mpegaudiodec.c
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1 /*
2  * MPEG Audio decoder
3  * Copyright (c) 2001, 2002 Fabrice Bellard
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
28 #include "avcodec.h"
29 #include "get_bits.h"
30 #include "internal.h"
31 #include "mathops.h"
32 #include "mpegaudiodsp.h"
33 #include "dsputil.h"
34 
35 /*
36  * TODO:
37  * - test lsf / mpeg25 extensively.
38  */
39 
40 #include "mpegaudio.h"
41 #include "mpegaudiodecheader.h"
42 
43 #define BACKSTEP_SIZE 512
44 #define EXTRABYTES 24
45 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
46 
47 /* layer 3 "granule" */
48 typedef struct GranuleDef {
56  int table_select[3];
57  int subblock_gain[3];
60  int region_size[3]; /* number of huffman codes in each region */
61  int preflag;
62  int short_start, long_end; /* long/short band indexes */
64  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
65 } GranuleDef;
66 
67 typedef struct MPADecodeContext {
71  /* next header (used in free format parsing) */
78  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
79  GranuleDef granules[2][2]; /* Used in Layer 3 */
80  int adu_mode;
88 
89 #if CONFIG_FLOAT
90 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
91 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
92 # define FIXR(x) ((float)(x))
93 # define FIXHR(x) ((float)(x))
94 # define MULH3(x, y, s) ((s)*(y)*(x))
95 # define MULLx(x, y, s) ((y)*(x))
96 # define RENAME(a) a ## _float
97 # define OUT_FMT AV_SAMPLE_FMT_FLT
98 # define OUT_FMT_P AV_SAMPLE_FMT_FLTP
99 #else
100 # define SHR(a,b) ((a)>>(b))
101 /* WARNING: only correct for positive numbers */
102 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
103 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
104 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
105 # define MULH3(x, y, s) MULH((s)*(x), y)
106 # define MULLx(x, y, s) MULL(x,y,s)
107 # define RENAME(a) a ## _fixed
108 # define OUT_FMT AV_SAMPLE_FMT_S16
109 # define OUT_FMT_P AV_SAMPLE_FMT_S16P
110 #endif
111 
112 /****************/
113 
114 #define HEADER_SIZE 4
115 
116 #include "mpegaudiodata.h"
117 #include "mpegaudiodectab.h"
118 
119 /* vlc structure for decoding layer 3 huffman tables */
120 static VLC huff_vlc[16];
122  0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
123  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
124  ][2];
125 static const int huff_vlc_tables_sizes[16] = {
126  0, 128, 128, 128, 130, 128, 154, 166,
127  142, 204, 190, 170, 542, 460, 662, 414
128 };
129 static VLC huff_quad_vlc[2];
130 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
131 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
132 /* computed from band_size_long */
133 static uint16_t band_index_long[9][23];
134 #include "mpegaudio_tablegen.h"
135 /* intensity stereo coef table */
136 static INTFLOAT is_table[2][16];
137 static INTFLOAT is_table_lsf[2][2][16];
138 static INTFLOAT csa_table[8][4];
139 
140 static int16_t division_tab3[1<<6 ];
141 static int16_t division_tab5[1<<8 ];
142 static int16_t division_tab9[1<<11];
143 
144 static int16_t * const division_tabs[4] = {
146 };
147 
148 /* lower 2 bits: modulo 3, higher bits: shift */
149 static uint16_t scale_factor_modshift[64];
150 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
152 /* mult table for layer 2 group quantization */
153 
154 #define SCALE_GEN(v) \
155 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
156 
157 static const int32_t scale_factor_mult2[3][3] = {
158  SCALE_GEN(4.0 / 3.0), /* 3 steps */
159  SCALE_GEN(4.0 / 5.0), /* 5 steps */
160  SCALE_GEN(4.0 / 9.0), /* 9 steps */
161 };
162 
168 {
169  int i, k, j = 0;
170  g->region_size[2] = 576 / 2;
171  for (i = 0; i < 3; i++) {
172  k = FFMIN(g->region_size[i], g->big_values);
173  g->region_size[i] = k - j;
174  j = k;
175  }
176 }
177 
179 {
180  if (g->block_type == 2) {
181  if (s->sample_rate_index != 8)
182  g->region_size[0] = (36 / 2);
183  else
184  g->region_size[0] = (72 / 2);
185  } else {
186  if (s->sample_rate_index <= 2)
187  g->region_size[0] = (36 / 2);
188  else if (s->sample_rate_index != 8)
189  g->region_size[0] = (54 / 2);
190  else
191  g->region_size[0] = (108 / 2);
192  }
193  g->region_size[1] = (576 / 2);
194 }
195 
196 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
197 {
198  int l;
199  g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
200  /* should not overflow */
201  l = FFMIN(ra1 + ra2 + 2, 22);
202  g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
203 }
204 
206 {
207  if (g->block_type == 2) {
208  if (g->switch_point) {
209  /* if switched mode, we handle the 36 first samples as
210  long blocks. For 8000Hz, we handle the 72 first
211  exponents as long blocks */
212  if (s->sample_rate_index <= 2)
213  g->long_end = 8;
214  else
215  g->long_end = 6;
216 
217  g->short_start = 3;
218  } else {
219  g->long_end = 0;
220  g->short_start = 0;
221  }
222  } else {
223  g->short_start = 13;
224  g->long_end = 22;
225  }
226 }
227 
228 /* layer 1 unscaling */
229 /* n = number of bits of the mantissa minus 1 */
230 static inline int l1_unscale(int n, int mant, int scale_factor)
231 {
232  int shift, mod;
233  int64_t val;
234 
235  shift = scale_factor_modshift[scale_factor];
236  mod = shift & 3;
237  shift >>= 2;
238  val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
239  shift += n;
240  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
241  return (int)((val + (1LL << (shift - 1))) >> shift);
242 }
243 
244 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
245 {
246  int shift, mod, val;
247 
248  shift = scale_factor_modshift[scale_factor];
249  mod = shift & 3;
250  shift >>= 2;
251 
252  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
253  /* NOTE: at this point, 0 <= shift <= 21 */
254  if (shift > 0)
255  val = (val + (1 << (shift - 1))) >> shift;
256  return val;
257 }
258 
259 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
260 static inline int l3_unscale(int value, int exponent)
261 {
262  unsigned int m;
263  int e;
264 
265  e = table_4_3_exp [4 * value + (exponent & 3)];
266  m = table_4_3_value[4 * value + (exponent & 3)];
267  e -= exponent >> 2;
268  assert(e >= 1);
269  if (e > 31)
270  return 0;
271  m = (m + (1 << (e - 1))) >> e;
272 
273  return m;
274 }
275 
276 static av_cold void decode_init_static(void)
277 {
278  int i, j, k;
279  int offset;
280 
281  /* scale factors table for layer 1/2 */
282  for (i = 0; i < 64; i++) {
283  int shift, mod;
284  /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
285  shift = i / 3;
286  mod = i % 3;
287  scale_factor_modshift[i] = mod | (shift << 2);
288  }
289 
290  /* scale factor multiply for layer 1 */
291  for (i = 0; i < 15; i++) {
292  int n, norm;
293  n = i + 2;
294  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
295  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
296  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
297  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
298  av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
299  scale_factor_mult[i][0],
300  scale_factor_mult[i][1],
301  scale_factor_mult[i][2]);
302  }
303 
304  RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
305 
306  /* huffman decode tables */
307  offset = 0;
308  for (i = 1; i < 16; i++) {
309  const HuffTable *h = &mpa_huff_tables[i];
310  int xsize, x, y;
311  uint8_t tmp_bits [512] = { 0 };
312  uint16_t tmp_codes[512] = { 0 };
313 
314  xsize = h->xsize;
315 
316  j = 0;
317  for (x = 0; x < xsize; x++) {
318  for (y = 0; y < xsize; y++) {
319  tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
320  tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
321  }
322  }
323 
324  /* XXX: fail test */
325  huff_vlc[i].table = huff_vlc_tables+offset;
326  huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
327  init_vlc(&huff_vlc[i], 7, 512,
328  tmp_bits, 1, 1, tmp_codes, 2, 2,
330  offset += huff_vlc_tables_sizes[i];
331  }
332  assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
333 
334  offset = 0;
335  for (i = 0; i < 2; i++) {
336  huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
337  huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
338  init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
339  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
341  offset += huff_quad_vlc_tables_sizes[i];
342  }
343  assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
344 
345  for (i = 0; i < 9; i++) {
346  k = 0;
347  for (j = 0; j < 22; j++) {
348  band_index_long[i][j] = k;
349  k += band_size_long[i][j];
350  }
351  band_index_long[i][22] = k;
352  }
353 
354  /* compute n ^ (4/3) and store it in mantissa/exp format */
355 
357 
358  for (i = 0; i < 4; i++) {
359  if (ff_mpa_quant_bits[i] < 0) {
360  for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
361  int val1, val2, val3, steps;
362  int val = j;
363  steps = ff_mpa_quant_steps[i];
364  val1 = val % steps;
365  val /= steps;
366  val2 = val % steps;
367  val3 = val / steps;
368  division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
369  }
370  }
371  }
372 
373 
374  for (i = 0; i < 7; i++) {
375  float f;
376  INTFLOAT v;
377  if (i != 6) {
378  f = tan((double)i * M_PI / 12.0);
379  v = FIXR(f / (1.0 + f));
380  } else {
381  v = FIXR(1.0);
382  }
383  is_table[0][ i] = v;
384  is_table[1][6 - i] = v;
385  }
386  /* invalid values */
387  for (i = 7; i < 16; i++)
388  is_table[0][i] = is_table[1][i] = 0.0;
389 
390  for (i = 0; i < 16; i++) {
391  double f;
392  int e, k;
393 
394  for (j = 0; j < 2; j++) {
395  e = -(j + 1) * ((i + 1) >> 1);
396  f = pow(2.0, e / 4.0);
397  k = i & 1;
398  is_table_lsf[j][k ^ 1][i] = FIXR(f);
399  is_table_lsf[j][k ][i] = FIXR(1.0);
400  av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
401  i, j, (float) is_table_lsf[j][0][i],
402  (float) is_table_lsf[j][1][i]);
403  }
404  }
405 
406  for (i = 0; i < 8; i++) {
407  float ci, cs, ca;
408  ci = ci_table[i];
409  cs = 1.0 / sqrt(1.0 + ci * ci);
410  ca = cs * ci;
411 #if !CONFIG_FLOAT
412  csa_table[i][0] = FIXHR(cs/4);
413  csa_table[i][1] = FIXHR(ca/4);
414  csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
415  csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
416 #else
417  csa_table[i][0] = cs;
418  csa_table[i][1] = ca;
419  csa_table[i][2] = ca + cs;
420  csa_table[i][3] = ca - cs;
421 #endif
422  }
423 }
424 
425 static av_cold int decode_init(AVCodecContext * avctx)
426 {
427  static int initialized_tables = 0;
428  MPADecodeContext *s = avctx->priv_data;
429 
430  if (!initialized_tables) {
432  initialized_tables = 1;
433  }
434 
435  s->avctx = avctx;
436 
437  ff_mpadsp_init(&s->mpadsp);
438  ff_dsputil_init(&s->dsp, avctx);
439 
440  if (avctx->request_sample_fmt == OUT_FMT &&
441  avctx->codec_id != AV_CODEC_ID_MP3ON4)
442  avctx->sample_fmt = OUT_FMT;
443  else
444  avctx->sample_fmt = OUT_FMT_P;
445  s->err_recognition = avctx->err_recognition;
446 
447  if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
448  s->adu_mode = 1;
449 
451  avctx->coded_frame = &s->frame;
452 
453  return 0;
454 }
455 
456 #define C3 FIXHR(0.86602540378443864676/2)
457 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
458 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
459 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
460 
461 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
462  cases. */
463 static void imdct12(INTFLOAT *out, INTFLOAT *in)
464 {
465  INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
466 
467  in0 = in[0*3];
468  in1 = in[1*3] + in[0*3];
469  in2 = in[2*3] + in[1*3];
470  in3 = in[3*3] + in[2*3];
471  in4 = in[4*3] + in[3*3];
472  in5 = in[5*3] + in[4*3];
473  in5 += in3;
474  in3 += in1;
475 
476  in2 = MULH3(in2, C3, 2);
477  in3 = MULH3(in3, C3, 4);
478 
479  t1 = in0 - in4;
480  t2 = MULH3(in1 - in5, C4, 2);
481 
482  out[ 7] =
483  out[10] = t1 + t2;
484  out[ 1] =
485  out[ 4] = t1 - t2;
486 
487  in0 += SHR(in4, 1);
488  in4 = in0 + in2;
489  in5 += 2*in1;
490  in1 = MULH3(in5 + in3, C5, 1);
491  out[ 8] =
492  out[ 9] = in4 + in1;
493  out[ 2] =
494  out[ 3] = in4 - in1;
495 
496  in0 -= in2;
497  in5 = MULH3(in5 - in3, C6, 2);
498  out[ 0] =
499  out[ 5] = in0 - in5;
500  out[ 6] =
501  out[11] = in0 + in5;
502 }
503 
504 /* return the number of decoded frames */
506 {
507  int bound, i, v, n, ch, j, mant;
508  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
509  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
510 
511  if (s->mode == MPA_JSTEREO)
512  bound = (s->mode_ext + 1) * 4;
513  else
514  bound = SBLIMIT;
515 
516  /* allocation bits */
517  for (i = 0; i < bound; i++) {
518  for (ch = 0; ch < s->nb_channels; ch++) {
519  allocation[ch][i] = get_bits(&s->gb, 4);
520  }
521  }
522  for (i = bound; i < SBLIMIT; i++)
523  allocation[0][i] = get_bits(&s->gb, 4);
524 
525  /* scale factors */
526  for (i = 0; i < bound; i++) {
527  for (ch = 0; ch < s->nb_channels; ch++) {
528  if (allocation[ch][i])
529  scale_factors[ch][i] = get_bits(&s->gb, 6);
530  }
531  }
532  for (i = bound; i < SBLIMIT; i++) {
533  if (allocation[0][i]) {
534  scale_factors[0][i] = get_bits(&s->gb, 6);
535  scale_factors[1][i] = get_bits(&s->gb, 6);
536  }
537  }
538 
539  /* compute samples */
540  for (j = 0; j < 12; j++) {
541  for (i = 0; i < bound; i++) {
542  for (ch = 0; ch < s->nb_channels; ch++) {
543  n = allocation[ch][i];
544  if (n) {
545  mant = get_bits(&s->gb, n + 1);
546  v = l1_unscale(n, mant, scale_factors[ch][i]);
547  } else {
548  v = 0;
549  }
550  s->sb_samples[ch][j][i] = v;
551  }
552  }
553  for (i = bound; i < SBLIMIT; i++) {
554  n = allocation[0][i];
555  if (n) {
556  mant = get_bits(&s->gb, n + 1);
557  v = l1_unscale(n, mant, scale_factors[0][i]);
558  s->sb_samples[0][j][i] = v;
559  v = l1_unscale(n, mant, scale_factors[1][i]);
560  s->sb_samples[1][j][i] = v;
561  } else {
562  s->sb_samples[0][j][i] = 0;
563  s->sb_samples[1][j][i] = 0;
564  }
565  }
566  }
567  return 12;
568 }
569 
571 {
572  int sblimit; /* number of used subbands */
573  const unsigned char *alloc_table;
574  int table, bit_alloc_bits, i, j, ch, bound, v;
575  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
576  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
577  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
578  int scale, qindex, bits, steps, k, l, m, b;
579 
580  /* select decoding table */
581  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
582  s->sample_rate, s->lsf);
583  sblimit = ff_mpa_sblimit_table[table];
584  alloc_table = ff_mpa_alloc_tables[table];
585 
586  if (s->mode == MPA_JSTEREO)
587  bound = (s->mode_ext + 1) * 4;
588  else
589  bound = sblimit;
590 
591  av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
592 
593  /* sanity check */
594  if (bound > sblimit)
595  bound = sblimit;
596 
597  /* parse bit allocation */
598  j = 0;
599  for (i = 0; i < bound; i++) {
600  bit_alloc_bits = alloc_table[j];
601  for (ch = 0; ch < s->nb_channels; ch++)
602  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
603  j += 1 << bit_alloc_bits;
604  }
605  for (i = bound; i < sblimit; i++) {
606  bit_alloc_bits = alloc_table[j];
607  v = get_bits(&s->gb, bit_alloc_bits);
608  bit_alloc[0][i] = v;
609  bit_alloc[1][i] = v;
610  j += 1 << bit_alloc_bits;
611  }
612 
613  /* scale codes */
614  for (i = 0; i < sblimit; i++) {
615  for (ch = 0; ch < s->nb_channels; ch++) {
616  if (bit_alloc[ch][i])
617  scale_code[ch][i] = get_bits(&s->gb, 2);
618  }
619  }
620 
621  /* scale factors */
622  for (i = 0; i < sblimit; i++) {
623  for (ch = 0; ch < s->nb_channels; ch++) {
624  if (bit_alloc[ch][i]) {
625  sf = scale_factors[ch][i];
626  switch (scale_code[ch][i]) {
627  default:
628  case 0:
629  sf[0] = get_bits(&s->gb, 6);
630  sf[1] = get_bits(&s->gb, 6);
631  sf[2] = get_bits(&s->gb, 6);
632  break;
633  case 2:
634  sf[0] = get_bits(&s->gb, 6);
635  sf[1] = sf[0];
636  sf[2] = sf[0];
637  break;
638  case 1:
639  sf[0] = get_bits(&s->gb, 6);
640  sf[2] = get_bits(&s->gb, 6);
641  sf[1] = sf[0];
642  break;
643  case 3:
644  sf[0] = get_bits(&s->gb, 6);
645  sf[2] = get_bits(&s->gb, 6);
646  sf[1] = sf[2];
647  break;
648  }
649  }
650  }
651  }
652 
653  /* samples */
654  for (k = 0; k < 3; k++) {
655  for (l = 0; l < 12; l += 3) {
656  j = 0;
657  for (i = 0; i < bound; i++) {
658  bit_alloc_bits = alloc_table[j];
659  for (ch = 0; ch < s->nb_channels; ch++) {
660  b = bit_alloc[ch][i];
661  if (b) {
662  scale = scale_factors[ch][i][k];
663  qindex = alloc_table[j+b];
664  bits = ff_mpa_quant_bits[qindex];
665  if (bits < 0) {
666  int v2;
667  /* 3 values at the same time */
668  v = get_bits(&s->gb, -bits);
669  v2 = division_tabs[qindex][v];
670  steps = ff_mpa_quant_steps[qindex];
671 
672  s->sb_samples[ch][k * 12 + l + 0][i] =
673  l2_unscale_group(steps, v2 & 15, scale);
674  s->sb_samples[ch][k * 12 + l + 1][i] =
675  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
676  s->sb_samples[ch][k * 12 + l + 2][i] =
677  l2_unscale_group(steps, v2 >> 8 , scale);
678  } else {
679  for (m = 0; m < 3; m++) {
680  v = get_bits(&s->gb, bits);
681  v = l1_unscale(bits - 1, v, scale);
682  s->sb_samples[ch][k * 12 + l + m][i] = v;
683  }
684  }
685  } else {
686  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
687  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
688  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
689  }
690  }
691  /* next subband in alloc table */
692  j += 1 << bit_alloc_bits;
693  }
694  /* XXX: find a way to avoid this duplication of code */
695  for (i = bound; i < sblimit; i++) {
696  bit_alloc_bits = alloc_table[j];
697  b = bit_alloc[0][i];
698  if (b) {
699  int mant, scale0, scale1;
700  scale0 = scale_factors[0][i][k];
701  scale1 = scale_factors[1][i][k];
702  qindex = alloc_table[j+b];
703  bits = ff_mpa_quant_bits[qindex];
704  if (bits < 0) {
705  /* 3 values at the same time */
706  v = get_bits(&s->gb, -bits);
707  steps = ff_mpa_quant_steps[qindex];
708  mant = v % steps;
709  v = v / steps;
710  s->sb_samples[0][k * 12 + l + 0][i] =
711  l2_unscale_group(steps, mant, scale0);
712  s->sb_samples[1][k * 12 + l + 0][i] =
713  l2_unscale_group(steps, mant, scale1);
714  mant = v % steps;
715  v = v / steps;
716  s->sb_samples[0][k * 12 + l + 1][i] =
717  l2_unscale_group(steps, mant, scale0);
718  s->sb_samples[1][k * 12 + l + 1][i] =
719  l2_unscale_group(steps, mant, scale1);
720  s->sb_samples[0][k * 12 + l + 2][i] =
721  l2_unscale_group(steps, v, scale0);
722  s->sb_samples[1][k * 12 + l + 2][i] =
723  l2_unscale_group(steps, v, scale1);
724  } else {
725  for (m = 0; m < 3; m++) {
726  mant = get_bits(&s->gb, bits);
727  s->sb_samples[0][k * 12 + l + m][i] =
728  l1_unscale(bits - 1, mant, scale0);
729  s->sb_samples[1][k * 12 + l + m][i] =
730  l1_unscale(bits - 1, mant, scale1);
731  }
732  }
733  } else {
734  s->sb_samples[0][k * 12 + l + 0][i] = 0;
735  s->sb_samples[0][k * 12 + l + 1][i] = 0;
736  s->sb_samples[0][k * 12 + l + 2][i] = 0;
737  s->sb_samples[1][k * 12 + l + 0][i] = 0;
738  s->sb_samples[1][k * 12 + l + 1][i] = 0;
739  s->sb_samples[1][k * 12 + l + 2][i] = 0;
740  }
741  /* next subband in alloc table */
742  j += 1 << bit_alloc_bits;
743  }
744  /* fill remaining samples to zero */
745  for (i = sblimit; i < SBLIMIT; i++) {
746  for (ch = 0; ch < s->nb_channels; ch++) {
747  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
748  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
749  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
750  }
751  }
752  }
753  }
754  return 3 * 12;
755 }
756 
757 #define SPLIT(dst,sf,n) \
758  if (n == 3) { \
759  int m = (sf * 171) >> 9; \
760  dst = sf - 3 * m; \
761  sf = m; \
762  } else if (n == 4) { \
763  dst = sf & 3; \
764  sf >>= 2; \
765  } else if (n == 5) { \
766  int m = (sf * 205) >> 10; \
767  dst = sf - 5 * m; \
768  sf = m; \
769  } else if (n == 6) { \
770  int m = (sf * 171) >> 10; \
771  dst = sf - 6 * m; \
772  sf = m; \
773  } else { \
774  dst = 0; \
775  }
776 
777 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
778  int n3)
779 {
780  SPLIT(slen[3], sf, n3)
781  SPLIT(slen[2], sf, n2)
782  SPLIT(slen[1], sf, n1)
783  slen[0] = sf;
784 }
785 
787  int16_t *exponents)
788 {
789  const uint8_t *bstab, *pretab;
790  int len, i, j, k, l, v0, shift, gain, gains[3];
791  int16_t *exp_ptr;
792 
793  exp_ptr = exponents;
794  gain = g->global_gain - 210;
795  shift = g->scalefac_scale + 1;
796 
797  bstab = band_size_long[s->sample_rate_index];
798  pretab = mpa_pretab[g->preflag];
799  for (i = 0; i < g->long_end; i++) {
800  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
801  len = bstab[i];
802  for (j = len; j > 0; j--)
803  *exp_ptr++ = v0;
804  }
805 
806  if (g->short_start < 13) {
807  bstab = band_size_short[s->sample_rate_index];
808  gains[0] = gain - (g->subblock_gain[0] << 3);
809  gains[1] = gain - (g->subblock_gain[1] << 3);
810  gains[2] = gain - (g->subblock_gain[2] << 3);
811  k = g->long_end;
812  for (i = g->short_start; i < 13; i++) {
813  len = bstab[i];
814  for (l = 0; l < 3; l++) {
815  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
816  for (j = len; j > 0; j--)
817  *exp_ptr++ = v0;
818  }
819  }
820  }
821 }
822 
823 /* handle n = 0 too */
824 static inline int get_bitsz(GetBitContext *s, int n)
825 {
826  return n ? get_bits(s, n) : 0;
827 }
828 
829 
830 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
831  int *end_pos2)
832 {
833  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
834  s->gb = s->in_gb;
835  s->in_gb.buffer = NULL;
836  assert((get_bits_count(&s->gb) & 7) == 0);
837  skip_bits_long(&s->gb, *pos - *end_pos);
838  *end_pos2 =
839  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
840  *pos = get_bits_count(&s->gb);
841  }
842 }
843 
844 /* Following is a optimized code for
845  INTFLOAT v = *src
846  if(get_bits1(&s->gb))
847  v = -v;
848  *dst = v;
849 */
850 #if CONFIG_FLOAT
851 #define READ_FLIP_SIGN(dst,src) \
852  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
853  AV_WN32A(dst, v);
854 #else
855 #define READ_FLIP_SIGN(dst,src) \
856  v = -get_bits1(&s->gb); \
857  *(dst) = (*(src) ^ v) - v;
858 #endif
859 
861  int16_t *exponents, int end_pos2)
862 {
863  int s_index;
864  int i;
865  int last_pos, bits_left;
866  VLC *vlc;
867  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
868 
869  /* low frequencies (called big values) */
870  s_index = 0;
871  for (i = 0; i < 3; i++) {
872  int j, k, l, linbits;
873  j = g->region_size[i];
874  if (j == 0)
875  continue;
876  /* select vlc table */
877  k = g->table_select[i];
878  l = mpa_huff_data[k][0];
879  linbits = mpa_huff_data[k][1];
880  vlc = &huff_vlc[l];
881 
882  if (!l) {
883  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
884  s_index += 2 * j;
885  continue;
886  }
887 
888  /* read huffcode and compute each couple */
889  for (; j > 0; j--) {
890  int exponent, x, y;
891  int v;
892  int pos = get_bits_count(&s->gb);
893 
894  if (pos >= end_pos){
895  switch_buffer(s, &pos, &end_pos, &end_pos2);
896  if (pos >= end_pos)
897  break;
898  }
899  y = get_vlc2(&s->gb, vlc->table, 7, 3);
900 
901  if (!y) {
902  g->sb_hybrid[s_index ] =
903  g->sb_hybrid[s_index+1] = 0;
904  s_index += 2;
905  continue;
906  }
907 
908  exponent= exponents[s_index];
909 
910  av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
911  i, g->region_size[i] - j, x, y, exponent);
912  if (y & 16) {
913  x = y >> 5;
914  y = y & 0x0f;
915  if (x < 15) {
916  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
917  } else {
918  x += get_bitsz(&s->gb, linbits);
919  v = l3_unscale(x, exponent);
920  if (get_bits1(&s->gb))
921  v = -v;
922  g->sb_hybrid[s_index] = v;
923  }
924  if (y < 15) {
925  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
926  } else {
927  y += get_bitsz(&s->gb, linbits);
928  v = l3_unscale(y, exponent);
929  if (get_bits1(&s->gb))
930  v = -v;
931  g->sb_hybrid[s_index+1] = v;
932  }
933  } else {
934  x = y >> 5;
935  y = y & 0x0f;
936  x += y;
937  if (x < 15) {
938  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
939  } else {
940  x += get_bitsz(&s->gb, linbits);
941  v = l3_unscale(x, exponent);
942  if (get_bits1(&s->gb))
943  v = -v;
944  g->sb_hybrid[s_index+!!y] = v;
945  }
946  g->sb_hybrid[s_index + !y] = 0;
947  }
948  s_index += 2;
949  }
950  }
951 
952  /* high frequencies */
953  vlc = &huff_quad_vlc[g->count1table_select];
954  last_pos = 0;
955  while (s_index <= 572) {
956  int pos, code;
957  pos = get_bits_count(&s->gb);
958  if (pos >= end_pos) {
959  if (pos > end_pos2 && last_pos) {
960  /* some encoders generate an incorrect size for this
961  part. We must go back into the data */
962  s_index -= 4;
963  skip_bits_long(&s->gb, last_pos - pos);
964  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
966  s_index=0;
967  break;
968  }
969  switch_buffer(s, &pos, &end_pos, &end_pos2);
970  if (pos >= end_pos)
971  break;
972  }
973  last_pos = pos;
974 
975  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
976  av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
977  g->sb_hybrid[s_index+0] =
978  g->sb_hybrid[s_index+1] =
979  g->sb_hybrid[s_index+2] =
980  g->sb_hybrid[s_index+3] = 0;
981  while (code) {
982  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
983  int v;
984  int pos = s_index + idxtab[code];
985  code ^= 8 >> idxtab[code];
986  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
987  }
988  s_index += 4;
989  }
990  /* skip extension bits */
991  bits_left = end_pos2 - get_bits_count(&s->gb);
992  if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {
993  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
994  s_index=0;
995  } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
996  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
997  s_index = 0;
998  }
999  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
1000  skip_bits_long(&s->gb, bits_left);
1001 
1002  i = get_bits_count(&s->gb);
1003  switch_buffer(s, &i, &end_pos, &end_pos2);
1004 
1005  return 0;
1006 }
1007 
1008 /* Reorder short blocks from bitstream order to interleaved order. It
1009  would be faster to do it in parsing, but the code would be far more
1010  complicated */
1012 {
1013  int i, j, len;
1014  INTFLOAT *ptr, *dst, *ptr1;
1015  INTFLOAT tmp[576];
1016 
1017  if (g->block_type != 2)
1018  return;
1019 
1020  if (g->switch_point) {
1021  if (s->sample_rate_index != 8)
1022  ptr = g->sb_hybrid + 36;
1023  else
1024  ptr = g->sb_hybrid + 72;
1025  } else {
1026  ptr = g->sb_hybrid;
1027  }
1028 
1029  for (i = g->short_start; i < 13; i++) {
1030  len = band_size_short[s->sample_rate_index][i];
1031  ptr1 = ptr;
1032  dst = tmp;
1033  for (j = len; j > 0; j--) {
1034  *dst++ = ptr[0*len];
1035  *dst++ = ptr[1*len];
1036  *dst++ = ptr[2*len];
1037  ptr++;
1038  }
1039  ptr += 2 * len;
1040  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1041  }
1042 }
1043 
1044 #define ISQRT2 FIXR(0.70710678118654752440)
1045 
1047 {
1048  int i, j, k, l;
1049  int sf_max, sf, len, non_zero_found;
1050  INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1051  int non_zero_found_short[3];
1052 
1053  /* intensity stereo */
1054  if (s->mode_ext & MODE_EXT_I_STEREO) {
1055  if (!s->lsf) {
1056  is_tab = is_table;
1057  sf_max = 7;
1058  } else {
1059  is_tab = is_table_lsf[g1->scalefac_compress & 1];
1060  sf_max = 16;
1061  }
1062 
1063  tab0 = g0->sb_hybrid + 576;
1064  tab1 = g1->sb_hybrid + 576;
1065 
1066  non_zero_found_short[0] = 0;
1067  non_zero_found_short[1] = 0;
1068  non_zero_found_short[2] = 0;
1069  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1070  for (i = 12; i >= g1->short_start; i--) {
1071  /* for last band, use previous scale factor */
1072  if (i != 11)
1073  k -= 3;
1074  len = band_size_short[s->sample_rate_index][i];
1075  for (l = 2; l >= 0; l--) {
1076  tab0 -= len;
1077  tab1 -= len;
1078  if (!non_zero_found_short[l]) {
1079  /* test if non zero band. if so, stop doing i-stereo */
1080  for (j = 0; j < len; j++) {
1081  if (tab1[j] != 0) {
1082  non_zero_found_short[l] = 1;
1083  goto found1;
1084  }
1085  }
1086  sf = g1->scale_factors[k + l];
1087  if (sf >= sf_max)
1088  goto found1;
1089 
1090  v1 = is_tab[0][sf];
1091  v2 = is_tab[1][sf];
1092  for (j = 0; j < len; j++) {
1093  tmp0 = tab0[j];
1094  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1095  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1096  }
1097  } else {
1098 found1:
1099  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1100  /* lower part of the spectrum : do ms stereo
1101  if enabled */
1102  for (j = 0; j < len; j++) {
1103  tmp0 = tab0[j];
1104  tmp1 = tab1[j];
1105  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1106  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1107  }
1108  }
1109  }
1110  }
1111  }
1112 
1113  non_zero_found = non_zero_found_short[0] |
1114  non_zero_found_short[1] |
1115  non_zero_found_short[2];
1116 
1117  for (i = g1->long_end - 1;i >= 0;i--) {
1118  len = band_size_long[s->sample_rate_index][i];
1119  tab0 -= len;
1120  tab1 -= len;
1121  /* test if non zero band. if so, stop doing i-stereo */
1122  if (!non_zero_found) {
1123  for (j = 0; j < len; j++) {
1124  if (tab1[j] != 0) {
1125  non_zero_found = 1;
1126  goto found2;
1127  }
1128  }
1129  /* for last band, use previous scale factor */
1130  k = (i == 21) ? 20 : i;
1131  sf = g1->scale_factors[k];
1132  if (sf >= sf_max)
1133  goto found2;
1134  v1 = is_tab[0][sf];
1135  v2 = is_tab[1][sf];
1136  for (j = 0; j < len; j++) {
1137  tmp0 = tab0[j];
1138  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1139  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1140  }
1141  } else {
1142 found2:
1143  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1144  /* lower part of the spectrum : do ms stereo
1145  if enabled */
1146  for (j = 0; j < len; j++) {
1147  tmp0 = tab0[j];
1148  tmp1 = tab1[j];
1149  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1150  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1151  }
1152  }
1153  }
1154  }
1155  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1156  /* ms stereo ONLY */
1157  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1158  global gain */
1159 #if CONFIG_FLOAT
1160  s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1161 #else
1162  tab0 = g0->sb_hybrid;
1163  tab1 = g1->sb_hybrid;
1164  for (i = 0; i < 576; i++) {
1165  tmp0 = tab0[i];
1166  tmp1 = tab1[i];
1167  tab0[i] = tmp0 + tmp1;
1168  tab1[i] = tmp0 - tmp1;
1169  }
1170 #endif
1171  }
1172 }
1173 
1174 #if CONFIG_FLOAT
1175 #define AA(j) do { \
1176  float tmp0 = ptr[-1-j]; \
1177  float tmp1 = ptr[ j]; \
1178  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1179  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1180  } while (0)
1181 #else
1182 #define AA(j) do { \
1183  int tmp0 = ptr[-1-j]; \
1184  int tmp1 = ptr[ j]; \
1185  int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1186  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1187  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1188  } while (0)
1189 #endif
1190 
1192 {
1193  INTFLOAT *ptr;
1194  int n, i;
1195 
1196  /* we antialias only "long" bands */
1197  if (g->block_type == 2) {
1198  if (!g->switch_point)
1199  return;
1200  /* XXX: check this for 8000Hz case */
1201  n = 1;
1202  } else {
1203  n = SBLIMIT - 1;
1204  }
1205 
1206  ptr = g->sb_hybrid + 18;
1207  for (i = n; i > 0; i--) {
1208  AA(0);
1209  AA(1);
1210  AA(2);
1211  AA(3);
1212  AA(4);
1213  AA(5);
1214  AA(6);
1215  AA(7);
1216 
1217  ptr += 18;
1218  }
1219 }
1220 
1222  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1223 {
1224  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1225  INTFLOAT out2[12];
1226  int i, j, mdct_long_end, sblimit;
1227 
1228  /* find last non zero block */
1229  ptr = g->sb_hybrid + 576;
1230  ptr1 = g->sb_hybrid + 2 * 18;
1231  while (ptr >= ptr1) {
1232  int32_t *p;
1233  ptr -= 6;
1234  p = (int32_t*)ptr;
1235  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1236  break;
1237  }
1238  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1239 
1240  if (g->block_type == 2) {
1241  /* XXX: check for 8000 Hz */
1242  if (g->switch_point)
1243  mdct_long_end = 2;
1244  else
1245  mdct_long_end = 0;
1246  } else {
1247  mdct_long_end = sblimit;
1248  }
1249 
1250  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1251  mdct_long_end, g->switch_point,
1252  g->block_type);
1253 
1254  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1255  ptr = g->sb_hybrid + 18 * mdct_long_end;
1256 
1257  for (j = mdct_long_end; j < sblimit; j++) {
1258  /* select frequency inversion */
1259  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1260  out_ptr = sb_samples + j;
1261 
1262  for (i = 0; i < 6; i++) {
1263  *out_ptr = buf[4*i];
1264  out_ptr += SBLIMIT;
1265  }
1266  imdct12(out2, ptr + 0);
1267  for (i = 0; i < 6; i++) {
1268  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1269  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1270  out_ptr += SBLIMIT;
1271  }
1272  imdct12(out2, ptr + 1);
1273  for (i = 0; i < 6; i++) {
1274  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1275  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1276  out_ptr += SBLIMIT;
1277  }
1278  imdct12(out2, ptr + 2);
1279  for (i = 0; i < 6; i++) {
1280  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1281  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1282  buf[4*(i + 6*2)] = 0;
1283  }
1284  ptr += 18;
1285  buf += (j&3) != 3 ? 1 : (4*18-3);
1286  }
1287  /* zero bands */
1288  for (j = sblimit; j < SBLIMIT; j++) {
1289  /* overlap */
1290  out_ptr = sb_samples + j;
1291  for (i = 0; i < 18; i++) {
1292  *out_ptr = buf[4*i];
1293  buf[4*i] = 0;
1294  out_ptr += SBLIMIT;
1295  }
1296  buf += (j&3) != 3 ? 1 : (4*18-3);
1297  }
1298 }
1299 
1300 /* main layer3 decoding function */
1302 {
1303  int nb_granules, main_data_begin;
1304  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1305  GranuleDef *g;
1306  int16_t exponents[576]; //FIXME try INTFLOAT
1307 
1308  /* read side info */
1309  if (s->lsf) {
1310  main_data_begin = get_bits(&s->gb, 8);
1311  skip_bits(&s->gb, s->nb_channels);
1312  nb_granules = 1;
1313  } else {
1314  main_data_begin = get_bits(&s->gb, 9);
1315  if (s->nb_channels == 2)
1316  skip_bits(&s->gb, 3);
1317  else
1318  skip_bits(&s->gb, 5);
1319  nb_granules = 2;
1320  for (ch = 0; ch < s->nb_channels; ch++) {
1321  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1322  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1323  }
1324  }
1325 
1326  for (gr = 0; gr < nb_granules; gr++) {
1327  for (ch = 0; ch < s->nb_channels; ch++) {
1328  av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1329  g = &s->granules[ch][gr];
1330  g->part2_3_length = get_bits(&s->gb, 12);
1331  g->big_values = get_bits(&s->gb, 9);
1332  if (g->big_values > 288) {
1333  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1334  return AVERROR_INVALIDDATA;
1335  }
1336 
1337  g->global_gain = get_bits(&s->gb, 8);
1338  /* if MS stereo only is selected, we precompute the
1339  1/sqrt(2) renormalization factor */
1340  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1342  g->global_gain -= 2;
1343  if (s->lsf)
1344  g->scalefac_compress = get_bits(&s->gb, 9);
1345  else
1346  g->scalefac_compress = get_bits(&s->gb, 4);
1347  blocksplit_flag = get_bits1(&s->gb);
1348  if (blocksplit_flag) {
1349  g->block_type = get_bits(&s->gb, 2);
1350  if (g->block_type == 0) {
1351  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1352  return AVERROR_INVALIDDATA;
1353  }
1354  g->switch_point = get_bits1(&s->gb);
1355  for (i = 0; i < 2; i++)
1356  g->table_select[i] = get_bits(&s->gb, 5);
1357  for (i = 0; i < 3; i++)
1358  g->subblock_gain[i] = get_bits(&s->gb, 3);
1359  ff_init_short_region(s, g);
1360  } else {
1361  int region_address1, region_address2;
1362  g->block_type = 0;
1363  g->switch_point = 0;
1364  for (i = 0; i < 3; i++)
1365  g->table_select[i] = get_bits(&s->gb, 5);
1366  /* compute huffman coded region sizes */
1367  region_address1 = get_bits(&s->gb, 4);
1368  region_address2 = get_bits(&s->gb, 3);
1369  av_dlog(s->avctx, "region1=%d region2=%d\n",
1370  region_address1, region_address2);
1371  ff_init_long_region(s, g, region_address1, region_address2);
1372  }
1375 
1376  g->preflag = 0;
1377  if (!s->lsf)
1378  g->preflag = get_bits1(&s->gb);
1379  g->scalefac_scale = get_bits1(&s->gb);
1380  g->count1table_select = get_bits1(&s->gb);
1381  av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1382  g->block_type, g->switch_point);
1383  }
1384  }
1385 
1386  if (!s->adu_mode) {
1387  int skip;
1388  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1389  int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0,
1390  FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
1391  assert((get_bits_count(&s->gb) & 7) == 0);
1392  /* now we get bits from the main_data_begin offset */
1393  av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1394  main_data_begin, s->last_buf_size);
1395 
1396  memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
1397  s->in_gb = s->gb;
1398  init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1399 #if !UNCHECKED_BITSTREAM_READER
1400  s->gb.size_in_bits_plus8 += extrasize * 8;
1401 #endif
1402  s->last_buf_size <<= 3;
1403  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1404  for (ch = 0; ch < s->nb_channels; ch++) {
1405  g = &s->granules[ch][gr];
1406  s->last_buf_size += g->part2_3_length;
1407  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1408  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1409  }
1410  }
1411  skip = s->last_buf_size - 8 * main_data_begin;
1412  if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
1413  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
1414  s->gb = s->in_gb;
1415  s->in_gb.buffer = NULL;
1416  } else {
1417  skip_bits_long(&s->gb, skip);
1418  }
1419  } else {
1420  gr = 0;
1421  }
1422 
1423  for (; gr < nb_granules; gr++) {
1424  for (ch = 0; ch < s->nb_channels; ch++) {
1425  g = &s->granules[ch][gr];
1426  bits_pos = get_bits_count(&s->gb);
1427 
1428  if (!s->lsf) {
1429  uint8_t *sc;
1430  int slen, slen1, slen2;
1431 
1432  /* MPEG1 scale factors */
1433  slen1 = slen_table[0][g->scalefac_compress];
1434  slen2 = slen_table[1][g->scalefac_compress];
1435  av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1436  if (g->block_type == 2) {
1437  n = g->switch_point ? 17 : 18;
1438  j = 0;
1439  if (slen1) {
1440  for (i = 0; i < n; i++)
1441  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1442  } else {
1443  for (i = 0; i < n; i++)
1444  g->scale_factors[j++] = 0;
1445  }
1446  if (slen2) {
1447  for (i = 0; i < 18; i++)
1448  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1449  for (i = 0; i < 3; i++)
1450  g->scale_factors[j++] = 0;
1451  } else {
1452  for (i = 0; i < 21; i++)
1453  g->scale_factors[j++] = 0;
1454  }
1455  } else {
1456  sc = s->granules[ch][0].scale_factors;
1457  j = 0;
1458  for (k = 0; k < 4; k++) {
1459  n = k == 0 ? 6 : 5;
1460  if ((g->scfsi & (0x8 >> k)) == 0) {
1461  slen = (k < 2) ? slen1 : slen2;
1462  if (slen) {
1463  for (i = 0; i < n; i++)
1464  g->scale_factors[j++] = get_bits(&s->gb, slen);
1465  } else {
1466  for (i = 0; i < n; i++)
1467  g->scale_factors[j++] = 0;
1468  }
1469  } else {
1470  /* simply copy from last granule */
1471  for (i = 0; i < n; i++) {
1472  g->scale_factors[j] = sc[j];
1473  j++;
1474  }
1475  }
1476  }
1477  g->scale_factors[j++] = 0;
1478  }
1479  } else {
1480  int tindex, tindex2, slen[4], sl, sf;
1481 
1482  /* LSF scale factors */
1483  if (g->block_type == 2)
1484  tindex = g->switch_point ? 2 : 1;
1485  else
1486  tindex = 0;
1487 
1488  sf = g->scalefac_compress;
1489  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1490  /* intensity stereo case */
1491  sf >>= 1;
1492  if (sf < 180) {
1493  lsf_sf_expand(slen, sf, 6, 6, 0);
1494  tindex2 = 3;
1495  } else if (sf < 244) {
1496  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1497  tindex2 = 4;
1498  } else {
1499  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1500  tindex2 = 5;
1501  }
1502  } else {
1503  /* normal case */
1504  if (sf < 400) {
1505  lsf_sf_expand(slen, sf, 5, 4, 4);
1506  tindex2 = 0;
1507  } else if (sf < 500) {
1508  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1509  tindex2 = 1;
1510  } else {
1511  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1512  tindex2 = 2;
1513  g->preflag = 1;
1514  }
1515  }
1516 
1517  j = 0;
1518  for (k = 0; k < 4; k++) {
1519  n = lsf_nsf_table[tindex2][tindex][k];
1520  sl = slen[k];
1521  if (sl) {
1522  for (i = 0; i < n; i++)
1523  g->scale_factors[j++] = get_bits(&s->gb, sl);
1524  } else {
1525  for (i = 0; i < n; i++)
1526  g->scale_factors[j++] = 0;
1527  }
1528  }
1529  /* XXX: should compute exact size */
1530  for (; j < 40; j++)
1531  g->scale_factors[j] = 0;
1532  }
1533 
1534  exponents_from_scale_factors(s, g, exponents);
1535 
1536  /* read Huffman coded residue */
1537  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1538  } /* ch */
1539 
1540  if (s->mode == MPA_JSTEREO)
1541  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1542 
1543  for (ch = 0; ch < s->nb_channels; ch++) {
1544  g = &s->granules[ch][gr];
1545 
1546  reorder_block(s, g);
1547  compute_antialias(s, g);
1548  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1549  }
1550  } /* gr */
1551  if (get_bits_count(&s->gb) < 0)
1552  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1553  return nb_granules * 18;
1554 }
1555 
1557  const uint8_t *buf, int buf_size)
1558 {
1559  int i, nb_frames, ch, ret;
1560  OUT_INT *samples_ptr;
1561 
1562  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1563 
1564  /* skip error protection field */
1565  if (s->error_protection)
1566  skip_bits(&s->gb, 16);
1567 
1568  switch(s->layer) {
1569  case 1:
1570  s->avctx->frame_size = 384;
1571  nb_frames = mp_decode_layer1(s);
1572  break;
1573  case 2:
1574  s->avctx->frame_size = 1152;
1575  nb_frames = mp_decode_layer2(s);
1576  break;
1577  case 3:
1578  s->avctx->frame_size = s->lsf ? 576 : 1152;
1579  default:
1580  nb_frames = mp_decode_layer3(s);
1581 
1582  if (nb_frames < 0)
1583  return nb_frames;
1584 
1585  s->last_buf_size=0;
1586  if (s->in_gb.buffer) {
1587  align_get_bits(&s->gb);
1588  i = get_bits_left(&s->gb)>>3;
1589  if (i >= 0 && i <= BACKSTEP_SIZE) {
1590  memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1591  s->last_buf_size=i;
1592  } else
1593  av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1594  s->gb = s->in_gb;
1595  s->in_gb.buffer = NULL;
1596  }
1597 
1598  align_get_bits(&s->gb);
1599  assert((get_bits_count(&s->gb) & 7) == 0);
1600  i = get_bits_left(&s->gb) >> 3;
1601 
1602  if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1603  if (i < 0)
1604  av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1605  i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1606  }
1607  assert(i <= buf_size - HEADER_SIZE && i >= 0);
1608  memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1609  s->last_buf_size += i;
1610  }
1611 
1612  /* get output buffer */
1613  if (!samples) {
1614  s->frame.nb_samples = s->avctx->frame_size;
1615  if ((ret = ff_get_buffer(s->avctx, &s->frame)) < 0) {
1616  av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1617  return ret;
1618  }
1619  samples = (OUT_INT **)s->frame.extended_data;
1620  }
1621 
1622  /* apply the synthesis filter */
1623  for (ch = 0; ch < s->nb_channels; ch++) {
1624  int sample_stride;
1625  if (s->avctx->sample_fmt == OUT_FMT_P) {
1626  samples_ptr = samples[ch];
1627  sample_stride = 1;
1628  } else {
1629  samples_ptr = samples[0] + ch;
1630  sample_stride = s->nb_channels;
1631  }
1632  for (i = 0; i < nb_frames; i++) {
1634  &(s->synth_buf_offset[ch]),
1635  RENAME(ff_mpa_synth_window),
1636  &s->dither_state, samples_ptr,
1637  sample_stride, s->sb_samples[ch][i]);
1638  samples_ptr += 32 * sample_stride;
1639  }
1640  }
1641 
1642  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1643 }
1644 
1645 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1646  AVPacket *avpkt)
1647 {
1648  const uint8_t *buf = avpkt->data;
1649  int buf_size = avpkt->size;
1650  MPADecodeContext *s = avctx->priv_data;
1651  uint32_t header;
1652  int ret;
1653 
1654  if (buf_size < HEADER_SIZE)
1655  return AVERROR_INVALIDDATA;
1656 
1657  header = AV_RB32(buf);
1658  if (ff_mpa_check_header(header) < 0) {
1659  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1660  return AVERROR_INVALIDDATA;
1661  }
1662 
1663  if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1664  /* free format: prepare to compute frame size */
1665  s->frame_size = -1;
1666  return AVERROR_INVALIDDATA;
1667  }
1668  /* update codec info */
1669  avctx->channels = s->nb_channels;
1670  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1671  if (!avctx->bit_rate)
1672  avctx->bit_rate = s->bit_rate;
1673 
1674  if (s->frame_size <= 0 || s->frame_size > buf_size) {
1675  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1676  return AVERROR_INVALIDDATA;
1677  } else if (s->frame_size < buf_size) {
1678  buf_size= s->frame_size;
1679  }
1680 
1681  ret = mp_decode_frame(s, NULL, buf, buf_size);
1682  if (ret >= 0) {
1683  *got_frame_ptr = 1;
1684  *(AVFrame *)data = s->frame;
1685  avctx->sample_rate = s->sample_rate;
1686  //FIXME maybe move the other codec info stuff from above here too
1687  } else {
1688  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1689  /* Only return an error if the bad frame makes up the whole packet or
1690  * the error is related to buffer management.
1691  * If there is more data in the packet, just consume the bad frame
1692  * instead of returning an error, which would discard the whole
1693  * packet. */
1694  *got_frame_ptr = 0;
1695  if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1696  return ret;
1697  }
1698  s->frame_size = 0;
1699  return buf_size;
1700 }
1701 
1702 static void mp_flush(MPADecodeContext *ctx)
1703 {
1704  memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1705  ctx->last_buf_size = 0;
1706 }
1707 
1708 static void flush(AVCodecContext *avctx)
1709 {
1710  mp_flush(avctx->priv_data);
1711 }
1712 
1713 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1714 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1715  int *got_frame_ptr, AVPacket *avpkt)
1716 {
1717  const uint8_t *buf = avpkt->data;
1718  int buf_size = avpkt->size;
1719  MPADecodeContext *s = avctx->priv_data;
1720  uint32_t header;
1721  int len, ret;
1722 
1723  len = buf_size;
1724 
1725  // Discard too short frames
1726  if (buf_size < HEADER_SIZE) {
1727  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1728  return AVERROR_INVALIDDATA;
1729  }
1730 
1731 
1732  if (len > MPA_MAX_CODED_FRAME_SIZE)
1734 
1735  // Get header and restore sync word
1736  header = AV_RB32(buf) | 0xffe00000;
1737 
1738  if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
1739  av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1740  return AVERROR_INVALIDDATA;
1741  }
1742 
1744  /* update codec info */
1745  avctx->sample_rate = s->sample_rate;
1746  avctx->channels = s->nb_channels;
1747  if (!avctx->bit_rate)
1748  avctx->bit_rate = s->bit_rate;
1749 
1750  s->frame_size = len;
1751 
1752  ret = mp_decode_frame(s, NULL, buf, buf_size);
1753  if (ret < 0) {
1754  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1755  return ret;
1756  }
1757 
1758  *got_frame_ptr = 1;
1759  *(AVFrame *)data = s->frame;
1760 
1761  return buf_size;
1762 }
1763 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1764 
1765 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1766 
1770 typedef struct MP3On4DecodeContext {
1771  AVFrame *frame;
1772  int frames;
1773  int syncword;
1774  const uint8_t *coff;
1775  MPADecodeContext *mp3decctx[5];
1776 } MP3On4DecodeContext;
1777 
1778 #include "mpeg4audio.h"
1779 
1780 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1781 
1782 /* number of mp3 decoder instances */
1783 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1784 
1785 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1786 static const uint8_t chan_offset[8][5] = {
1787  { 0 },
1788  { 0 }, // C
1789  { 0 }, // FLR
1790  { 2, 0 }, // C FLR
1791  { 2, 0, 3 }, // C FLR BS
1792  { 2, 0, 3 }, // C FLR BLRS
1793  { 2, 0, 4, 3 }, // C FLR BLRS LFE
1794  { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1795 };
1796 
1797 /* mp3on4 channel layouts */
1798 static const int16_t chan_layout[8] = {
1799  0,
1807 };
1808 
1809 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1810 {
1811  MP3On4DecodeContext *s = avctx->priv_data;
1812  int i;
1813 
1814  for (i = 0; i < s->frames; i++)
1815  av_free(s->mp3decctx[i]);
1816 
1817  return 0;
1818 }
1819 
1820 
1821 static int decode_init_mp3on4(AVCodecContext * avctx)
1822 {
1823  MP3On4DecodeContext *s = avctx->priv_data;
1824  MPEG4AudioConfig cfg;
1825  int i;
1826 
1827  if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
1828  av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1829  return AVERROR_INVALIDDATA;
1830  }
1831 
1833  avctx->extradata_size * 8, 1);
1834  if (!cfg.chan_config || cfg.chan_config > 7) {
1835  av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1836  return AVERROR_INVALIDDATA;
1837  }
1838  s->frames = mp3Frames[cfg.chan_config];
1839  s->coff = chan_offset[cfg.chan_config];
1841  avctx->channel_layout = chan_layout[cfg.chan_config];
1842 
1843  if (cfg.sample_rate < 16000)
1844  s->syncword = 0xffe00000;
1845  else
1846  s->syncword = 0xfff00000;
1847 
1848  /* Init the first mp3 decoder in standard way, so that all tables get builded
1849  * We replace avctx->priv_data with the context of the first decoder so that
1850  * decode_init() does not have to be changed.
1851  * Other decoders will be initialized here copying data from the first context
1852  */
1853  // Allocate zeroed memory for the first decoder context
1854  s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1855  if (!s->mp3decctx[0])
1856  goto alloc_fail;
1857  // Put decoder context in place to make init_decode() happy
1858  avctx->priv_data = s->mp3decctx[0];
1859  decode_init(avctx);
1860  s->frame = avctx->coded_frame;
1861  // Restore mp3on4 context pointer
1862  avctx->priv_data = s;
1863  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1864 
1865  /* Create a separate codec/context for each frame (first is already ok).
1866  * Each frame is 1 or 2 channels - up to 5 frames allowed
1867  */
1868  for (i = 1; i < s->frames; i++) {
1869  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1870  if (!s->mp3decctx[i])
1871  goto alloc_fail;
1872  s->mp3decctx[i]->adu_mode = 1;
1873  s->mp3decctx[i]->avctx = avctx;
1874  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1875  }
1876 
1877  return 0;
1878 alloc_fail:
1879  decode_close_mp3on4(avctx);
1880  return AVERROR(ENOMEM);
1881 }
1882 
1883 
1884 static void flush_mp3on4(AVCodecContext *avctx)
1885 {
1886  int i;
1887  MP3On4DecodeContext *s = avctx->priv_data;
1888 
1889  for (i = 0; i < s->frames; i++)
1890  mp_flush(s->mp3decctx[i]);
1891 }
1892 
1893 
1894 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1895  int *got_frame_ptr, AVPacket *avpkt)
1896 {
1897  const uint8_t *buf = avpkt->data;
1898  int buf_size = avpkt->size;
1899  MP3On4DecodeContext *s = avctx->priv_data;
1900  MPADecodeContext *m;
1901  int fsize, len = buf_size, out_size = 0;
1902  uint32_t header;
1903  OUT_INT **out_samples;
1904  OUT_INT *outptr[2];
1905  int fr, ch, ret;
1906 
1907  /* get output buffer */
1908  s->frame->nb_samples = MPA_FRAME_SIZE;
1909  if ((ret = ff_get_buffer(avctx, s->frame)) < 0) {
1910  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1911  return ret;
1912  }
1913  out_samples = (OUT_INT **)s->frame->extended_data;
1914 
1915  // Discard too short frames
1916  if (buf_size < HEADER_SIZE)
1917  return AVERROR_INVALIDDATA;
1918 
1919  avctx->bit_rate = 0;
1920 
1921  ch = 0;
1922  for (fr = 0; fr < s->frames; fr++) {
1923  fsize = AV_RB16(buf) >> 4;
1924  fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1925  m = s->mp3decctx[fr];
1926  assert(m != NULL);
1927 
1928  if (fsize < HEADER_SIZE) {
1929  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1930  return AVERROR_INVALIDDATA;
1931  }
1932  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1933 
1934  if (ff_mpa_check_header(header) < 0) // Bad header, discard block
1935  break;
1936 
1938 
1939  if (ch + m->nb_channels > avctx->channels ||
1940  s->coff[fr] + m->nb_channels > avctx->channels) {
1941  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1942  "channel count\n");
1943  return AVERROR_INVALIDDATA;
1944  }
1945  ch += m->nb_channels;
1946 
1947  outptr[0] = out_samples[s->coff[fr]];
1948  if (m->nb_channels > 1)
1949  outptr[1] = out_samples[s->coff[fr] + 1];
1950 
1951  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
1952  return ret;
1953 
1954  out_size += ret;
1955  buf += fsize;
1956  len -= fsize;
1957 
1958  avctx->bit_rate += m->bit_rate;
1959  }
1960 
1961  /* update codec info */
1962  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1963 
1964  s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1965  *got_frame_ptr = 1;
1966  *(AVFrame *)data = *s->frame;
1967 
1968  return buf_size;
1969 }
1970 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
1971 
1972 #if !CONFIG_FLOAT
1973 #if CONFIG_MP1_DECODER
1974 AVCodec ff_mp1_decoder = {
1975  .name = "mp1",
1976  .type = AVMEDIA_TYPE_AUDIO,
1977  .id = AV_CODEC_ID_MP1,
1978  .priv_data_size = sizeof(MPADecodeContext),
1979  .init = decode_init,
1980  .decode = decode_frame,
1981  .capabilities = CODEC_CAP_DR1,
1982  .flush = flush,
1983  .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
1984  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
1987 };
1988 #endif
1989 #if CONFIG_MP2_DECODER
1990 AVCodec ff_mp2_decoder = {
1991  .name = "mp2",
1992  .type = AVMEDIA_TYPE_AUDIO,
1993  .id = AV_CODEC_ID_MP2,
1994  .priv_data_size = sizeof(MPADecodeContext),
1995  .init = decode_init,
1996  .decode = decode_frame,
1997  .capabilities = CODEC_CAP_DR1,
1998  .flush = flush,
1999  .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2000  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2003 };
2004 #endif
2005 #if CONFIG_MP3_DECODER
2006 AVCodec ff_mp3_decoder = {
2007  .name = "mp3",
2008  .type = AVMEDIA_TYPE_AUDIO,
2009  .id = AV_CODEC_ID_MP3,
2010  .priv_data_size = sizeof(MPADecodeContext),
2011  .init = decode_init,
2012  .decode = decode_frame,
2013  .capabilities = CODEC_CAP_DR1,
2014  .flush = flush,
2015  .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2016  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2019 };
2020 #endif
2021 #if CONFIG_MP3ADU_DECODER
2022 AVCodec ff_mp3adu_decoder = {
2023  .name = "mp3adu",
2024  .type = AVMEDIA_TYPE_AUDIO,
2025  .id = AV_CODEC_ID_MP3ADU,
2026  .priv_data_size = sizeof(MPADecodeContext),
2027  .init = decode_init,
2028  .decode = decode_frame_adu,
2029  .capabilities = CODEC_CAP_DR1,
2030  .flush = flush,
2031  .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2032  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2035 };
2036 #endif
2037 #if CONFIG_MP3ON4_DECODER
2038 AVCodec ff_mp3on4_decoder = {
2039  .name = "mp3on4",
2040  .type = AVMEDIA_TYPE_AUDIO,
2041  .id = AV_CODEC_ID_MP3ON4,
2042  .priv_data_size = sizeof(MP3On4DecodeContext),
2043  .init = decode_init_mp3on4,
2044  .close = decode_close_mp3on4,
2045  .decode = decode_frame_mp3on4,
2046  .capabilities = CODEC_CAP_DR1,
2047  .flush = flush_mp3on4,
2048  .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
2049  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2051 };
2052 #endif
2053 #endif
#define MUL64(a, b)
Definition: mathops.h:51
#define BACKSTEP_SIZE
Definition: mpegaudiodec.c:43
#define MPA_MAX_CODED_FRAME_SIZE
Definition: mpegaudio.h:39
#define AV_CH_LAYOUT_7POINT1
av_cold void ff_dsputil_init(DSPContext *c, AVCodecContext *avctx)
Definition: dsputil.c:2656
static uint32_t table_4_3_value[TABLE_4_3_SIZE]
static const uint8_t lsf_nsf_table[6][3][4]
static int16_t * samples
#define SBLIMIT
Definition: mpegaudio.h:43
#define FIXR(a)
Definition: mpegaudiodec.c:103
This structure describes decoded (raw) audio or video data.
Definition: avcodec.h:989
static int l1_unscale(int n, int mant, int scale_factor)
Definition: mpegaudiodec.c:230
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:237
#define AV_CH_LAYOUT_SURROUND
#define C4
Definition: mpegaudiodec.c:457
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:197
AVFrame * coded_frame
the picture in the bitstream
Definition: avcodec.h:2725
#define READ_FLIP_SIGN(dst, src)
Definition: mpegaudiodec.c:855
static int8_t table_4_3_exp[TABLE_4_3_SIZE]
static void align_get_bits(GetBitContext *s)
Definition: get_bits.h:412
#define MPA_JSTEREO
Definition: mpegaudio.h:46
#define OUT_FMT
Definition: mpegaudiodec.c:108
static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
int size
Definition: avcodec.h:916
struct MPADecodeContext MPADecodeContext
#define ISQRT2
const uint8_t * buffer
Definition: get_bits.h:53
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:55
#define AV_CH_LAYOUT_4POINT0
#define VLC_TYPE
Definition: get_bits.h:61
int subblock_gain[3]
Definition: mpegaudiodec.c:57
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
#define AA(j)
#define AV_CH_LAYOUT_STEREO
#define C6
Definition: mpegaudiodec.c:459
static av_cold void decode_init_static(void)
Definition: mpegaudiodec.c:276
uint8_t scale_factors[40]
Definition: mpegaudiodec.c:63
signed 16 bits
Definition: samplefmt.h:52
AVCodec.
Definition: avcodec.h:2960
#define AV_CH_LAYOUT_5POINT0
mpeg audio layer common tables.
static const int huff_vlc_tables_sizes[16]
Definition: mpegaudiodec.c:125
static const uint8_t slen_table[2][16]
int scalefac_compress
Definition: mpegaudiodec.c:53
int32_t MPA_INT
Definition: mpegaudio.h:69
static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2, int n3)
Definition: mpegaudiodec.c:777
static int16_t division_tab5[1<< 8]
Definition: mpegaudiodec.c:141
int16_t OUT_INT
Definition: mpegaudio.h:70
#define SHR(a, b)
Definition: mpegaudiodec.c:100
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
Definition: mimic.c:228
uint8_t bits
Definition: crc.c:31
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2112
uint8_t
GetBitContext in_gb
Definition: mpegaudiodec.c:74
static av_cold int decode_init(AVCodecContext *avctx)
Definition: mpegaudiodec.c:425
static int mp_decode_layer2(MPADecodeContext *s)
Definition: mpegaudiodec.c:570
static int32_t scale_factor_mult[15][3]
Definition: mpegaudiodec.c:151
#define SCALE_GEN(v)
Definition: mpegaudiodec.c:154
uint8_t switch_point
Definition: mpegaudiodec.c:55
av_cold void RENAME() ff_mpa_synth_init(MPA_INT *window)
const int ff_mpa_quant_steps[17]
Definition: mpegaudiodata.c:47
#define AV_RB32
Definition: intreadwrite.h:130
#define b
Definition: input.c:52
static 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:1454
static const uint8_t mpa_huff_data[32][2]
const char data[16]
Definition: mxf.c:66
static const uint8_t mpa_quad_codes[2][16]
uint8_t * data
Definition: avcodec.h:915
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:192
bitstream reader API header.
GetBitContext gb
Definition: mpegaudiodec.c:73
#define MULH3(x, y, s)
Definition: mpegaudiodec.c:105
static int bit_alloc(AC3EncodeContext *s, int snr_offset)
Run the bit allocation with a given SNR offset.
Definition: ac3enc.c:1062
AVCodecContext * avctx
Definition: mpegaudiodec.c:83
static int init(AVCodecParserContext *s)
Definition: h264_parser.c:335
static int16_t division_tab3[1<< 6]
Definition: mpegaudiodec.c:140
#define AV_CH_LAYOUT_5POINT1
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:547
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
Definition: mem.c:139
static INTFLOAT csa_table[8][4]
Definition: mpegaudiodec.c:138
#define MODE_EXT_MS_STEREO
Definition: mpegaudiodata.h:34
#define AV_RB16
Definition: intreadwrite.h:53
static VLC_TYPE huff_quad_vlc_tables[128+16][2]
Definition: mpegaudiodec.c:130
#define HEADER_SIZE
Definition: mpegaudiodec.c:114
enum AVSampleFormat request_sample_fmt
Used to request a sample format from the decoder.
Definition: avcodec.h:2186
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:88
g
Definition: yuv2rgb.c:540
#define t1
Definition: regdef.h:29
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:348
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define C5
Definition: mpegaudiodec.c:458
static int mp_decode_layer1(MPADecodeContext *s)
Definition: mpegaudiodec.c:505
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:146
const char * name
Name of the codec implementation.
Definition: avcodec.h:2967
static int ff_mpa_check_header(uint32_t header)
static const int32_t scale_factor_mult2[3][3]
Definition: mpegaudiodec.c:157
Definition: get_bits.h:63
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2165
void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:26
static int huffman_decode(MPADecodeContext *s, GranuleDef *g, int16_t *exponents, int end_pos2)
Definition: mpegaudiodec.c:860
#define LAST_BUF_SIZE
Definition: mpegaudiodec.c:45
#define INTFLOAT
Definition: dct32.c:34
#define MPA_MAX_CHANNELS
Definition: mpegaudio.h:41
int bit_rate
the average bitrate
Definition: avcodec.h:1404
audio channel layout utility functions
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:2602
static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2)
Definition: mpegaudiodec.c:830
static int mp_decode_layer3(MPADecodeContext *s)
static int16_t division_tab9[1<< 11]
Definition: mpegaudiodec.c:142
#define MULLx(x, y, s)
Definition: mpegaudiodec.c:106
#define C3
Definition: mpegaudiodec.c:456
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame)
Get a buffer for a frame.
Definition: utils.c:464
uint32_t free_format_next_header
Definition: mpegaudiodec.c:72
int size_in_bits
Definition: get_bits.h:55
int32_t
int table_select[3]
Definition: mpegaudiodec.c:56
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:515
static int16_t *const division_tabs[4]
Definition: mpegaudiodec.c:144
static int l2_unscale_group(int steps, int mant, int scale_factor)
Definition: mpegaudiodec.c:244
int big_values
Definition: mpegaudiodec.c:51
static int l3_unscale(int value, int exponent)
Definition: mpegaudiodec.c:260
MPADSPContext mpadsp
Definition: mpegaudiodec.c:84
#define OUT_FMT_P
Definition: mpegaudiodec.c:109
#define INIT_VLC_USE_NEW_STATIC
Definition: get_bits.h:433
int bits
Definition: get_bits.h:64
static const uint8_t mpa_quad_bits[2][16]
int table_allocated
Definition: get_bits.h:66
static uint16_t band_index_long[9][23]
Definition: mpegaudiodec.c:133
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2124
NULL
Definition: eval.c:52
const uint8_t * bits
static const int huff_quad_vlc_tables_sizes[2]
Definition: mpegaudiodec.c:131
static INTFLOAT is_table_lsf[2][2][16]
Definition: mpegaudiodec.c:137
external API header
int sb_hybrid[SBLIMIT *18]
Definition: mpegaudiodec.c:64
enum AVCodecID codec_id
Definition: avcodec.h:1350
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:63
int sample_rate
samples per second
Definition: avcodec.h:2104
MPA_INT synth_buf[MPA_MAX_CHANNELS][512 *2]
Definition: mpegaudiodec.c:75
#define RENAME(a)
Definition: mpegaudiodec.c:107
main external API structure.
Definition: avcodec.h:1339
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:326
uint8_t scalefac_scale
Definition: mpegaudiodec.c:58
int global_gain
Definition: mpegaudiodec.c:52
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
Definition: get_bits.h:418
const int16_t * tab1
Definition: mace.c:144
int extradata_size
Definition: avcodec.h:1455
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:268
void avcodec_get_frame_defaults(AVFrame *frame)
Set the fields of the given AVFrame to default values.
Definition: utils.c:604
uint8_t count1table_select
Definition: mpegaudiodec.c:59
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:260
static void ff_region_offset2size(GranuleDef *g)
Convert region offsets to region sizes and truncate size to big_values.
Definition: mpegaudiodec.c:167
static void compute_imdct(MPADecodeContext *s, GranuleDef *g, INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
int short_start
Definition: mpegaudiodec.c:62
static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples, const uint8_t *buf, int buf_size)
#define MODE_EXT_I_STEREO
Definition: mpegaudiodata.h:35
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:372
const uint16_t * codes
static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g, int16_t *exponents)
Definition: mpegaudiodec.c:786
int part2_3_length
Definition: mpegaudiodec.c:50
#define FRAC_BITS
Definition: lsp.c:27
static const uint8_t band_size_long[9][22]
#define MPA_DECODE_HEADER
static const uint16_t scale[4]
static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
#define FIXHR(a)
Definition: mpegaudiodec.c:104
#define v0
Definition: regdef.h:26
#define SPLIT(dst, sf, n)
Definition: mpegaudiodec.c:757
MPEG Audio header decoder.
static VLC huff_quad_vlc[2]
Definition: mpegaudiodec.c:129
static VLC huff_vlc[16]
Definition: mpegaudiodec.c:120
common internal api header.
static uint16_t scale_factor_modshift[64]
Definition: mpegaudiodec.c:149
static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
Definition: mpegaudiodec.c:196
static INTFLOAT is_table[2][16]
Definition: mpegaudiodec.c:136
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
Definition: mpegaudiodec.c:205
void RENAME() ff_mpa_synth_filter(MPADSPContext *s, MPA_INT *synth_buf_ptr, int *synth_buf_offset, MPA_INT *window, int *dither_state, OUT_INT *samples, int incr, MPA_INT *sb_samples)
mpeg audio declarations for both encoder and decoder.
static void imdct12(INTFLOAT *out, INTFLOAT *in)
Definition: mpegaudiodec.c:463
static void reorder_block(MPADecodeContext *s, GranuleDef *g)
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 to retrieve audio configuration.
Definition: mpeg4audio.c:79
DSP utils.
INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT *18]
Definition: mpegaudiodec.c:78
void * priv_data
Definition: avcodec.h:1382
DSPContext dsp
Definition: mpegaudiodec.c:85
uint8_t block_type
Definition: mpegaudiodec.c:54
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:2105
const uint8_t ff_mpeg4audio_channels[8]
Definition: mpeg4audio.c:60
MPA_DECODE_HEADER uint8_t last_buf[LAST_BUF_SIZE]
Definition: mpegaudiodec.c:69
VLC_TYPE(* table)[2]
code, bits
Definition: get_bits.h:65
static VLC_TYPE huff_vlc_tables[0+128+128+128+130+128+154+166+142+204+190+170+542+460+662+414][2]
Definition: mpegaudiodec.c:124
struct GranuleDef GranuleDef
int synth_buf_offset[MPA_MAX_CHANNELS]
Definition: mpegaudiodec.c:76
signed 16 bits, planar
Definition: samplefmt.h:58
int region_size[3]
Definition: mpegaudiodec.c:60
static int get_bitsz(GetBitContext *s, int n)
Definition: mpegaudiodec.c:824
mpeg audio layer decoder tables.
int sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]
Definition: mpegaudiodec.c:77
static const HuffTable mpa_huff_tables[16]
static void flush(AVCodecContext *avctx)
static const float ci_table[8]
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: avcodec.h:1028
#define AV_CH_LAYOUT_MONO
static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)
Definition: mpegaudiodec.c:178
#define MPA_FRAME_SIZE
Definition: mpegaudio.h:36
This structure stores compressed data.
Definition: avcodec.h:898
int nb_samples
number of audio samples (per channel) described by this frame
Definition: avcodec.h:1042
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:158
for(j=16;j >0;--j)
#define t2
Definition: regdef.h:30
static int alloc_table(VLC *vlc, int size, int use_static)
Definition: bitstream.c:100
DSPContext.
Definition: dsputil.h:194
static const uint8_t band_size_short[9][13]
int adu_mode
0 for standard mp3, 1 for adu formatted mp3
Definition: mpegaudiodec.c:80
static void mp_flush(MPADecodeContext *ctx)
GranuleDef granules[2][2]
Definition: mpegaudiodec.c:79
uint8_t scfsi
Definition: mpegaudiodec.c:49
if(!(ptr_align%ac->ptr_align)&&samples_align >=aligned_len)