ch = 0; ch < s->nb_channels; ch++) {
745  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
746  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
747  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
748  }
749  }
750  }
751  }
752  return 3 * 12;
753 }
754 
755 #define SPLIT(dst,sf,n) \
756  if (n == 3) { \
757  int m = (sf * 171) >> 9; \
758  dst = sf - 3 * m; \
759  sf = m; \
760  } else if (n == 4) { \
761  dst = sf & 3; \
762  sf >>= 2; \
763  } else if (n == 5) { \
764  int m = (sf * 205) >> 10; \
765  dst = sf - 5 * m; \
766  sf = m; \
767  } else if (n == 6) { \
768  int m = (sf * 171) >> 10; \
769  dst = sf - 6 * m; \
770  sf = m; \
771  } else { \
772  dst = 0; \
773  }
774 
775 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
776  int n3)
777 {
778  SPLIT(slen[3], sf, n3)
779  SPLIT(slen[2], sf, n2)
780  SPLIT(slen[1], sf, n1)
781  slen[0] = sf;
782 }
783 
785  int16_t *exponents)
786 {
787  const uint8_t *bstab, *pretab;
788  int len, i, j, k, l, v0, shift, gain, gains[3];
789  int16_t *exp_ptr;
790 
791  exp_ptr = exponents;
792  gain = g->global_gain - 210;
793  shift = g->scalefac_scale + 1;
794 
795  bstab = band_size_long[s->sample_rate_index];
796  pretab = mpa_pretab[g->preflag];
797  for (i = 0; i < g->long_end; i++) {
798  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
799  len = bstab[i];
800  for (j = len; j > 0; j--)
801  *exp_ptr++ = v0;
802  }
803 
804  if (g->short_start < 13) {
805  bstab = band_size_short[s->sample_rate_index];
806  gains[0] = gain - (g->subblock_gain[0] << 3);
807  gains[1] = gain - (g->subblock_gain[1] << 3);
808  gains[2] = gain - (g->subblock_gain[2] << 3);
809  k = g->long_end;
810  for (i = g->short_start; i < 13; i++) {
811  len = bstab[i];
812  for (l = 0; l < 3; l++) {
813  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
814  for (j = len; j > 0; j--)
815  *exp_ptr++ = v0;
816  }
817  }
818  }
819 }
820 
821 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
822  int *end_pos2)
823 {
824  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits - s->extrasize * 8) {
825  s->gb = s->in_gb;
826  s->in_gb.buffer = NULL;
827  s->extrasize = 0;
828  av_assert2((get_bits_count(&s->gb) & 7) == 0);
829  skip_bits_long(&s->gb, *pos - *end_pos);
830  *end_pos2 =
831  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
832  *pos = get_bits_count(&s->gb);
833  }
834 }
835 
836 /* Following is an optimized code for
837  INTFLOAT v = *src
838  if(get_bits1(&s->gb))
839  v = -v;
840  *dst = v;
841 */
842 #if USE_FLOATS
843 #define READ_FLIP_SIGN(dst,src) \
844  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
845  AV_WN32A(dst, v);
846 #else
847 #define READ_FLIP_SIGN(dst,src) \
848  v = -get_bits1(&s->gb); \
849  *(dst) = (*(src) ^ v) - v;
850 #endif
851 
853  int16_t *exponents, int end_pos2)
854 {
855  int s_index;
856  int i;
857  int last_pos, bits_left;
858  VLC *vlc;
859  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits - s->extrasize * 8);
860 
861  /* low frequencies (called big values) */
862  s_index = 0;
863  for (i = 0; i < 3; i++) {
864  int j, k, l, linbits;
865  j = g->region_size[i];
866  if (j == 0)
867  continue;
868  /* select vlc table */
869  k = g->table_select[i];
870  l = mpa_huff_data[k][0];
871  linbits = mpa_huff_data[k][1];
872  vlc = &huff_vlc[l];
873 
874  if (!l) {
875  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
876  s_index += 2 * j;
877  continue;
878  }
879 
880  /* read huffcode and compute each couple */
881  for (; j > 0; j--) {
882  int exponent, x, y;
883  int v;
884  int pos = get_bits_count(&s->gb);
885 
886  if (pos >= end_pos){
887  switch_buffer(s, &pos, &end_pos, &end_pos2);
888  if (pos >= end_pos)
889  break;
890  }
891  y = get_vlc2(&s->gb, vlc->table, 7, 3);
892 
893  if (!y) {
894  g->sb_hybrid[s_index ] =
895  g->sb_hybrid[s_index+1] = 0;
896  s_index += 2;
897  continue;
898  }
899 
900  exponent= exponents[s_index];
901 
902  ff_dlog(s->avctx, "region=%d n=%d y=%d exp=%d\n",
903  i, g->region_size[i] - j, y, exponent);
904  if (y & 16) {
905  x = y >> 5;
906  y = y & 0x0f;
907  if (x < 15) {
908  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
909  } else {
910  x += get_bitsz(&s->gb, linbits);
911  v = l3_unscale(x, exponent);
912  if (get_bits1(&s->gb))
913  v = -v;
914  g->sb_hybrid[s_index] = v;
915  }
916  if (y < 15) {
917  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
918  } else {
919  y += get_bitsz(&s->gb, linbits);
920  v = l3_unscale(y, exponent);
921  if (get_bits1(&s->gb))
922  v = -v;
923  g->sb_hybrid[s_index+1] = v;
924  }
925  } else {
926  x = y >> 5;
927  y = y & 0x0f;
928  x += y;
929  if (x < 15) {
930  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
931  } else {
932  x += get_bitsz(&s->gb, linbits);
933  v = l3_unscale(x, exponent);
934  if (get_bits1(&s->gb))
935  v = -v;
936  g->sb_hybrid[s_index+!!y] = v;
937  }
938  g->sb_hybrid[s_index + !y] = 0;
939  }
940  s_index += 2;
941  }
942  }
943 
944  /* high frequencies */
945  vlc = &huff_quad_vlc[g->count1table_select];
946  last_pos = 0;
947  while (s_index <= 572) {
948  int pos, code;
949  pos = get_bits_count(&s->gb);
950  if (pos >= end_pos) {
951  if (pos > end_pos2 && last_pos) {
952  /* some encoders generate an incorrect size for this
953  part. We must go back into the data */
954  s_index -= 4;
955  skip_bits_long(&s->gb, last_pos - pos);
956  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
958  s_index=0;
959  break;
960  }
961  switch_buffer(s, &pos, &end_pos, &end_pos2);
962  if (pos >= end_pos)
963  break;
964  }
965  last_pos = pos;
966 
967  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
968  ff_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
969  g->sb_hybrid[s_index+0] =
970  g->sb_hybrid[s_index+1] =
971  g->sb_hybrid[s_index+2] =
972  g->sb_hybrid[s_index+3] = 0;
973  while (code) {
974  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
975  int v;
976  int pos = s_index + idxtab[code];
977  code ^= 8 >> idxtab[code];
978  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
979  }
980  s_index += 4;
981  }
982  /* skip extension bits */
983  bits_left = end_pos2 - get_bits_count(&s->gb);
984  if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
985  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
986  s_index=0;
987  } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
988  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
989  s_index = 0;
990  }
991  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
992  skip_bits_long(&s->gb, bits_left);
993 
994  i = get_bits_count(&s->gb);
995  switch_buffer(s, &i, &end_pos, &end_pos2);
996 
997  return 0;
998 }
999 
1000 /* Reorder short blocks from bitstream order to interleaved order. It
1001  would be faster to do it in parsing, but the code would be far more
1002  complicated */
1004 {
1005  int i, j, len;
1006  INTFLOAT *ptr, *dst, *ptr1;
1007  INTFLOAT tmp[576];
1008 
1009  if (g->block_type != 2)
1010  return;
1011 
1012  if (g->switch_point) {
1013  if (s->sample_rate_index != 8)
1014  ptr = g->sb_hybrid + 36;
1015  else
1016  ptr = g->sb_hybrid + 72;
1017  } else {
1018  ptr = g->sb_hybrid;
1019  }
1020 
1021  for (i = g->short_start; i < 13; i++) {
1022  len = band_size_short[s->sample_rate_index][i];
1023  ptr1 = ptr;
1024  dst = tmp;
1025  for (j = len; j > 0; j--) {
1026  *dst++ = ptr[0*len];
1027  *dst++ = ptr[1*len];
1028  *dst++ = ptr[2*len];
1029  ptr++;
1030  }
1031  ptr += 2 * len;
1032  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1033  }
1034 }
1035 
1036 #define ISQRT2 FIXR(0.70710678118654752440)
1037 
1039 {
1040  int i, j, k, l;
1041  int sf_max, sf, len, non_zero_found;
1042  INTFLOAT (*is_tab)[16], *tab0, *tab1, v1, v2;
1043  SUINTFLOAT tmp0, tmp1;
1044  int non_zero_found_short[3];
1045 
1046  /* intensity stereo */
1047  if (s->mode_ext & MODE_EXT_I_STEREO) {
1048  if (!s->lsf) {
1049  is_tab = is_table;
1050  sf_max = 7;
1051  } else {
1052  is_tab = is_table_lsf[g1->scalefac_compress & 1];
1053  sf_max = 16;
1054  }
1055 
1056  tab0 = g0->sb_hybrid + 576;
1057  tab1 = g1->sb_hybrid + 576;
1058 
1059  non_zero_found_short[0] = 0;
1060  non_zero_found_short[1] = 0;
1061  non_zero_found_short[2] = 0;
1062  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1063  for (i = 12; i >= g1->short_start; i--) {
1064  /* for last band, use previous scale factor */
1065  if (i != 11)
1066  k -= 3;
1067  len = band_size_short[s->sample_rate_index][i];
1068  for (l = 2; l >= 0; l--) {
1069  tab0 -= len;
1070  tab1 -= len;
1071  if (!non_zero_found_short[l]) {
1072  /* test if non zero band. if so, stop doing i-stereo */
1073  for (j = 0; j < len; j++) {
1074  if (tab1[j] != 0) {
1075  non_zero_found_short[l] = 1;
1076  goto found1;
1077  }
1078  }
1079  sf = g1->scale_factors[k + l];
1080  if (sf >= sf_max)
1081  goto found1;
1082 
1083  v1 = is_tab[0][sf];
1084  v2 = is_tab[1][sf];
1085  for (j = 0; j < len; j++) {
1086  tmp0 = tab0[j];
1087  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1088  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1089  }
1090  } else {
1091 found1:
1092  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1093  /* lower part of the spectrum : do ms stereo
1094  if enabled */
1095  for (j = 0; j < len; j++) {
1096  tmp0 = tab0[j];
1097  tmp1 = tab1[j];
1098  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1099  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1100  }
1101  }
1102  }
1103  }
1104  }
1105 
1106  non_zero_found = non_zero_found_short[0] |
1107  non_zero_found_short[1] |
1108  non_zero_found_short[2];
1109 
1110  for (i = g1->long_end - 1;i >= 0;i--) {
1111  len = band_size_long[s->sample_rate_index][i];
1112  tab0 -= len;
1113  tab1 -= len;
1114  /* test if non zero band. if so, stop doing i-stereo */
1115  if (!non_zero_found) {
1116  for (j = 0; j < len; j++) {
1117  if (tab1[j] != 0) {
1118  non_zero_found = 1;
1119  goto found2;
1120  }
1121  }
1122  /* for last band, use previous scale factor */
1123  k = (i == 21) ? 20 : i;
1124  sf = g1->scale_factors[k];
1125  if (sf >= sf_max)
1126  goto found2;
1127  v1 = is_tab[0][sf];
1128  v2 = is_tab[1][sf];
1129  for (j = 0; j < len; j++) {
1130  tmp0 = tab0[j];
1131  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1132  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1133  }
1134  } else {
1135 found2:
1136  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1137  /* lower part of the spectrum : do ms stereo
1138  if enabled */
1139  for (j = 0; j < len; j++) {
1140  tmp0 = tab0[j];
1141  tmp1 = tab1[j];
1142  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1143  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1144  }
1145  }
1146  }
1147  }
1148  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1149  /* ms stereo ONLY */
1150  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1151  global gain */
1152 #if USE_FLOATS
1153  s->fdsp->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1154 #else
1155  tab0 = g0->sb_hybrid;
1156  tab1 = g1->sb_hybrid;
1157  for (i = 0; i < 576; i++) {
1158  tmp0 = tab0[i];
1159  tmp1 = tab1[i];
1160  tab0[i] = tmp0 + tmp1;
1161  tab1[i] = tmp0 - tmp1;
1162  }
1163 #endif
1164  }
1165 }
1166 
1167 #if USE_FLOATS
1168 #if HAVE_MIPSFPU
1170 #endif /* HAVE_MIPSFPU */
1171 #else
1172 #if HAVE_MIPSDSP
1174 #endif /* HAVE_MIPSDSP */
1175 #endif /* USE_FLOATS */
1176 
1177 #ifndef compute_antialias
1178 #if USE_FLOATS
1179 #define AA(j) do { \
1180  float tmp0 = ptr[-1-j]; \
1181  float tmp1 = ptr[ j]; \
1182  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1183  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1184  } while (0)
1185 #else
1186 #define AA(j) do { \
1187  SUINT tmp0 = ptr[-1-j]; \
1188  SUINT tmp1 = ptr[ j]; \
1189  SUINT tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1190  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1191  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1192  } while (0)
1193 #endif
1194 
1196 {
1197  INTFLOAT *ptr;
1198  int n, i;
1199 
1200  /* we antialias only "long" bands */
1201  if (g->block_type == 2) {
1202  if (!g->switch_point)
1203  return;
1204  /* XXX: check this for 8000Hz case */
1205  n = 1;
1206  } else {
1207  n = SBLIMIT - 1;
1208  }
1209 
1210  ptr = g->sb_hybrid + 18;
1211  for (i = n; i > 0; i--) {
1212  AA(0);
1213  AA(1);
1214  AA(2);
1215  AA(3);
1216  AA(4);
1217  AA(5);
1218  AA(6);
1219  AA(7);
1220 
1221  ptr += 18;
1222  }
1223 }
1224 #endif /* compute_antialias */
1225 
1227  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1228 {
1229  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1230  INTFLOAT out2[12];
1231  int i, j, mdct_long_end, sblimit;
1232 
1233  /* find last non zero block */
1234  ptr = g->sb_hybrid + 576;
1235  ptr1 = g->sb_hybrid + 2 * 18;
1236  while (ptr >= ptr1) {
1237  int32_t *p;
1238  ptr -= 6;
1239  p = (int32_t*)ptr;
1240  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1241  break;
1242  }
1243  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1244 
1245  if (g->block_type == 2) {
1246  /* XXX: check for 8000 Hz */
1247  if (g->switch_point)
1248  mdct_long_end = 2;
1249  else
1250  mdct_long_end = 0;
1251  } else {
1252  mdct_long_end = sblimit;
1253  }
1254 
1255  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1256  mdct_long_end, g->switch_point,
1257  g->block_type);
1258 
1259  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1260  ptr = g->sb_hybrid + 18 * mdct_long_end;
1261 
1262  for (j = mdct_long_end; j < sblimit; j++) {
1263  /* select frequency inversion */
1264  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1265  out_ptr = sb_samples + j;
1266 
1267  for (i = 0; i < 6; i++) {
1268  *out_ptr = buf[4*i];
1269  out_ptr += SBLIMIT;
1270  }
1271  imdct12(out2, ptr + 0);
1272  for (i = 0; i < 6; i++) {
1273  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1274  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1275  out_ptr += SBLIMIT;
1276  }
1277  imdct12(out2, ptr + 1);
1278  for (i = 0; i < 6; i++) {
1279  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1280  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1281  out_ptr += SBLIMIT;
1282  }
1283  imdct12(out2, ptr + 2);
1284  for (i = 0; i < 6; i++) {
1285  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1286  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1287  buf[4*(i + 6*2)] = 0;
1288  }
1289  ptr += 18;
1290  buf += (j&3) != 3 ? 1 : (4*18-3);
1291  }
1292  /* zero bands */
1293  for (j = sblimit; j < SBLIMIT; j++) {
1294  /* overlap */
1295  out_ptr = sb_samples + j;
1296  for (i = 0; i < 18; i++) {
1297  *out_ptr = buf[4*i];
1298  buf[4*i] = 0;
1299  out_ptr += SBLIMIT;
1300  }
1301  buf += (j&3) != 3 ? 1 : (4*18-3);
1302  }
1303 }
1304 
1305 /* main layer3 decoding function */
1307 {
1308  int nb_granules, main_data_begin;
1309  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1310  GranuleDef *g;
1311  int16_t exponents[576]; //FIXME try INTFLOAT
1312 
1313  /* read side info */
1314  if (s->lsf) {
1315  main_data_begin = get_bits(&s->gb, 8);
1316  skip_bits(&s->gb, s->nb_channels);
1317  nb_granules = 1;
1318  } else {
1319  main_data_begin = get_bits(&s->gb, 9);
1320  if (s->nb_channels == 2)
1321  skip_bits(&s->gb, 3);
1322  else
1323  skip_bits(&s->gb, 5);
1324  nb_granules = 2;
1325  for (ch = 0; ch < s->nb_channels; ch++) {
1326  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1327  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1328  }
1329  }
1330 
1331  for (gr = 0; gr < nb_granules; gr++) {
1332  for (ch = 0; ch < s->nb_channels; ch++) {
1333  ff_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1334  g = &s->granules[ch][gr];
1335  g->part2_3_length = get_bits(&s->gb, 12);
1336  g->big_values = get_bits(&s->gb, 9);
1337  if (g->big_values > 288) {
1338  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1339  return AVERROR_INVALIDDATA;
1340  }
1341 
1342  g->global_gain = get_bits(&s->gb, 8);
1343  /* if MS stereo only is selected, we precompute the
1344  1/sqrt(2) renormalization factor */
1345  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1347  g->global_gain -= 2;
1348  if (s->lsf)
1349  g->scalefac_compress = get_bits(&s->gb, 9);
1350  else
1351  g->scalefac_compress = get_bits(&s->gb, 4);
1352  blocksplit_flag = get_bits1(&s->gb);
1353  if (blocksplit_flag) {
1354  g->block_type = get_bits(&s->gb, 2);
1355  if (g->block_type == 0) {
1356  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1357  return AVERROR_INVALIDDATA;
1358  }
1359  g->switch_point = get_bits1(&s->gb);
1360  for (i = 0; i < 2; i++)
1361  g->table_select[i] = get_bits(&s->gb, 5);
1362  for (i = 0; i < 3; i++)
1363  g->subblock_gain[i] = get_bits(&s->gb, 3);
1364  init_short_region(s, g);
1365  } else {
1366  int region_address1, region_address2;
1367  g->block_type = 0;
1368  g->switch_point = 0;
1369  for (i = 0; i < 3; i++)
1370  g->table_select[i] = get_bits(&s->gb, 5);
1371  /* compute huffman coded region sizes */
1372  region_address1 = get_bits(&s->gb, 4);
1373  region_address2 = get_bits(&s->gb, 3);
1374  ff_dlog(s->avctx, "region1=%d region2=%d\n",
1375  region_address1, region_address2);
1376  init_long_region(s, g, region_address1, region_address2);
1377  }
1378  region_offset2size(g);
1379  compute_band_indexes(s, g);
1380 
1381  g->preflag = 0;
1382  if (!s->lsf)
1383  g->preflag = get_bits1(&s->gb);
1384  g->scalefac_scale = get_bits1(&s->gb);
1385  g->count1table_select = get_bits1(&s->gb);
1386  ff_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1387  g->block_type, g->switch_point);
1388  }
1389  }
1390 
1391  if (!s->adu_mode) {
1392  int skip;
1393  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1394  s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0,
1395  FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
1396  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1397  /* now we get bits from the main_data_begin offset */
1398  ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1399  main_data_begin, s->last_buf_size);
1400 
1401  memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize);
1402  s->in_gb = s->gb;
1403  init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8);
1404  s->last_buf_size <<= 3;
1405  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1406  for (ch = 0; ch < s->nb_channels; ch++) {
1407  g = &s->granules[ch][gr];
1408  s->last_buf_size += g->part2_3_length;
1409  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1410  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1411  }
1412  }
1413  skip = s->last_buf_size - 8 * main_data_begin;
1414  if (skip >= s->gb.size_in_bits - s->extrasize * 8 && s->in_gb.buffer) {
1415  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8);
1416  s->gb = s->in_gb;
1417  s->in_gb.buffer = NULL;
1418  s->extrasize = 0;
1419  } else {
1420  skip_bits_long(&s->gb, skip);
1421  }
1422  } else {
1423  gr = 0;
1424  s->extrasize = 0;
1425  }
1426 
1427  for (; gr < nb_granules; gr++) {
1428  for (ch = 0; ch < s->nb_channels; ch++) {
1429  g = &s->granules[ch][gr];
1430  bits_pos = get_bits_count(&s->gb);
1431 
1432  if (!s->lsf) {
1433  uint8_t *sc;
1434  int slen, slen1, slen2;
1435 
1436  /* MPEG-1 scale factors */
1437  slen1 = slen_table[0][g->scalefac_compress];
1438  slen2 = slen_table[1][g->scalefac_compress];
1439  ff_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1440  if (g->block_type == 2) {
1441  n = g->switch_point ? 17 : 18;
1442  j = 0;
1443  if (slen1) {
1444  for (i = 0; i < n; i++)
1445  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1446  } else {
1447  for (i = 0; i < n; i++)
1448  g->scale_factors[j++] = 0;
1449  }
1450  if (slen2) {
1451  for (i = 0; i < 18; i++)
1452  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1453  for (i = 0; i < 3; i++)
1454  g->scale_factors[j++] = 0;
1455  } else {
1456  for (i = 0; i < 21; i++)
1457  g->scale_factors[j++] = 0;
1458  }
1459  } else {
1460  sc = s->granules[ch][0].scale_factors;
1461  j = 0;
1462  for (k = 0; k < 4; k++) {
1463  n = k == 0 ? 6 : 5;
1464  if ((g->scfsi & (0x8 >> k)) == 0) {
1465  slen = (k < 2) ? slen1 : slen2;
1466  if (slen) {
1467  for (i = 0; i < n; i++)
1468  g->scale_factors[j++] = get_bits(&s->gb, slen);
1469  } else {
1470  for (i = 0; i < n; i++)
1471  g->scale_factors[j++] = 0;
1472  }
1473  } else {
1474  /* simply copy from last granule */
1475  for (i = 0; i < n; i++) {
1476  g->scale_factors[j] = sc[j];
1477  j++;
1478  }
1479  }
1480  }
1481  g->scale_factors[j++] = 0;
1482  }
1483  } else {
1484  int tindex, tindex2, slen[4], sl, sf;
1485 
1486  /* LSF scale factors */
1487  if (g->block_type == 2)
1488  tindex = g->switch_point ? 2 : 1;
1489  else
1490  tindex = 0;
1491 
1492  sf = g->scalefac_compress;
1493  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1494  /* intensity stereo case */
1495  sf >>= 1;
1496  if (sf < 180) {
1497  lsf_sf_expand(slen, sf, 6, 6, 0);
1498  tindex2 = 3;
1499  } else if (sf < 244) {
1500  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1501  tindex2 = 4;
1502  } else {
1503  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1504  tindex2 = 5;
1505  }
1506  } else {
1507  /* normal case */
1508  if (sf < 400) {
1509  lsf_sf_expand(slen, sf, 5, 4, 4);
1510  tindex2 = 0;
1511  } else if (sf < 500) {
1512  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1513  tindex2 = 1;
1514  } else {
1515  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1516  tindex2 = 2;
1517  g->preflag = 1;
1518  }
1519  }
1520 
1521  j = 0;
1522  for (k = 0; k < 4; k++) {
1523  n = lsf_nsf_table[tindex2][tindex][k];
1524  sl = slen[k];
1525  if (sl) {
1526  for (i = 0; i < n; i++)
1527  g->scale_factors[j++] = get_bits(&s->gb, sl);
1528  } else {
1529  for (i = 0; i < n; i++)
1530  g->scale_factors[j++] = 0;
1531  }
1532  }
1533  /* XXX: should compute exact size */
1534  for (; j < 40; j++)
1535  g->scale_factors[j] = 0;
1536  }
1537 
1538  exponents_from_scale_factors(s, g, exponents);
1539 
1540  /* read Huffman coded residue */
1541  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1542  } /* ch */
1543 
1544  if (s->mode == MPA_JSTEREO)
1545  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1546 
1547  for (ch = 0; ch < s->nb_channels; ch++) {
1548  g = &s->granules[ch][gr];
1549 
1550  reorder_block(s, g);
1551  compute_antialias(s, g);
1552  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1553  }
1554  } /* gr */
1555  if (get_bits_count(&s->gb) < 0)
1556  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1557  return nb_granules * 18;
1558 }
1559 
1561  const uint8_t *buf, int buf_size)
1562 {
1563  int i, nb_frames, ch, ret;
1564  OUT_INT *samples_ptr;
1565 
1566  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1567 
1568  /* skip error protection field */
1569  if (s->error_protection)
1570  skip_bits(&s->gb, 16);
1571 
1572  switch(s->layer) {
1573  case 1:
1574  s->avctx->frame_size = 384;
1575  nb_frames = mp_decode_layer1(s);
1576  break;
1577  case 2:
1578  s->avctx->frame_size = 1152;
1579  nb_frames = mp_decode_layer2(s);
1580  break;
1581  case 3:
1582  s->avctx->frame_size = s->lsf ? 576 : 1152;
1583  default:
1584  nb_frames = mp_decode_layer3(s);
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) - s->extrasize;
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  s->extrasize = 0;
1598  }
1599 
1600  align_get_bits(&s->gb);
1601  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1602  i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
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  av_assert1(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  if(nb_frames < 0)
1614  return nb_frames;
1615 
1616  /* get output buffer */
1617  if (!samples) {
1618  av_assert0(s->frame);
1619  s->frame->nb_samples = s->avctx->frame_size;
1620  if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
1621  return ret;
1622  samples = (OUT_INT **)s->frame->extended_data;
1623  }
1624 
1625  /* apply the synthesis filter */
1626  for (ch = 0; ch < s->nb_channels; ch++) {
1627  int sample_stride;
1628  if (s->avctx->sample_fmt == OUT_FMT_P) {
1629  samples_ptr = samples[ch];
1630  sample_stride = 1;
1631  } else {
1632  samples_ptr = samples[0] + ch;
1633  sample_stride = s->nb_channels;
1634  }
1635  for (i = 0; i < nb_frames; i++) {
1636  RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
1637  &(s->synth_buf_offset[ch]),
1638  RENAME(ff_mpa_synth_window),
1639  &s->dither_state, samples_ptr,
1640  sample_stride, s->sb_samples[ch][i]);
1641  samples_ptr += 32 * sample_stride;
1642  }
1643  }
1644 
1645  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1646 }
1647 
1648 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1649  AVPacket *avpkt)
1650 {
1651  const uint8_t *buf = avpkt->data;
1652  int buf_size = avpkt->size;
1653  MPADecodeContext *s = avctx->priv_data;
1654  uint32_t header;
1655  int ret;
1656 
1657  int skipped = 0;
1658  while(buf_size && !*buf){
1659  buf++;
1660  buf_size--;
1661  skipped++;
1662  }
1663 
1664  if (buf_size < HEADER_SIZE)
1665  return AVERROR_INVALIDDATA;
1666 
1667  header = AV_RB32(buf);
1668  if (header>>8 == AV_RB32("TAG")>>8) {
1669  av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
1670  return buf_size + skipped;
1671  }
1672  ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
1673  if (ret < 0) {
1674  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1675  return AVERROR_INVALIDDATA;
1676  } else if (ret == 1) {
1677  /* free format: prepare to compute frame size */
1678  s->frame_size = -1;
1679  return AVERROR_INVALIDDATA;
1680  }
1681  /* update codec info */
1682  avctx->channels = s->nb_channels;
1683  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1684  if (!avctx->bit_rate)
1685  avctx->bit_rate = s->bit_rate;
1686 
1687  if (s->frame_size <= 0) {
1688  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1689  return AVERROR_INVALIDDATA;
1690  } else if (s->frame_size < buf_size) {
1691  av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
1692  buf_size= s->frame_size;
1693  }
1694 
1695  s->frame = data;
1696 
1697  ret = mp_decode_frame(s, NULL, buf, buf_size);
1698  if (ret >= 0) {
1699  s->frame->nb_samples = avctx->frame_size;
1700  *got_frame_ptr = 1;
1701  avctx->sample_rate = s->sample_rate;
1702  //FIXME maybe move the other codec info stuff from above here too
1703  } else {
1704  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1705  /* Only return an error if the bad frame makes up the whole packet or
1706  * the error is related to buffer management.
1707  * If there is more data in the packet, just consume the bad frame
1708  * instead of returning an error, which would discard the whole
1709  * packet. */
1710  *got_frame_ptr = 0;
1711  if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1712  return ret;
1713  }
1714  s->frame_size = 0;
1715  return buf_size + skipped;
1716 }
1717 
1719 {
1720  memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1721  memset(ctx->mdct_buf, 0, sizeof(ctx->mdct_buf));
1722  ctx->last_buf_size = 0;
1723  ctx->dither_state = 0;
1724 }
1725 
1726 static void flush(AVCodecContext *avctx)
1727 {
1728  mp_flush(avctx->priv_data);
1729 }
1730 
1731 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1732 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1733  int *got_frame_ptr, AVPacket *avpkt)
1734 {
1735  const uint8_t *buf = avpkt->data;
1736  int buf_size = avpkt->size;
1737  MPADecodeContext *s = avctx->priv_data;
1738  uint32_t header;
1739  int len, ret;
1740  int av_unused out_size;
1741 
1742  len = buf_size;
1743 
1744  // Discard too short frames
1745  if (buf_size < HEADER_SIZE) {
1746  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1747  return AVERROR_INVALIDDATA;
1748  }
1749 
1750 
1751  if (len > MPA_MAX_CODED_FRAME_SIZE)
1753 
1754  // Get header and restore sync word
1755  header = AV_RB32(buf) | 0xffe00000;
1756 
1757  ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
1758  if (ret < 0) {
1759  av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1760  return ret;
1761  }
1762  /* update codec info */
1763  avctx->sample_rate = s->sample_rate;
1764  avctx->channels = s->nb_channels;
1765  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1766  if (!avctx->bit_rate)
1767  avctx->bit_rate = s->bit_rate;
1768 
1769  s->frame_size = len;
1770 
1771  s->frame = data;
1772 
1773  ret = mp_decode_frame(s, NULL, buf, buf_size);
1774  if (ret < 0) {
1775  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1776  return ret;
1777  }
1778 
1779  *got_frame_ptr = 1;
1780 
1781  return buf_size;
1782 }
1783 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1784 
1785 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1786 
1787 /**
1788  * Context for MP3On4 decoder
1789  */
1790 typedef struct MP3On4DecodeContext {
1791  int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
1792  int syncword; ///< syncword patch
1793  const uint8_t *coff; ///< channel offsets in output buffer
1794  MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
1795 } MP3On4DecodeContext;
1796 
1797 #include "mpeg4audio.h"
1798 
1799 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1800 
1801 /* number of mp3 decoder instances */
1802 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1803 
1804 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1805 static const uint8_t chan_offset[8][5] = {
1806  { 0 },
1807  { 0 }, // C
1808  { 0 }, // FLR
1809  { 2, 0 }, // C FLR
1810  { 2, 0, 3 }, // C FLR BS
1811  { 2, 0, 3 }, // C FLR BLRS
1812  { 2, 0, 4, 3 }, // C FLR BLRS LFE
1813  { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1814 };
1815 
1816 /* mp3on4 channel layouts */
1817 static const int16_t chan_layout[8] = {
1818  0,
1826 };
1827 
1828 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1829 {
1830  MP3On4DecodeContext *s = avctx->priv_data;
1831  int i;
1832 
1833  if (s->mp3decctx[0])
1834  av_freep(&s->mp3decctx[0]->fdsp);
1835 
1836  for (i = 0; i < s->frames; i++)
1837  av_freep(&s->mp3decctx[i]);
1838 
1839  return 0;
1840 }
1841 
1842 
1843 static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
1844 {
1845  MP3On4DecodeContext *s = avctx->priv_data;
1846  MPEG4AudioConfig cfg;
1847  int i;
1848 
1849  if ((avctx->extradata_size < 2) || !avctx->extradata) {
1850  av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1851  return AVERROR_INVALIDDATA;
1852  }
1853 
1855  avctx->extradata_size * 8, 1);
1856  if (!cfg.chan_config || cfg.chan_config > 7) {
1857  av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1858  return AVERROR_INVALIDDATA;
1859  }
1860  s->frames = mp3Frames[cfg.chan_config];
1861  s->coff = chan_offset[cfg.chan_config];
1863  avctx->channel_layout = chan_layout[cfg.chan_config];
1864 
1865  if (cfg.sample_rate < 16000)
1866  s->syncword = 0xffe00000;
1867  else
1868  s->syncword = 0xfff00000;
1869 
1870  /* Init the first mp3 decoder in standard way, so that all tables get builded
1871  * We replace avctx->priv_data with the context of the first decoder so that
1872  * decode_init() does not have to be changed.
1873  * Other decoders will be initialized here copying data from the first context
1874  */
1875  // Allocate zeroed memory for the first decoder context
1876  s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1877  if (!s->mp3decctx[0])
1878  goto alloc_fail;
1879  // Put decoder context in place to make init_decode() happy
1880  avctx->priv_data = s->mp3decctx[0];
1881  decode_init(avctx);
1882  // Restore mp3on4 context pointer
1883  avctx->priv_data = s;
1884  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1885 
1886  /* Create a separate codec/context for each frame (first is already ok).
1887  * Each frame is 1 or 2 channels - up to 5 frames allowed
1888  */
1889  for (i = 1; i < s->frames; i++) {
1890  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1891  if (!s->mp3decctx[i])
1892  goto alloc_fail;
1893  s->mp3decctx[i]->adu_mode = 1;
1894  s->mp3decctx[i]->avctx = avctx;
1895  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1896  s->mp3decctx[i]->fdsp = s->mp3decctx[0]->fdsp;
1897  }
1898 
1899  return 0;
1900 alloc_fail:
1901  decode_close_mp3on4(avctx);
1902  return AVERROR(ENOMEM);
1903 }
1904 
1905 
1906 static void flush_mp3on4(AVCodecContext *avctx)
1907 {
1908  int i;
1909  MP3On4DecodeContext *s = avctx->priv_data;
1910 
1911  for (i = 0; i < s->frames; i++)
1912  mp_flush(s->mp3decctx[i]);
1913 }
1914 
1915 
1916 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1917  int *got_frame_ptr, AVPacket *avpkt)
1918 {
1919  AVFrame *frame = data;
1920  const uint8_t *buf = avpkt->data;
1921  int buf_size = avpkt->size;
1922  MP3On4DecodeContext *s = avctx->priv_data;
1923  MPADecodeContext *m;
1924  int fsize, len = buf_size, out_size = 0;
1925  uint32_t header;
1926  OUT_INT **out_samples;
1927  OUT_INT *outptr[2];
1928  int fr, ch, ret;
1929 
1930  /* get output buffer */
1931  frame->nb_samples = MPA_FRAME_SIZE;
1932  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1933  return ret;
1934  out_samples = (OUT_INT **)frame->extended_data;
1935 
1936  // Discard too short frames
1937  if (buf_size < HEADER_SIZE)
1938  return AVERROR_INVALIDDATA;
1939 
1940  avctx->bit_rate = 0;
1941 
1942  ch = 0;
1943  for (fr = 0; fr < s->frames; fr++) {
1944  fsize = AV_RB16(buf) >> 4;
1945  fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1946  m = s->mp3decctx[fr];
1947  av_assert1(m);
1948 
1949  if (fsize < HEADER_SIZE) {
1950  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1951  return AVERROR_INVALIDDATA;
1952  }
1953  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1954 
1956  if (ret < 0) {
1957  av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n");
1958  return AVERROR_INVALIDDATA;
1959  }
1960 
1961  if (ch + m->nb_channels > avctx->channels ||
1962  s->coff[fr] + m->nb_channels > avctx->channels) {
1963  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1964  "channel count\n");
1965  return AVERROR_INVALIDDATA;
1966  }
1967  ch += m->nb_channels;
1968 
1969  outptr[0] = out_samples[s->coff[fr]];
1970  if (m->nb_channels > 1)
1971  outptr[1] = out_samples[s->coff[fr] + 1];
1972 
1973  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0) {
1974  av_log(avctx, AV_LOG_ERROR, "failed to decode channel %d\n", ch);
1975  memset(outptr[0], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
1976  if (m->nb_channels > 1)
1977  memset(outptr[1], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
1978  ret = m->nb_channels * MPA_FRAME_SIZE*sizeof(OUT_INT);
1979  }
1980 
1981  out_size += ret;
1982  buf += fsize;
1983  len -= fsize;
1984 
1985  avctx->bit_rate += m->bit_rate;
1986  }
1987  if (ch != avctx->channels) {
1988  av_log(avctx, AV_LOG_ERROR, "failed to decode all channels\n");
1989  return AVERROR_INVALIDDATA;
1990  }
1991 
1992  /* update codec info */
1993  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1994 
1995  frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1996  *got_frame_ptr = 1;
1997 
1998  return buf_size;
1999 }
2000 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
#define MUL64(a, b)
Definition: mathops.h:54
#define SUINTFLOAT
static av_cold void decode_init_static(void)
#define MPA_MAX_CODED_FRAME_SIZE
Definition: mpegaudio.h:40
static int32_t scale_factor_mult[15][3]
#define AV_EF_AGGRESSIVE
consider things that a sane encoder should not do as an error
Definition: avcodec.h:3050
static double bound(const double threshold, const double val)
#define NULL
Definition: coverity.c:32
const char const char void * val
Definition: avisynth_c.h:771
static int16_t division_tab9[1<< 11]
#define AV_CH_LAYOUT_7POINT1
const char * s
Definition: avisynth_c.h:768
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
static uint32_t table_4_3_value[TABLE_4_3_SIZE]
static const uint8_t lsf_nsf_table[6][3][4]
static int shift(int a, int b)
Definition: sonic.c:82
#define SBLIMIT
Definition: mpegaudio.h:44
This structure describes decoded (raw) audio or video data.
Definition: frame.h:201
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
Reference: libavcodec/mpegaudiodec.c.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define HEADER_SIZE
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1811
#define AV_CH_LAYOUT_SURROUND
static float win(SuperEqualizerContext *s, float n, int N)
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:204
const char * g
Definition: vf_curves.c:112
static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g, int16_t *exponents)
static int8_t table_4_3_exp[TABLE_4_3_SIZE]
#define MPA_JSTEREO
Definition: mpegaudio.h:47
#define RENAME(name)
Definition: ffv1.h:205
#define LAST_BUF_SIZE
int size
Definition: avcodec.h:1672
const char * b
Definition: vf_curves.c:113
#define AV_EF_COMPLIANT
consider all spec non compliances as errors
Definition: avcodec.h:3049
#define AV_EF_BUFFER
detect improper bitstream length
Definition: avcodec.h:3044
const uint8_t * buffer
Definition: get_bits.h:56
const int ff_mpa_quant_bits[17]
Definition: mpegaudiodata.c:55
static const uint8_t mpa_pretab[2][22]
#define FRAC_ONE
Definition: mpegaudio.h:58
int out_size
Definition: movenc.c:55
#define AV_CH_LAYOUT_4POINT0
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:3043
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:222
GLfloat v0
Definition: opengl_enc.c:107
#define AV_CH_LAYOUT_STEREO
static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2, int n3)
uint8_t scale_factors[40]
#define SUINT
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
#define AV_CH_LAYOUT_5POINT0
mpeg audio layer common tables.
static const uint8_t slen_table[2][16]
Macro definitions for various function/variable attributes.
int32_t MPA_INT
Definition: mpegaudio.h:75
float INTFLOAT
Definition: aac_defines.h:86
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
int16_t OUT_INT
Definition: mpegaudio.h:76
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
uint8_t bits
Definition: crc.c:296
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2516
uint8_t
#define FIXR(x)
Definition: aac_defines.h:92
#define av_cold
Definition: attributes.h:82
#define FRAC_BITS
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
AVFloatDSPContext * fdsp
const int ff_mpa_quant_steps[17]
Definition: mpegaudiodata.c:47
static int l2_unscale_group(int steps, int mant, int scale_factor)
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
static av_cold void mpegaudio_tableinit(void)
const unsigned char *const ff_mpa_alloc_tables[5]
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1861
static const uint8_t mpa_huff_data[32][2]
#define SPLIT(dst, sf, n)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
static INTFLOAT csa_table[8][4]
static AVFrame * frame
static int l3_unscale(int value, int exponent)
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:104
static const uint8_t mpa_quad_codes[2][16]
uint8_t * data
Definition: avcodec.h:1671
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:199
#define FFMIN3(a, b, c)
Definition: common.h:97
#define ff_dlog(a,...)
bitstream reader API header.
static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2)
static av_cold int decode_close(AVCodecContext *avctx)
Definition: ansi.c:468
static const uint8_t header[24]
Definition: sdr2.c:67
static int bit_alloc(AC3EncodeContext *s, int snr_offset)
Run the bit allocation with a given SNR offset.
Definition: ac3enc.c:1064
AVCodecContext * avctx
#define C6
#define av_log(a,...)
#define AV_CH_LAYOUT_5POINT1
#define U(x)
Definition: vp56_arith.h:37
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:587
static VLC huff_vlc[16]
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define MODE_EXT_MS_STEREO
Definition: mpegaudiodata.h:34
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:127
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
Definition: vlc.h:38
#define AVERROR(e)
Definition: error.h:43
enum AVSampleFormat request_sample_fmt
desired sample format
Definition: avcodec.h:2581
static const struct endianess table[]
#define OUT_FMT
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
Definition: float_dsp.h:164
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
static void init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
static uint16_t band_index_long[9][23]
static av_cold int decode_init(AVCodecContext *avctx)
#define t1
Definition: regdef.h:29
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1841
simple assert() macros that are a bit more flexible than ISO C assert().
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
static VLC_TYPE huff_quad_vlc_tables[128+16][2]
#define FFMAX(a, b)
Definition: common.h:94
static VLC_TYPE huff_vlc_tables[0+128+128+128+130+128+154+166+142+204+190+170+542+460+662+414][2]
Definition: vlc.h:26
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2559
static const int32_t scale_factor_mult2[3][3]
#define READ_FLIP_SIGN(dst, src)
#define OUT_FMT_P
#define MPA_MAX_CHANNELS
Definition: mpegaudio.h:42
audio channel layout utility functions
#define C5
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:928
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:3034
#define C4
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
#define FFMIN(a, b)
Definition: common.h:96
static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
int size_in_bits
Definition: get_bits.h:58
Reference: libavcodec/mpegaudiodec.c.
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
static int mp_decode_layer2(MPADecodeContext *s)
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:554
int n
Definition: avisynth_c.h:684
#define ISQRT2
int frames
Definition: movenc.c:65
#define INTFLOAT
#define FF_ARRAY_ELEMS(a)
#define MULLx(x, y, s)
int bits
Definition: vlc.h:27
#define BACKSTEP_SIZE
static const uint8_t mpa_quad_bits[2][16]
int table_allocated
Definition: vlc.h:29
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2528
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
const uint8_t * bits
Used to store optimal huffman encoding results.
Libavcodec external API header.
int sb_hybrid[SBLIMIT *18]
static const int huff_vlc_tables_sizes[16]
enum AVCodecID codec_id
Definition: avcodec.h:1763
int sample_rate
samples per second
Definition: avcodec.h:2508
MPA_INT synth_buf[MPA_MAX_CHANNELS][512 *2]
static int mp_decode_layer3(MPADecodeContext *s)
static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples, const uint8_t *buf, int buf_size)
main external API structure.
Definition: avcodec.h:1746
static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
static INTFLOAT is_table[2][16]
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1663
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
#define FIXHR(a)
static void mp_flush(MPADecodeContext *ctx)
void * buf
Definition: avisynth_c.h:690
const int16_t * tab1
Definition: mace.c:144
int extradata_size
Definition: avcodec.h:1862
Replacements for frequently missing libm functions.
static void reorder_block(MPADecodeContext *s, GranuleDef *g)
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:313
uint8_t count1table_select
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:306
#define MODE_EXT_I_STEREO
Definition: mpegaudiodata.h:35
static const int huff_quad_vlc_tables_sizes[2]
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:425
static uint16_t scale_factor_modshift[64]
const uint16_t * codes
static INTFLOAT is_table_lsf[2][2][16]
static int16_t division_tab5[1<< 8]
static void init_short_region(MPADecodeContext *s, GranuleDef *g)
static const uint8_t band_size_long[9][22]
#define MPA_DECODE_HEADER
#define SCALE_GEN(v)
static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
MPEG Audio header decoder.
static int16_t *const division_tabs[4]
static void compute_imdct(MPADecodeContext *s, GranuleDef *g, INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:282
#define exp2(x)
Definition: libm.h:288
#define INIT_VLC_USE_NEW_STATIC
Definition: vlc.h:55
#define SHR(a, b)
static void imdct12(INTFLOAT *out, SUINTFLOAT *in)
mpeg audio declarations for both encoder and decoder.
const int ff_mpa_sblimit_table[5]
Definition: mpegaudiodata.c:45
int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, int bit_size, int sync_extension)
Parse MPEG-4 systems extradata from a raw buffer to retrieve audio configuration. ...
Definition: mpeg4audio.c:155
static int mp_decode_layer1(MPADecodeContext *s)
INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT *18]
void * priv_data
Definition: avcodec.h:1788
int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf)
Definition: mpegaudio.c:31
int len
int channels
number of audio channels
Definition: avcodec.h:2509
const uint8_t ff_mpeg4audio_channels[8]
Definition: mpeg4audio.c:67
MPA_DECODE_HEADER uint8_t last_buf[LAST_BUF_SIZE]
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
int synth_buf_offset[MPA_MAX_CHANNELS]
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:464
static VLC huff_quad_vlc[2]
FILE * out
Definition: movenc.c:54
#define av_freep(p)
#define av_always_inline
Definition: attributes.h:39
#define M_PI
Definition: mathematics.h:52
#define VLC_TYPE
Definition: vlc.h:24
static int huffman_decode(MPADecodeContext *s, GranuleDef *g, int16_t *exponents, int end_pos2)
static int64_t fsize(FILE *f)
Definition: audiomatch.c:28
mpeg audio layer decoder tables.
static int l1_unscale(int n, int mant, int scale_factor)
int sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]
static const HuffTable mpa_huff_tables[16]
static const float ci_table[8]
#define AA(j)
#define MULH3(x, y, s)
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
#define AV_CH_LAYOUT_MONO
#define C3
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define MPA_FRAME_SIZE
Definition: mpegaudio.h:37
static void region_offset2size(GranuleDef *g)
Convert region offsets to region sizes and truncate size to big_values.
This structure stores compressed data.
Definition: avcodec.h:1648
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:27
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
static void flush(AVCodecContext *avctx)
for(j=16;j >0;--j)
#define t2
Definition: regdef.h:30
static av_always_inline int get_bitsz(GetBitContext *s, int n)
Read 0-25 bits.
Definition: get_bits.h:276
#define av_unused
Definition: attributes.h:125
static int alloc_table(VLC *vlc, int size, int use_static)
Definition: bitstream.c:110
static const uint8_t band_size_short[9][13]
int adu_mode
0 for standard mp3, 1 for adu formatted mp3
static int16_t division_tab3[1<< 6]
GranuleDef granules[2][2]
static uint8_t tmp[11]
Definition: aes_ctr.c:26