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mpegaudiodec.c
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1 /*
2  * MPEG Audio decoder
3  * Copyright (c) 2001, 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * MPEG Audio decoder
25  */
26 
27 #include "libavutil/attributes.h"
28 #include "libavutil/avassert.h"
29 #include "libavutil/channel_layout.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 {
52  uint8_t scfsi;
53  int part2_3_length;
54  int big_values;
55  int global_gain;
56  int scalefac_compress;
57  uint8_t block_type;
58  uint8_t switch_point;
59  int table_select[3];
60  int subblock_gain[3];
61  uint8_t scalefac_scale;
62  uint8_t count1table_select;
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 */
66  uint8_t scale_factors[40];
67  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
68 } GranuleDef;
69 
70 typedef struct MPADecodeContext {
71  MPA_DECODE_HEADER
72  uint8_t last_buf[LAST_BUF_SIZE];
73  int last_buf_size;
74  /* next header (used in free format parsing) */
75  uint32_t free_format_next_header;
76  GetBitContext gb;
77  GetBitContext in_gb;
78  DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
79  int synth_buf_offset[MPA_MAX_CHANNELS];
80  DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
81  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
82  GranuleDef granules[2][2]; /* Used in Layer 3 */
83  int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
84  int dither_state;
85  int err_recognition;
86  AVCodecContext* avctx;
87  MPADSPContext mpadsp;
88  AVFloatDSPContext fdsp;
89  AVFrame *frame;
90 } MPADecodeContext;
91 
92 #if CONFIG_FLOAT
93 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
94 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
95 # define FIXR(x) ((float)(x))
96 # define FIXHR(x) ((float)(x))
97 # define MULH3(x, y, s) ((s)*(y)*(x))
98 # define MULLx(x, y, s) ((y)*(x))
99 # define RENAME(a) a ## _float
100 # define OUT_FMT AV_SAMPLE_FMT_FLT
101 # define OUT_FMT_P AV_SAMPLE_FMT_FLTP
102 #else
103 # define SHR(a,b) ((a)>>(b))
104 /* WARNING: only correct for positive numbers */
105 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
106 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
107 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
108 # define MULH3(x, y, s) MULH((s)*(x), y)
109 # define MULLx(x, y, s) MULL(x,y,s)
110 # define RENAME(a) a ## _fixed
111 # define OUT_FMT AV_SAMPLE_FMT_S16
112 # define OUT_FMT_P AV_SAMPLE_FMT_S16P
113 #endif
114 
115 /****************/
116 
117 #define HEADER_SIZE 4
118 
119 #include "mpegaudiodata.h"
120 #include "mpegaudiodectab.h"
121 
122 /* vlc structure for decoding layer 3 huffman tables */
123 static VLC huff_vlc[16];
124 static VLC_TYPE huff_vlc_tables[
125  0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
126  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
127  ][2];
128 static const int huff_vlc_tables_sizes[16] = {
129  0, 128, 128, 128, 130, 128, 154, 166,
130  142, 204, 190, 170, 542, 460, 662, 414
131 };
132 static VLC huff_quad_vlc[2];
133 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
134 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
135 /* computed from band_size_long */
136 static uint16_t band_index_long[9][23];
137 #include "mpegaudio_tablegen.h"
138 /* intensity stereo coef table */
139 static INTFLOAT is_table[2][16];
140 static INTFLOAT is_table_lsf[2][2][16];
141 static INTFLOAT csa_table[8][4];
142 
143 static int16_t division_tab3[1<<6 ];
144 static int16_t division_tab5[1<<8 ];
145 static int16_t division_tab9[1<<11];
146 
147 static int16_t * const division_tabs[4] = {
148  division_tab3, division_tab5, NULL, division_tab9
149 };
150 
151 /* lower 2 bits: modulo 3, higher bits: shift */
152 static uint16_t scale_factor_modshift[64];
153 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
154 static int32_t scale_factor_mult[15][3];
155 /* mult table for layer 2 group quantization */
156 
157 #define SCALE_GEN(v) \
158 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
159 
160 static const int32_t scale_factor_mult2[3][3] = {
161  SCALE_GEN(4.0 / 3.0), /* 3 steps */
162  SCALE_GEN(4.0 / 5.0), /* 5 steps */
163  SCALE_GEN(4.0 / 9.0), /* 9 steps */
164 };
165 
166 /**
167  * Convert region offsets to region sizes and truncate
168  * size to big_values.
169  */
170 static void region_offset2size(GranuleDef *g)
171 {
172  int i, k, j = 0;
173  g->region_size[2] = 576 / 2;
174  for (i = 0; i < 3; i++) {
175  k = FFMIN(g->region_size[i], g->big_values);
176  g->region_size[i] = k - j;
177  j = k;
178  }
179 }
180 
181 static void init_short_region(MPADecodeContext *s, GranuleDef *g)
182 {
183  if (g->block_type == 2) {
184  if (s->sample_rate_index != 8)
185  g->region_size[0] = (36 / 2);
186  else
187  g->region_size[0] = (72 / 2);
188  } else {
189  if (s->sample_rate_index <= 2)
190  g->region_size[0] = (36 / 2);
191  else if (s->sample_rate_index != 8)
192  g->region_size[0] = (54 / 2);
193  else
194  g->region_size[0] = (108 / 2);
195  }
196  g->region_size[1] = (576 / 2);
197 }
198 
199 static void init_long_region(MPADecodeContext *s, GranuleDef *g,
200  int ra1, int ra2)
201 {
202  int l;
203  g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
204  /* should not overflow */
205  l = FFMIN(ra1 + ra2 + 2, 22);
206  g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
207 }
208 
209 static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
210 {
211  if (g->block_type == 2) {
212  if (g->switch_point) {
213  if(s->sample_rate_index == 8)
214  avpriv_request_sample(s->avctx, "switch point in 8khz");
215  /* if switched mode, we handle the 36 first samples as
216  long blocks. For 8000Hz, we handle the 72 first
217  exponents as long blocks */
218  if (s->sample_rate_index <= 2)
219  g->long_end = 8;
220  else
221  g->long_end = 6;
222 
223  g->short_start = 3;
224  } else {
225  g->long_end = 0;
226  g->short_start = 0;
227  }
228  } else {
229  g->short_start = 13;
230  g->long_end = 22;
231  }
232 }
233 
234 /* layer 1 unscaling */
235 /* n = number of bits of the mantissa minus 1 */
236 static inline int l1_unscale(int n, int mant, int scale_factor)
237 {
238  int shift, mod;
239  int64_t val;
240 
241  shift = scale_factor_modshift[scale_factor];
242  mod = shift & 3;
243  shift >>= 2;
244  val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
245  shift += n;
246  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
247  return (int)((val + (1LL << (shift - 1))) >> shift);
248 }
249 
250 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
251 {
252  int shift, mod, val;
253 
254  shift = scale_factor_modshift[scale_factor];
255  mod = shift & 3;
256  shift >>= 2;
257 
258  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
259  /* NOTE: at this point, 0 <= shift <= 21 */
260  if (shift > 0)
261  val = (val + (1 << (shift - 1))) >> shift;
262  return val;
263 }
264 
265 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
266 static inline int l3_unscale(int value, int exponent)
267 {
268  unsigned int m;
269  int e;
270 
271  e = table_4_3_exp [4 * value + (exponent & 3)];
272  m = table_4_3_value[4 * value + (exponent & 3)];
273  e -= exponent >> 2;
274 #ifdef DEBUG
275  if(e < 1)
276  av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
277 #endif
278  if (e > 31)
279  return 0;
280  m = (m + (1 << (e - 1))) >> e;
281 
282  return m;
283 }
284 
285 static av_cold void decode_init_static(void)
286 {
287  int i, j, k;
288  int offset;
289 
290  /* scale factors table for layer 1/2 */
291  for (i = 0; i < 64; i++) {
292  int shift, mod;
293  /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
294  shift = i / 3;
295  mod = i % 3;
296  scale_factor_modshift[i] = mod | (shift << 2);
297  }
298 
299  /* scale factor multiply for layer 1 */
300  for (i = 0; i < 15; i++) {
301  int n, norm;
302  n = i + 2;
303  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
304  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
305  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
306  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
307  av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
308  scale_factor_mult[i][0],
309  scale_factor_mult[i][1],
310  scale_factor_mult[i][2]);
311  }
312 
313  RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
314 
315  /* huffman decode tables */
316  offset = 0;
317  for (i = 1; i < 16; i++) {
318  const HuffTable *h = &mpa_huff_tables[i];
319  int xsize, x, y;
320  uint8_t tmp_bits [512] = { 0 };
321  uint16_t tmp_codes[512] = { 0 };
322 
323  xsize = h->xsize;
324 
325  j = 0;
326  for (x = 0; x < xsize; x++) {
327  for (y = 0; y < xsize; y++) {
328  tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
329  tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
330  }
331  }
332 
333  /* XXX: fail test */
334  huff_vlc[i].table = huff_vlc_tables+offset;
335  huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
336  init_vlc(&huff_vlc[i], 7, 512,
337  tmp_bits, 1, 1, tmp_codes, 2, 2,
338  INIT_VLC_USE_NEW_STATIC);
339  offset += huff_vlc_tables_sizes[i];
340  }
341  av_assert0(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
342 
343  offset = 0;
344  for (i = 0; i < 2; i++) {
345  huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
346  huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
347  init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
348  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
349  INIT_VLC_USE_NEW_STATIC);
350  offset += huff_quad_vlc_tables_sizes[i];
351  }
352  av_assert0(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
353 
354  for (i = 0; i < 9; i++) {
355  k = 0;
356  for (j = 0; j < 22; j++) {
357  band_index_long[i][j] = k;
358  k += band_size_long[i][j];
359  }
360  band_index_long[i][22] = k;
361  }
362 
363  /* compute n ^ (4/3) and store it in mantissa/exp format */
364 
365  mpegaudio_tableinit();
366 
367  for (i = 0; i < 4; i++) {
368  if (ff_mpa_quant_bits[i] < 0) {
369  for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
370  int val1, val2, val3, steps;
371  int val = j;
372  steps = ff_mpa_quant_steps[i];
373  val1 = val % steps;
374  val /= steps;
375  val2 = val % steps;
376  val3 = val / steps;
377  division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
378  }
379  }
380  }
381 
382 
383  for (i = 0; i < 7; i++) {
384  float f;
385  INTFLOAT v;
386  if (i != 6) {
387  f = tan((double)i * M_PI / 12.0);
388  v = FIXR(f / (1.0 + f));
389  } else {
390  v = FIXR(1.0);
391  }
392  is_table[0][ i] = v;
393  is_table[1][6 - i] = v;
394  }
395  /* invalid values */
396  for (i = 7; i < 16; i++)
397  is_table[0][i] = is_table[1][i] = 0.0;
398 
399  for (i = 0; i < 16; i++) {
400  double f;
401  int e, k;
402 
403  for (j = 0; j < 2; j++) {
404  e = -(j + 1) * ((i + 1) >> 1);
405  f = exp2(e / 4.0);
406  k = i & 1;
407  is_table_lsf[j][k ^ 1][i] = FIXR(f);
408  is_table_lsf[j][k ][i] = FIXR(1.0);
409  av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
410  i, j, (float) is_table_lsf[j][0][i],
411  (float) is_table_lsf[j][1][i]);
412  }
413  }
414 
415  for (i = 0; i < 8; i++) {
416  float ci, cs, ca;
417  ci = ci_table[i];
418  cs = 1.0 / sqrt(1.0 + ci * ci);
419  ca = cs * ci;
420 #if !CONFIG_FLOAT
421  csa_table[i][0] = FIXHR(cs/4);
422  csa_table[i][1] = FIXHR(ca/4);
423  csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
424  csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
425 #else
426  csa_table[i][0] = cs;
427  csa_table[i][1] = ca;
428  csa_table[i][2] = ca + cs;
429  csa_table[i][3] = ca - cs;
430 #endif
431  }
432 }
433 
434 static av_cold int decode_init(AVCodecContext * avctx)
435 {
436  static int initialized_tables = 0;
437  MPADecodeContext *s = avctx->priv_data;
438 
439  if (!initialized_tables) {
440  decode_init_static();
441  initialized_tables = 1;
442  }
443 
444  s->avctx = avctx;
445 
446  avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
447  ff_mpadsp_init(&s->mpadsp);
448 
449  if (avctx->request_sample_fmt == OUT_FMT &&
450  avctx->codec_id != AV_CODEC_ID_MP3ON4)
451  avctx->sample_fmt = OUT_FMT;
452  else
453  avctx->sample_fmt = OUT_FMT_P;
454  s->err_recognition = avctx->err_recognition;
455 
456  if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
457  s->adu_mode = 1;
458 
459  return 0;
460 }
461 
462 #define C3 FIXHR(0.86602540378443864676/2)
463 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
464 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
465 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
466 
467 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
468  cases. */
469 static void imdct12(INTFLOAT *out, INTFLOAT *in)
470 {
471  INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
472 
473  in0 = in[0*3];
474  in1 = in[1*3] + in[0*3];
475  in2 = in[2*3] + in[1*3];
476  in3 = in[3*3] + in[2*3];
477  in4 = in[4*3] + in[3*3];
478  in5 = in[5*3] + in[4*3];
479  in5 += in3;
480  in3 += in1;
481 
482  in2 = MULH3(in2, C3, 2);
483  in3 = MULH3(in3, C3, 4);
484 
485  t1 = in0 - in4;
486  t2 = MULH3(in1 - in5, C4, 2);
487 
488  out[ 7] =
489  out[10] = t1 + t2;
490  out[ 1] =
491  out[ 4] = t1 - t2;
492 
493  in0 += SHR(in4, 1);
494  in4 = in0 + in2;
495  in5 += 2*in1;
496  in1 = MULH3(in5 + in3, C5, 1);
497  out[ 8] =
498  out[ 9] = in4 + in1;
499  out[ 2] =
500  out[ 3] = in4 - in1;
501 
502  in0 -= in2;
503  in5 = MULH3(in5 - in3, C6, 2);
504  out[ 0] =
505  out[ 5] = in0 - in5;
506  out[ 6] =
507  out[11] = in0 + in5;
508 }
509 
510 /* return the number of decoded frames */
511 static int mp_decode_layer1(MPADecodeContext *s)
512 {
513  int bound, i, v, n, ch, j, mant;
514  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
515  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
516 
517  if (s->mode == MPA_JSTEREO)
518  bound = (s->mode_ext + 1) * 4;
519  else
520  bound = SBLIMIT;
521 
522  /* allocation bits */
523  for (i = 0; i < bound; i++) {
524  for (ch = 0; ch < s->nb_channels; ch++) {
525  allocation[ch][i] = get_bits(&s->gb, 4);
526  }
527  }
528  for (i = bound; i < SBLIMIT; i++)
529  allocation[0][i] = get_bits(&s->gb, 4);
530 
531  /* scale factors */
532  for (i = 0; i < bound; i++) {
533  for (ch = 0; ch < s->nb_channels; ch++) {
534  if (allocation[ch][i])
535  scale_factors[ch][i] = get_bits(&s->gb, 6);
536  }
537  }
538  for (i = bound; i < SBLIMIT; i++) {
539  if (allocation[0][i]) {
540  scale_factors[0][i] = get_bits(&s->gb, 6);
541  scale_factors[1][i] = get_bits(&s->gb, 6);
542  }
543  }
544 
545  /* compute samples */
546  for (j = 0; j < 12; j++) {
547  for (i = 0; i < bound; i++) {
548  for (ch = 0; ch < s->nb_channels; ch++) {
549  n = allocation[ch][i];
550  if (n) {
551  mant = get_bits(&s->gb, n + 1);
552  v = l1_unscale(n, mant, scale_factors[ch][i]);
553  } else {
554  v = 0;
555  }
556  s->sb_samples[ch][j][i] = v;
557  }
558  }
559  for (i = bound; i < SBLIMIT; i++) {
560  n = allocation[0][i];
561  if (n) {
562  mant = get_bits(&s->gb, n + 1);
563  v = l1_unscale(n, mant, scale_factors[0][i]);
564  s->sb_samples[0][j][i] = v;
565  v = l1_unscale(n, mant, scale_factors[1][i]);
566  s->sb_samples[1][j][i] = v;
567  } else {
568  s->sb_samples[0][j][i] = 0;
569  s->sb_samples[1][j][i] = 0;
570  }
571  }
572  }
573  return 12;
574 }
575 
576 static int mp_decode_layer2(MPADecodeContext *s)
577 {
578  int sblimit; /* number of used subbands */
579  const unsigned char *alloc_table;
580  int table, bit_alloc_bits, i, j, ch, bound, v;
581  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
582  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
583  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
584  int scale, qindex, bits, steps, k, l, m, b;
585 
586  /* select decoding table */
587  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
588  s->sample_rate, s->lsf);
589  sblimit = ff_mpa_sblimit_table[table];
590  alloc_table = ff_mpa_alloc_tables[table];
591 
592  if (s->mode == MPA_JSTEREO)
593  bound = (s->mode_ext + 1) * 4;
594  else
595  bound = sblimit;
596 
597  av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
598 
599  /* sanity check */
600  if (bound > sblimit)
601  bound = sblimit;
602 
603  /* parse bit allocation */
604  j = 0;
605  for (i = 0; i < bound; i++) {
606  bit_alloc_bits = alloc_table[j];
607  for (ch = 0; ch < s->nb_channels; ch++)
608  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
609  j += 1 << bit_alloc_bits;
610  }
611  for (i = bound; i < sblimit; i++) {
612  bit_alloc_bits = alloc_table[j];
613  v = get_bits(&s->gb, bit_alloc_bits);
614  bit_alloc[0][i] = v;
615  bit_alloc[1][i] = v;
616  j += 1 << bit_alloc_bits;
617  }
618 
619  /* scale codes */
620  for (i = 0; i < sblimit; i++) {
621  for (ch = 0; ch < s->nb_channels; ch++) {
622  if (bit_alloc[ch][i])
623  scale_code[ch][i] = get_bits(&s->gb, 2);
624  }
625  }
626 
627  /* scale factors */
628  for (i = 0; i < sblimit; i++) {
629  for (ch = 0; ch < s->nb_channels; ch++) {
630  if (bit_alloc[ch][i]) {
631  sf = scale_factors[ch][i];
632  switch (scale_code[ch][i]) {
633  default:
634  case 0:
635  sf[0] = get_bits(&s->gb, 6);
636  sf[1] = get_bits(&s->gb, 6);
637  sf[2] = get_bits(&s->gb, 6);
638  break;
639  case 2:
640  sf[0] = get_bits(&s->gb, 6);
641  sf[1] = sf[0];
642  sf[2] = sf[0];
643  break;
644  case 1:
645  sf[0] = get_bits(&s->gb, 6);
646  sf[2] = get_bits(&s->gb, 6);
647  sf[1] = sf[0];
648  break;
649  case 3:
650  sf[0] = get_bits(&s->gb, 6);
651  sf[2] = get_bits(&s->gb, 6);
652  sf[1] = sf[2];
653  break;
654  }
655  }
656  }
657  }
658 
659  /* samples */
660  for (k = 0; k < 3; k++) {
661  for (l = 0; l < 12; l += 3) {
662  j = 0;
663  for (i = 0; i < bound; i++) {
664  bit_alloc_bits = alloc_table[j];
665  for (ch = 0; ch < s->nb_channels; ch++) {
666  b = bit_alloc[ch][i];
667  if (b) {
668  scale = scale_factors[ch][i][k];
669  qindex = alloc_table[j+b];
670  bits = ff_mpa_quant_bits[qindex];
671  if (bits < 0) {
672  int v2;
673  /* 3 values at the same time */
674  v = get_bits(&s->gb, -bits);
675  v2 = division_tabs[qindex][v];
676  steps = ff_mpa_quant_steps[qindex];
677 
678  s->sb_samples[ch][k * 12 + l + 0][i] =
679  l2_unscale_group(steps, v2 & 15, scale);
680  s->sb_samples[ch][k * 12 + l + 1][i] =
681  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
682  s->sb_samples[ch][k * 12 + l + 2][i] =
683  l2_unscale_group(steps, v2 >> 8 , scale);
684  } else {
685  for (m = 0; m < 3; m++) {
686  v = get_bits(&s->gb, bits);
687  v = l1_unscale(bits - 1, v, scale);
688  s->sb_samples[ch][k * 12 + l + m][i] = v;
689  }
690  }
691  } else {
692  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
693  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
694  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
695  }
696  }
697  /* next subband in alloc table */
698  j += 1 << bit_alloc_bits;
699  }
700  /* XXX: find a way to avoid this duplication of code */
701  for (i = bound; i < sblimit; i++) {
702  bit_alloc_bits = alloc_table[j];
703  b = bit_alloc[0][i];
704  if (b) {
705  int mant, scale0, scale1;
706  scale0 = scale_factors[0][i][k];
707  scale1 = scale_factors[1][i][k];
708  qindex = alloc_table[j+b];
709  bits = ff_mpa_quant_bits[qindex];
710  if (bits < 0) {
711  /* 3 values at the same time */
712  v = get_bits(&s->gb, -bits);
713  steps = ff_mpa_quant_steps[qindex];
714  mant = v % steps;
715  v = v / steps;
716  s->sb_samples[0][k * 12 + l + 0][i] =
717  l2_unscale_group(steps, mant, scale0);
718  s->sb_samples[1][k * 12 + l + 0][i] =
719  l2_unscale_group(steps, mant, scale1);
720  mant = v % steps;
721  v = v / steps;
722  s->sb_samples[0][k * 12 + l + 1][i] =
723  l2_unscale_group(steps, mant, scale0);
724  s->sb_samples[1][k * 12 + l + 1][i] =
725  l2_unscale_group(steps, mant, scale1);
726  s->sb_samples[0][k * 12 + l + 2][i] =
727  l2_unscale_group(steps, v, scale0);
728  s->sb_samples[1][k * 12 + l + 2][i] =
729  l2_unscale_group(steps, v, scale1);
730  } else {
731  for (m = 0; m < 3; m++) {
732  mant = get_bits(&s->gb, bits);
733  s->sb_samples[0][k * 12 + l + m][i] =
734  l1_unscale(bits - 1, mant, scale0);
735  s->sb_samples[1][k * 12 + l + m][i] =
736  l1_unscale(bits - 1, mant, scale1);
737  }
738  }
739  } else {
740  s->sb_samples[0][k * 12 + l + 0][i] = 0;
741  s->sb_samples[0][k * 12 + l + 1][i] = 0;
742  s->sb_samples[0][k * 12 + l + 2][i] = 0;
743  s->sb_samples[1][k * 12 + l + 0][i] = 0;
744  s->sb_samples[1][k * 12 + l + 1][i] = 0;
745  s->sb_samples[1][k * 12 + l + 2][i] = 0;
746  }
747  /* next subband in alloc table */
748  j += 1 << bit_alloc_bits;
749  }
750  /* fill remaining samples to zero */
751  for (i = sblimit; i < SBLIMIT; i++) {
752  for (ch = 0; ch < s->nb_channels; ch++) {
753  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
754  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
755  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
756  }
757  }
758  }
759  }
760  return 3 * 12;
761 }
762 
763 #define SPLIT(dst,sf,n) \
764  if (n == 3) { \
765  int m = (sf * 171) >> 9; \
766  dst = sf - 3 * m; \
767  sf = m; \
768  } else if (n == 4) { \
769  dst = sf & 3; \
770  sf >>= 2; \
771  } else if (n == 5) { \
772  int m = (sf * 205) >> 10; \
773  dst = sf - 5 * m; \
774  sf = m; \
775  } else if (n == 6) { \
776  int m = (sf * 171) >> 10; \
777  dst = sf - 6 * m; \
778  sf = m; \
779  } else { \
780  dst = 0; \
781  }
782 
783 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
784  int n3)
785 {
786  SPLIT(slen[3], sf, n3)
787  SPLIT(slen[2], sf, n2)
788  SPLIT(slen[1], sf, n1)
789  slen[0] = sf;
790 }
791 
792 static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
793  int16_t *exponents)
794 {
795  const uint8_t *bstab, *pretab;
796  int len, i, j, k, l, v0, shift, gain, gains[3];
797  int16_t *exp_ptr;
798 
799  exp_ptr = exponents;
800  gain = g->global_gain - 210;
801  shift = g->scalefac_scale + 1;
802 
803  bstab = band_size_long[s->sample_rate_index];
804  pretab = mpa_pretab[g->preflag];
805  for (i = 0; i < g->long_end; i++) {
806  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
807  len = bstab[i];
808  for (j = len; j > 0; j--)
809  *exp_ptr++ = v0;
810  }
811 
812  if (g->short_start < 13) {
813  bstab = band_size_short[s->sample_rate_index];
814  gains[0] = gain - (g->subblock_gain[0] << 3);
815  gains[1] = gain - (g->subblock_gain[1] << 3);
816  gains[2] = gain - (g->subblock_gain[2] << 3);
817  k = g->long_end;
818  for (i = g->short_start; i < 13; i++) {
819  len = bstab[i];
820  for (l = 0; l < 3; l++) {
821  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
822  for (j = len; j > 0; j--)
823  *exp_ptr++ = v0;
824  }
825  }
826  }
827 }
828 
829 /* handle n = 0 too */
830 static inline int get_bitsz(GetBitContext *s, int n)
831 {
832  return n ? get_bits(s, n) : 0;
833 }
834 
835 
836 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
837  int *end_pos2)
838 {
839  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
840  s->gb = s->in_gb;
841  s->in_gb.buffer = NULL;
842  av_assert2((get_bits_count(&s->gb) & 7) == 0);
843  skip_bits_long(&s->gb, *pos - *end_pos);
844  *end_pos2 =
845  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
846  *pos = get_bits_count(&s->gb);
847  }
848 }
849 
850 /* Following is a optimized code for
851  INTFLOAT v = *src
852  if(get_bits1(&s->gb))
853  v = -v;
854  *dst = v;
855 */
856 #if CONFIG_FLOAT
857 #define READ_FLIP_SIGN(dst,src) \
858  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
859  AV_WN32A(dst, v);
860 #else
861 #define READ_FLIP_SIGN(dst,src) \
862  v = -get_bits1(&s->gb); \
863  *(dst) = (*(src) ^ v) - v;
864 #endif
865 
866 static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
867  int16_t *exponents, int end_pos2)
868 {
869  int s_index;
870  int i;
871  int last_pos, bits_left;
872  VLC *vlc;
873  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
874 
875  /* low frequencies (called big values) */
876  s_index = 0;
877  for (i = 0; i < 3; i++) {
878  int j, k, l, linbits;
879  j = g->region_size[i];
880  if (j == 0)
881  continue;
882  /* select vlc table */
883  k = g->table_select[i];
884  l = mpa_huff_data[k][0];
885  linbits = mpa_huff_data[k][1];
886  vlc = &huff_vlc[l];
887 
888  if (!l) {
889  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
890  s_index += 2 * j;
891  continue;
892  }
893 
894  /* read huffcode and compute each couple */
895  for (; j > 0; j--) {
896  int exponent, x, y;
897  int v;
898  int pos = get_bits_count(&s->gb);
899 
900  if (pos >= end_pos){
901  switch_buffer(s, &pos, &end_pos, &end_pos2);
902  if (pos >= end_pos)
903  break;
904  }
905  y = get_vlc2(&s->gb, vlc->table, 7, 3);
906 
907  if (!y) {
908  g->sb_hybrid[s_index ] =
909  g->sb_hybrid[s_index+1] = 0;
910  s_index += 2;
911  continue;
912  }
913 
914  exponent= exponents[s_index];
915 
916  av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
917  i, g->region_size[i] - j, x, y, exponent);
918  if (y & 16) {
919  x = y >> 5;
920  y = y & 0x0f;
921  if (x < 15) {
922  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
923  } else {
924  x += get_bitsz(&s->gb, linbits);
925  v = l3_unscale(x, exponent);
926  if (get_bits1(&s->gb))
927  v = -v;
928  g->sb_hybrid[s_index] = v;
929  }
930  if (y < 15) {
931  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
932  } else {
933  y += get_bitsz(&s->gb, linbits);
934  v = l3_unscale(y, exponent);
935  if (get_bits1(&s->gb))
936  v = -v;
937  g->sb_hybrid[s_index+1] = v;
938  }
939  } else {
940  x = y >> 5;
941  y = y & 0x0f;
942  x += y;
943  if (x < 15) {
944  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
945  } else {
946  x += get_bitsz(&s->gb, linbits);
947  v = l3_unscale(x, exponent);
948  if (get_bits1(&s->gb))
949  v = -v;
950  g->sb_hybrid[s_index+!!y] = v;
951  }
952  g->sb_hybrid[s_index + !y] = 0;
953  }
954  s_index += 2;
955  }
956  }
957 
958  /* high frequencies */
959  vlc = &huff_quad_vlc[g->count1table_select];
960  last_pos = 0;
961  while (s_index <= 572) {
962  int pos, code;
963  pos = get_bits_count(&s->gb);
964  if (pos >= end_pos) {
965  if (pos > end_pos2 && last_pos) {
966  /* some encoders generate an incorrect size for this
967  part. We must go back into the data */
968  s_index -= 4;
969  skip_bits_long(&s->gb, last_pos - pos);
970  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
971  if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
972  s_index=0;
973  break;
974  }
975  switch_buffer(s, &pos, &end_pos, &end_pos2);
976  if (pos >= end_pos)
977  break;
978  }
979  last_pos = pos;
980 
981  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
982  av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
983  g->sb_hybrid[s_index+0] =
984  g->sb_hybrid[s_index+1] =
985  g->sb_hybrid[s_index+2] =
986  g->sb_hybrid[s_index+3] = 0;
987  while (code) {
988  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
989  int v;
990  int pos = s_index + idxtab[code];
991  code ^= 8 >> idxtab[code];
992  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
993  }
994  s_index += 4;
995  }
996  /* skip extension bits */
997  bits_left = end_pos2 - get_bits_count(&s->gb);
998  if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
999  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1000  s_index=0;
1001  } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
1002  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1003  s_index = 0;
1004  }
1005  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
1006  skip_bits_long(&s->gb, bits_left);
1007 
1008  i = get_bits_count(&s->gb);
1009  switch_buffer(s, &i, &end_pos, &end_pos2);
1010 
1011  return 0;
1012 }
1013 
1014 /* Reorder short blocks from bitstream order to interleaved order. It
1015  would be faster to do it in parsing, but the code would be far more
1016  complicated */
1017 static void reorder_block(MPADecodeContext *s, GranuleDef *g)
1018 {
1019  int i, j, len;
1020  INTFLOAT *ptr, *dst, *ptr1;
1021  INTFLOAT tmp[576];
1022 
1023  if (g->block_type != 2)
1024  return;
1025 
1026  if (g->switch_point) {
1027  if (s->sample_rate_index != 8)
1028  ptr = g->sb_hybrid + 36;
1029  else
1030  ptr = g->sb_hybrid + 72;
1031  } else {
1032  ptr = g->sb_hybrid;
1033  }
1034 
1035  for (i = g->short_start; i < 13; i++) {
1036  len = band_size_short[s->sample_rate_index][i];
1037  ptr1 = ptr;
1038  dst = tmp;
1039  for (j = len; j > 0; j--) {
1040  *dst++ = ptr[0*len];
1041  *dst++ = ptr[1*len];
1042  *dst++ = ptr[2*len];
1043  ptr++;
1044  }
1045  ptr += 2 * len;
1046  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1047  }
1048 }
1049 
1050 #define ISQRT2 FIXR(0.70710678118654752440)
1051 
1052 static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
1053 {
1054  int i, j, k, l;
1055  int sf_max, sf, len, non_zero_found;
1056  INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1057  int non_zero_found_short[3];
1058 
1059  /* intensity stereo */
1060  if (s->mode_ext & MODE_EXT_I_STEREO) {
1061  if (!s->lsf) {
1062  is_tab = is_table;
1063  sf_max = 7;
1064  } else {
1065  is_tab = is_table_lsf[g1->scalefac_compress & 1];
1066  sf_max = 16;
1067  }
1068 
1069  tab0 = g0->sb_hybrid + 576;
1070  tab1 = g1->sb_hybrid + 576;
1071 
1072  non_zero_found_short[0] = 0;
1073  non_zero_found_short[1] = 0;
1074  non_zero_found_short[2] = 0;
1075  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1076  for (i = 12; i >= g1->short_start; i--) {
1077  /* for last band, use previous scale factor */
1078  if (i != 11)
1079  k -= 3;
1080  len = band_size_short[s->sample_rate_index][i];
1081  for (l = 2; l >= 0; l--) {
1082  tab0 -= len;
1083  tab1 -= len;
1084  if (!non_zero_found_short[l]) {
1085  /* test if non zero band. if so, stop doing i-stereo */
1086  for (j = 0; j < len; j++) {
1087  if (tab1[j] != 0) {
1088  non_zero_found_short[l] = 1;
1089  goto found1;
1090  }
1091  }
1092  sf = g1->scale_factors[k + l];
1093  if (sf >= sf_max)
1094  goto found1;
1095 
1096  v1 = is_tab[0][sf];
1097  v2 = is_tab[1][sf];
1098  for (j = 0; j < len; j++) {
1099  tmp0 = tab0[j];
1100  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1101  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1102  }
1103  } else {
1104 found1:
1105  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1106  /* lower part of the spectrum : do ms stereo
1107  if enabled */
1108  for (j = 0; j < len; j++) {
1109  tmp0 = tab0[j];
1110  tmp1 = tab1[j];
1111  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1112  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1113  }
1114  }
1115  }
1116  }
1117  }
1118 
1119  non_zero_found = non_zero_found_short[0] |
1120  non_zero_found_short[1] |
1121  non_zero_found_short[2];
1122 
1123  for (i = g1->long_end - 1;i >= 0;i--) {
1124  len = band_size_long[s->sample_rate_index][i];
1125  tab0 -= len;
1126  tab1 -= len;
1127  /* test if non zero band. if so, stop doing i-stereo */
1128  if (!non_zero_found) {
1129  for (j = 0; j < len; j++) {
1130  if (tab1[j] != 0) {
1131  non_zero_found = 1;
1132  goto found2;
1133  }
1134  }
1135  /* for last band, use previous scale factor */
1136  k = (i == 21) ? 20 : i;
1137  sf = g1->scale_factors[k];
1138  if (sf >= sf_max)
1139  goto found2;
1140  v1 = is_tab[0][sf];
1141  v2 = is_tab[1][sf];
1142  for (j = 0; j < len; j++) {
1143  tmp0 = tab0[j];
1144  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1145  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1146  }
1147  } else {
1148 found2:
1149  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1150  /* lower part of the spectrum : do ms stereo
1151  if enabled */
1152  for (j = 0; j < len; j++) {
1153  tmp0 = tab0[j];
1154  tmp1 = tab1[j];
1155  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1156  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1157  }
1158  }
1159  }
1160  }
1161  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1162  /* ms stereo ONLY */
1163  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1164  global gain */
1165 #if CONFIG_FLOAT
1166  s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1167 #else
1168  tab0 = g0->sb_hybrid;
1169  tab1 = g1->sb_hybrid;
1170  for (i = 0; i < 576; i++) {
1171  tmp0 = tab0[i];
1172  tmp1 = tab1[i];
1173  tab0[i] = tmp0 + tmp1;
1174  tab1[i] = tmp0 - tmp1;
1175  }
1176 #endif
1177  }
1178 }
1179 
1180 #if CONFIG_FLOAT
1181 #if HAVE_MIPSFPU
1182 # include "mips/compute_antialias_float.h"
1183 #endif /* HAVE_MIPSFPU */
1184 #else
1185 #if HAVE_MIPSDSPR1
1186 # include "mips/compute_antialias_fixed.h"
1187 #endif /* HAVE_MIPSDSPR1 */
1188 #endif /* CONFIG_FLOAT */
1189 
1190 #ifndef compute_antialias
1191 #if CONFIG_FLOAT
1192 #define AA(j) do { \
1193  float tmp0 = ptr[-1-j]; \
1194  float tmp1 = ptr[ j]; \
1195  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1196  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1197  } while (0)
1198 #else
1199 #define AA(j) do { \
1200  int tmp0 = ptr[-1-j]; \
1201  int tmp1 = ptr[ j]; \
1202  int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1203  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1204  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1205  } while (0)
1206 #endif
1207 
1208 static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
1209 {
1210  INTFLOAT *ptr;
1211  int n, i;
1212 
1213  /* we antialias only "long" bands */
1214  if (g->block_type == 2) {
1215  if (!g->switch_point)
1216  return;
1217  /* XXX: check this for 8000Hz case */
1218  n = 1;
1219  } else {
1220  n = SBLIMIT - 1;
1221  }
1222 
1223  ptr = g->sb_hybrid + 18;
1224  for (i = n; i > 0; i--) {
1225  AA(0);
1226  AA(1);
1227  AA(2);
1228  AA(3);
1229  AA(4);
1230  AA(5);
1231  AA(6);
1232  AA(7);
1233 
1234  ptr += 18;
1235  }
1236 }
1237 #endif /* compute_antialias */
1238 
1239 static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
1240  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1241 {
1242  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1243  INTFLOAT out2[12];
1244  int i, j, mdct_long_end, sblimit;
1245 
1246  /* find last non zero block */
1247  ptr = g->sb_hybrid + 576;
1248  ptr1 = g->sb_hybrid + 2 * 18;
1249  while (ptr >= ptr1) {
1250  int32_t *p;
1251  ptr -= 6;
1252  p = (int32_t*)ptr;
1253  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1254  break;
1255  }
1256  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1257 
1258  if (g->block_type == 2) {
1259  /* XXX: check for 8000 Hz */
1260  if (g->switch_point)
1261  mdct_long_end = 2;
1262  else
1263  mdct_long_end = 0;
1264  } else {
1265  mdct_long_end = sblimit;
1266  }
1267 
1268  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1269  mdct_long_end, g->switch_point,
1270  g->block_type);
1271 
1272  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1273  ptr = g->sb_hybrid + 18 * mdct_long_end;
1274 
1275  for (j = mdct_long_end; j < sblimit; j++) {
1276  /* select frequency inversion */
1277  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1278  out_ptr = sb_samples + j;
1279 
1280  for (i = 0; i < 6; i++) {
1281  *out_ptr = buf[4*i];
1282  out_ptr += SBLIMIT;
1283  }
1284  imdct12(out2, ptr + 0);
1285  for (i = 0; i < 6; i++) {
1286  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1287  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1288  out_ptr += SBLIMIT;
1289  }
1290  imdct12(out2, ptr + 1);
1291  for (i = 0; i < 6; i++) {
1292  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1293  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1294  out_ptr += SBLIMIT;
1295  }
1296  imdct12(out2, ptr + 2);
1297  for (i = 0; i < 6; i++) {
1298  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1299  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1300  buf[4*(i + 6*2)] = 0;
1301  }
1302  ptr += 18;
1303  buf += (j&3) != 3 ? 1 : (4*18-3);
1304  }
1305  /* zero bands */
1306  for (j = sblimit; j < SBLIMIT; j++) {
1307  /* overlap */
1308  out_ptr = sb_samples + j;
1309  for (i = 0; i < 18; i++) {
1310  *out_ptr = buf[4*i];
1311  buf[4*i] = 0;
1312  out_ptr += SBLIMIT;
1313  }
1314  buf += (j&3) != 3 ? 1 : (4*18-3);
1315  }
1316 }
1317 
1318 /* main layer3 decoding function */
1319 static int mp_decode_layer3(MPADecodeContext *s)
1320 {
1321  int nb_granules, main_data_begin;
1322  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1323  GranuleDef *g;
1324  int16_t exponents[576]; //FIXME try INTFLOAT
1325 
1326  /* read side info */
1327  if (s->lsf) {
1328  main_data_begin = get_bits(&s->gb, 8);
1329  skip_bits(&s->gb, s->nb_channels);
1330  nb_granules = 1;
1331  } else {
1332  main_data_begin = get_bits(&s->gb, 9);
1333  if (s->nb_channels == 2)
1334  skip_bits(&s->gb, 3);
1335  else
1336  skip_bits(&s->gb, 5);
1337  nb_granules = 2;
1338  for (ch = 0; ch < s->nb_channels; ch++) {
1339  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1340  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1341  }
1342  }
1343 
1344  for (gr = 0; gr < nb_granules; gr++) {
1345  for (ch = 0; ch < s->nb_channels; ch++) {
1346  av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1347  g = &s->granules[ch][gr];
1348  g->part2_3_length = get_bits(&s->gb, 12);
1349  g->big_values = get_bits(&s->gb, 9);
1350  if (g->big_values > 288) {
1351  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1352  return AVERROR_INVALIDDATA;
1353  }
1354 
1355  g->global_gain = get_bits(&s->gb, 8);
1356  /* if MS stereo only is selected, we precompute the
1357  1/sqrt(2) renormalization factor */
1358  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1359  MODE_EXT_MS_STEREO)
1360  g->global_gain -= 2;
1361  if (s->lsf)
1362  g->scalefac_compress = get_bits(&s->gb, 9);
1363  else
1364  g->scalefac_compress = get_bits(&s->gb, 4);
1365  blocksplit_flag = get_bits1(&s->gb);
1366  if (blocksplit_flag) {
1367  g->block_type = get_bits(&s->gb, 2);
1368  if (g->block_type == 0) {
1369  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1370  return AVERROR_INVALIDDATA;
1371  }
1372  g->switch_point = get_bits1(&s->gb);
1373  for (i = 0; i < 2; i++)
1374  g->table_select[i] = get_bits(&s->gb, 5);
1375  for (i = 0; i < 3; i++)
1376  g->subblock_gain[i] = get_bits(&s->gb, 3);
1377  init_short_region(s, g);
1378  } else {
1379  int region_address1, region_address2;
1380  g->block_type = 0;
1381  g->switch_point = 0;
1382  for (i = 0; i < 3; i++)
1383  g->table_select[i] = get_bits(&s->gb, 5);
1384  /* compute huffman coded region sizes */
1385  region_address1 = get_bits(&s->gb, 4);
1386  region_address2 = get_bits(&s->gb, 3);
1387  av_dlog(s->avctx, "region1=%d region2=%d\n",
1388  region_address1, region_address2);
1389  init_long_region(s, g, region_address1, region_address2);
1390  }
1391  region_offset2size(g);
1392  compute_band_indexes(s, g);
1393 
1394  g->preflag = 0;
1395  if (!s->lsf)
1396  g->preflag = get_bits1(&s->gb);
1397  g->scalefac_scale = get_bits1(&s->gb);
1398  g->count1table_select = get_bits1(&s->gb);
1399  av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1400  g->block_type, g->switch_point);
1401  }
1402  }
1403 
1404  if (!s->adu_mode) {
1405  int skip;
1406  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1407  int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES);
1408  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1409  /* now we get bits from the main_data_begin offset */
1410  av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1411  main_data_begin, s->last_buf_size);
1412 
1413  memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
1414  s->in_gb = s->gb;
1415  init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1416 #if !UNCHECKED_BITSTREAM_READER
1417  s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8;
1418 #endif
1419  s->last_buf_size <<= 3;
1420  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1421  for (ch = 0; ch < s->nb_channels; ch++) {
1422  g = &s->granules[ch][gr];
1423  s->last_buf_size += g->part2_3_length;
1424  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1425  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1426  }
1427  }
1428  skip = s->last_buf_size - 8 * main_data_begin;
1429  if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
1430  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
1431  s->gb = s->in_gb;
1432  s->in_gb.buffer = NULL;
1433  } else {
1434  skip_bits_long(&s->gb, skip);
1435  }
1436  } else {
1437  gr = 0;
1438  }
1439 
1440  for (; gr < nb_granules; gr++) {
1441  for (ch = 0; ch < s->nb_channels; ch++) {
1442  g = &s->granules[ch][gr];
1443  bits_pos = get_bits_count(&s->gb);
1444 
1445  if (!s->lsf) {
1446  uint8_t *sc;
1447  int slen, slen1, slen2;
1448 
1449  /* MPEG1 scale factors */
1450  slen1 = slen_table[0][g->scalefac_compress];
1451  slen2 = slen_table[1][g->scalefac_compress];
1452  av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1453  if (g->block_type == 2) {
1454  n = g->switch_point ? 17 : 18;
1455  j = 0;
1456  if (slen1) {
1457  for (i = 0; i < n; i++)
1458  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1459  } else {
1460  for (i = 0; i < n; i++)
1461  g->scale_factors[j++] = 0;
1462  }
1463  if (slen2) {
1464  for (i = 0; i < 18; i++)
1465  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1466  for (i = 0; i < 3; i++)
1467  g->scale_factors[j++] = 0;
1468  } else {
1469  for (i = 0; i < 21; i++)
1470  g->scale_factors[j++] = 0;
1471  }
1472  } else {
1473  sc = s->granules[ch][0].scale_factors;
1474  j = 0;
1475  for (k = 0; k < 4; k++) {
1476  n = k == 0 ? 6 : 5;
1477  if ((g->scfsi & (0x8 >> k)) == 0) {
1478  slen = (k < 2) ? slen1 : slen2;
1479  if (slen) {
1480  for (i = 0; i < n; i++)
1481  g->scale_factors[j++] = get_bits(&s->gb, slen);
1482  } else {
1483  for (i = 0; i < n; i++)
1484  g->scale_factors[j++] = 0;
1485  }
1486  } else {
1487  /* simply copy from last granule */
1488  for (i = 0; i < n; i++) {
1489  g->scale_factors[j] = sc[j];
1490  j++;
1491  }
1492  }
1493  }
1494  g->scale_factors[j++] = 0;
1495  }
1496  } else {
1497  int tindex, tindex2, slen[4], sl, sf;
1498 
1499  /* LSF scale factors */
1500  if (g->block_type == 2)
1501  tindex = g->switch_point ? 2 : 1;
1502  else
1503  tindex = 0;
1504 
1505  sf = g->scalefac_compress;
1506  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1507  /* intensity stereo case */
1508  sf >>= 1;
1509  if (sf < 180) {
1510  lsf_sf_expand(slen, sf, 6, 6, 0);
1511  tindex2 = 3;
1512  } else if (sf < 244) {
1513  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1514  tindex2 = 4;
1515  } else {
1516  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1517  tindex2 = 5;
1518  }
1519  } else {
1520  /* normal case */
1521  if (sf < 400) {
1522  lsf_sf_expand(slen, sf, 5, 4, 4);
1523  tindex2 = 0;
1524  } else if (sf < 500) {
1525  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1526  tindex2 = 1;
1527  } else {
1528  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1529  tindex2 = 2;
1530  g->preflag = 1;
1531  }
1532  }
1533 
1534  j = 0;
1535  for (k = 0; k < 4; k++) {
1536  n = lsf_nsf_table[tindex2][tindex][k];
1537  sl = slen[k];
1538  if (sl) {
1539  for (i = 0; i < n; i++)
1540  g->scale_factors[j++] = get_bits(&s->gb, sl);
1541  } else {
1542  for (i = 0; i < n; i++)
1543  g->scale_factors[j++] = 0;
1544  }
1545  }
1546  /* XXX: should compute exact size */
1547  for (; j < 40; j++)
1548  g->scale_factors[j] = 0;
1549  }
1550 
1551  exponents_from_scale_factors(s, g, exponents);
1552 
1553  /* read Huffman coded residue */
1554  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1555  } /* ch */
1556 
1557  if (s->mode == MPA_JSTEREO)
1558  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1559 
1560  for (ch = 0; ch < s->nb_channels; ch++) {
1561  g = &s->granules[ch][gr];
1562 
1563  reorder_block(s, g);
1564  compute_antialias(s, g);
1565  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1566  }
1567  } /* gr */
1568  if (get_bits_count(&s->gb) < 0)
1569  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1570  return nb_granules * 18;
1571 }
1572 
1573 static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
1574  const uint8_t *buf, int buf_size)
1575 {
1576  int i, nb_frames, ch, ret;
1577  OUT_INT *samples_ptr;
1578 
1579  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1580 
1581  /* skip error protection field */
1582  if (s->error_protection)
1583  skip_bits(&s->gb, 16);
1584 
1585  switch(s->layer) {
1586  case 1:
1587  s->avctx->frame_size = 384;
1588  nb_frames = mp_decode_layer1(s);
1589  break;
1590  case 2:
1591  s->avctx->frame_size = 1152;
1592  nb_frames = mp_decode_layer2(s);
1593  break;
1594  case 3:
1595  s->avctx->frame_size = s->lsf ? 576 : 1152;
1596  default:
1597  nb_frames = mp_decode_layer3(s);
1598 
1599  s->last_buf_size=0;
1600  if (s->in_gb.buffer) {
1601  align_get_bits(&s->gb);
1602  i = get_bits_left(&s->gb)>>3;
1603  if (i >= 0 && i <= BACKSTEP_SIZE) {
1604  memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1605  s->last_buf_size=i;
1606  } else
1607  av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1608  s->gb = s->in_gb;
1609  s->in_gb.buffer = NULL;
1610  }
1611 
1612  align_get_bits(&s->gb);
1613  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1614  i = get_bits_left(&s->gb) >> 3;
1615 
1616  if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1617  if (i < 0)
1618  av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1619  i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1620  }
1621  av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
1622  memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1623  s->last_buf_size += i;
1624  }
1625 
1626  if(nb_frames < 0)
1627  return nb_frames;
1628 
1629  /* get output buffer */
1630  if (!samples) {
1631  av_assert0(s->frame != NULL);
1632  s->frame->nb_samples = s->avctx->frame_size;
1633  if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
1634  return ret;
1635  samples = (OUT_INT **)s->frame->extended_data;
1636  }
1637 
1638  /* apply the synthesis filter */
1639  for (ch = 0; ch < s->nb_channels; ch++) {
1640  int sample_stride;
1641  if (s->avctx->sample_fmt == OUT_FMT_P) {
1642  samples_ptr = samples[ch];
1643  sample_stride = 1;
1644  } else {
1645  samples_ptr = samples[0] + ch;
1646  sample_stride = s->nb_channels;
1647  }
1648  for (i = 0; i < nb_frames; i++) {
1649  RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
1650  &(s->synth_buf_offset[ch]),
1651  RENAME(ff_mpa_synth_window),
1652  &s->dither_state, samples_ptr,
1653  sample_stride, s->sb_samples[ch][i]);
1654  samples_ptr += 32 * sample_stride;
1655  }
1656  }
1657 
1658  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1659 }
1660 
1661 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1662  AVPacket *avpkt)
1663 {
1664  const uint8_t *buf = avpkt->data;
1665  int buf_size = avpkt->size;
1666  MPADecodeContext *s = avctx->priv_data;
1667  uint32_t header;
1668  int ret;
1669 
1670  while(buf_size && !*buf){
1671  buf++;
1672  buf_size--;
1673  }
1674 
1675  if (buf_size < HEADER_SIZE)
1676  return AVERROR_INVALIDDATA;
1677 
1678  header = AV_RB32(buf);
1679  if (header>>8 == AV_RB32("TAG")>>8) {
1680  av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
1681  return buf_size;
1682  }
1683  if (ff_mpa_check_header(header) < 0) {
1684  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1685  return AVERROR_INVALIDDATA;
1686  }
1687 
1688  if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1689  /* free format: prepare to compute frame size */
1690  s->frame_size = -1;
1691  return AVERROR_INVALIDDATA;
1692  }
1693  /* update codec info */
1694  avctx->channels = s->nb_channels;
1695  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1696  if (!avctx->bit_rate)
1697  avctx->bit_rate = s->bit_rate;
1698 
1699  if (s->frame_size <= 0 || s->frame_size > buf_size) {
1700  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1701  return AVERROR_INVALIDDATA;
1702  } else if (s->frame_size < buf_size) {
1703  av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
1704  buf_size= s->frame_size;
1705  }
1706 
1707  s->frame = data;
1708 
1709  ret = mp_decode_frame(s, NULL, buf, buf_size);
1710  if (ret >= 0) {
1711  s->frame->nb_samples = avctx->frame_size;
1712  *got_frame_ptr = 1;
1713  avctx->sample_rate = s->sample_rate;
1714  //FIXME maybe move the other codec info stuff from above here too
1715  } else {
1716  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1717  /* Only return an error if the bad frame makes up the whole packet or
1718  * the error is related to buffer management.
1719  * If there is more data in the packet, just consume the bad frame
1720  * instead of returning an error, which would discard the whole
1721  * packet. */
1722  *got_frame_ptr = 0;
1723  if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1724  return ret;
1725  }
1726  s->frame_size = 0;
1727  return buf_size;
1728 }
1729 
1730 static void mp_flush(MPADecodeContext *ctx)
1731 {
1732  memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1733  ctx->last_buf_size = 0;
1734 }
1735 
1736 static void flush(AVCodecContext *avctx)
1737 {
1738  mp_flush(avctx->priv_data);
1739 }
1740 
1741 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1742 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1743  int *got_frame_ptr, AVPacket *avpkt)
1744 {
1745  const uint8_t *buf = avpkt->data;
1746  int buf_size = avpkt->size;
1747  MPADecodeContext *s = avctx->priv_data;
1748  uint32_t header;
1749  int len, ret;
1750  int av_unused out_size;
1751 
1752  len = buf_size;
1753 
1754  // Discard too short frames
1755  if (buf_size < HEADER_SIZE) {
1756  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1757  return AVERROR_INVALIDDATA;
1758  }
1759 
1760 
1761  if (len > MPA_MAX_CODED_FRAME_SIZE)
1762  len = MPA_MAX_CODED_FRAME_SIZE;
1763 
1764  // Get header and restore sync word
1765  header = AV_RB32(buf) | 0xffe00000;
1766 
1767  if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
1768  av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1769  return AVERROR_INVALIDDATA;
1770  }
1771 
1772  avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
1773  /* update codec info */
1774  avctx->sample_rate = s->sample_rate;
1775  avctx->channels = s->nb_channels;
1776  if (!avctx->bit_rate)
1777  avctx->bit_rate = s->bit_rate;
1778 
1779  s->frame_size = len;
1780 
1781  s->frame = data;
1782 
1783  ret = mp_decode_frame(s, NULL, buf, buf_size);
1784  if (ret < 0) {
1785  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1786  return ret;
1787  }
1788 
1789  *got_frame_ptr = 1;
1790 
1791  return buf_size;
1792 }
1793 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1794 
1795 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1796 
1797 /**
1798  * Context for MP3On4 decoder
1799  */
1800 typedef struct MP3On4DecodeContext {
1801  int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
1802  int syncword; ///< syncword patch
1803  const uint8_t *coff; ///< channel offsets in output buffer
1804  MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
1805 } MP3On4DecodeContext;
1806 
1807 #include "mpeg4audio.h"
1808 
1809 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1810 
1811 /* number of mp3 decoder instances */
1812 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1813 
1814 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1815 static const uint8_t chan_offset[8][5] = {
1816  { 0 },
1817  { 0 }, // C
1818  { 0 }, // FLR
1819  { 2, 0 }, // C FLR
1820  { 2, 0, 3 }, // C FLR BS
1821  { 2, 0, 3 }, // C FLR BLRS
1822  { 2, 0, 4, 3 }, // C FLR BLRS LFE
1823  { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1824 };
1825 
1826 /* mp3on4 channel layouts */
1827 static const int16_t chan_layout[8] = {
1828  0,
1829  AV_CH_LAYOUT_MONO,
1830  AV_CH_LAYOUT_STEREO,
1831  AV_CH_LAYOUT_SURROUND,
1832  AV_CH_LAYOUT_4POINT0,
1833  AV_CH_LAYOUT_5POINT0,
1834  AV_CH_LAYOUT_5POINT1,
1835  AV_CH_LAYOUT_7POINT1
1836 };
1837 
1838 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1839 {
1840  MP3On4DecodeContext *s = avctx->priv_data;
1841  int i;
1842 
1843  for (i = 0; i < s->frames; i++)
1844  av_free(s->mp3decctx[i]);
1845 
1846  return 0;
1847 }
1848 
1849 
1850 static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
1851 {
1852  MP3On4DecodeContext *s = avctx->priv_data;
1853  MPEG4AudioConfig cfg;
1854  int i;
1855 
1856  if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
1857  av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1858  return AVERROR_INVALIDDATA;
1859  }
1860 
1861  avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
1862  avctx->extradata_size * 8, 1);
1863  if (!cfg.chan_config || cfg.chan_config > 7) {
1864  av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1865  return AVERROR_INVALIDDATA;
1866  }
1867  s->frames = mp3Frames[cfg.chan_config];
1868  s->coff = chan_offset[cfg.chan_config];
1869  avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
1870  avctx->channel_layout = chan_layout[cfg.chan_config];
1871 
1872  if (cfg.sample_rate < 16000)
1873  s->syncword = 0xffe00000;
1874  else
1875  s->syncword = 0xfff00000;
1876 
1877  /* Init the first mp3 decoder in standard way, so that all tables get builded
1878  * We replace avctx->priv_data with the context of the first decoder so that
1879  * decode_init() does not have to be changed.
1880  * Other decoders will be initialized here copying data from the first context
1881  */
1882  // Allocate zeroed memory for the first decoder context
1883  s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1884  if (!s->mp3decctx[0])
1885  goto alloc_fail;
1886  // Put decoder context in place to make init_decode() happy
1887  avctx->priv_data = s->mp3decctx[0];
1888  decode_init(avctx);
1889  // Restore mp3on4 context pointer
1890  avctx->priv_data = s;
1891  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1892 
1893  /* Create a separate codec/context for each frame (first is already ok).
1894  * Each frame is 1 or 2 channels - up to 5 frames allowed
1895  */
1896  for (i = 1; i < s->frames; i++) {
1897  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1898  if (!s->mp3decctx[i])
1899  goto alloc_fail;
1900  s->mp3decctx[i]->adu_mode = 1;
1901  s->mp3decctx[i]->avctx = avctx;
1902  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1903  }
1904 
1905  return 0;
1906 alloc_fail:
1907  decode_close_mp3on4(avctx);
1908  return AVERROR(ENOMEM);
1909 }
1910 
1911 
1912 static void flush_mp3on4(AVCodecContext *avctx)
1913 {
1914  int i;
1915  MP3On4DecodeContext *s = avctx->priv_data;
1916 
1917  for (i = 0; i < s->frames; i++)
1918  mp_flush(s->mp3decctx[i]);
1919 }
1920 
1921 
1922 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1923  int *got_frame_ptr, AVPacket *avpkt)
1924 {
1925  AVFrame *frame = data;
1926  const uint8_t *buf = avpkt->data;
1927  int buf_size = avpkt->size;
1928  MP3On4DecodeContext *s = avctx->priv_data;
1929  MPADecodeContext *m;
1930  int fsize, len = buf_size, out_size = 0;
1931  uint32_t header;
1932  OUT_INT **out_samples;
1933  OUT_INT *outptr[2];
1934  int fr, ch, ret;
1935 
1936  /* get output buffer */
1937  frame->nb_samples = MPA_FRAME_SIZE;
1938  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1939  return ret;
1940  out_samples = (OUT_INT **)frame->extended_data;
1941 
1942  // Discard too short frames
1943  if (buf_size < HEADER_SIZE)
1944  return AVERROR_INVALIDDATA;
1945 
1946  avctx->bit_rate = 0;
1947 
1948  ch = 0;
1949  for (fr = 0; fr < s->frames; fr++) {
1950  fsize = AV_RB16(buf) >> 4;
1951  fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1952  m = s->mp3decctx[fr];
1953  av_assert1(m);
1954 
1955  if (fsize < HEADER_SIZE) {
1956  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1957  return AVERROR_INVALIDDATA;
1958  }
1959  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1960 
1961  if (ff_mpa_check_header(header) < 0) // Bad header, discard block
1962  break;
1963 
1964  avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
1965 
1966  if (ch + m->nb_channels > avctx->channels || s->coff[fr] + m->nb_channels > avctx->channels) {
1967  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1968  "channel count\n");
1969  return AVERROR_INVALIDDATA;
1970  }
1971  ch += m->nb_channels;
1972 
1973  outptr[0] = out_samples[s->coff[fr]];
1974  if (m->nb_channels > 1)
1975  outptr[1] = out_samples[s->coff[fr] + 1];
1976 
1977  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
1978  return ret;
1979 
1980  out_size += ret;
1981  buf += fsize;
1982  len -= fsize;
1983 
1984  avctx->bit_rate += m->bit_rate;
1985  }
1986 
1987  /* update codec info */
1988  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1989 
1990  frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1991  *got_frame_ptr = 1;
1992 
1993  return buf_size;
1994 }
1995 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
1996 
1997 #if !CONFIG_FLOAT
1998 #if CONFIG_MP1_DECODER
1999 AVCodec ff_mp1_decoder = {
2000  .name = "mp1",
2001  .type = AVMEDIA_TYPE_AUDIO,
2002  .id = AV_CODEC_ID_MP1,
2003  .priv_data_size = sizeof(MPADecodeContext),
2004  .init = decode_init,
2005  .decode = decode_frame,
2006  .capabilities = CODEC_CAP_DR1,
2007  .flush = flush,
2008  .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
2009  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2010  AV_SAMPLE_FMT_S16,
2011  AV_SAMPLE_FMT_NONE },
2012 };
2013 #endif
2014 #if CONFIG_MP2_DECODER
2015 AVCodec ff_mp2_decoder = {
2016  .name = "mp2",
2017  .type = AVMEDIA_TYPE_AUDIO,
2018  .id = AV_CODEC_ID_MP2,
2019  .priv_data_size = sizeof(MPADecodeContext),
2020  .init = decode_init,
2021  .decode = decode_frame,
2022  .capabilities = CODEC_CAP_DR1,
2023  .flush = flush,
2024  .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2025  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2026  AV_SAMPLE_FMT_S16,
2027  AV_SAMPLE_FMT_NONE },
2028 };
2029 #endif
2030 #if CONFIG_MP3_DECODER
2031 AVCodec ff_mp3_decoder = {
2032  .name = "mp3",
2033  .type = AVMEDIA_TYPE_AUDIO,
2034  .id = AV_CODEC_ID_MP3,
2035  .priv_data_size = sizeof(MPADecodeContext),
2036  .init = decode_init,
2037  .decode = decode_frame,
2038  .capabilities = CODEC_CAP_DR1,
2039  .flush = flush,
2040  .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2041  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2042  AV_SAMPLE_FMT_S16,
2043  AV_SAMPLE_FMT_NONE },
2044 };
2045 #endif
2046 #if CONFIG_MP3ADU_DECODER
2047 AVCodec ff_mp3adu_decoder = {
2048  .name = "mp3adu",
2049  .type = AVMEDIA_TYPE_AUDIO,
2050  .id = AV_CODEC_ID_MP3ADU,
2051  .priv_data_size = sizeof(MPADecodeContext),
2052  .init = decode_init,
2053  .decode = decode_frame_adu,
2054  .capabilities = CODEC_CAP_DR1,
2055  .flush = flush,
2056  .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2057  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2058  AV_SAMPLE_FMT_S16,
2059  AV_SAMPLE_FMT_NONE },
2060 };
2061 #endif
2062 #if CONFIG_MP3ON4_DECODER
2063 AVCodec ff_mp3on4_decoder = {
2064  .name = "mp3on4",
2065  .type = AVMEDIA_TYPE_AUDIO,
2066  .id = AV_CODEC_ID_MP3ON4,
2067  .priv_data_size = sizeof(MP3On4DecodeContext),
2068  .init = decode_init_mp3on4,
2069  .close = decode_close_mp3on4,
2070  .decode = decode_frame_mp3on4,
2071  .capabilities = CODEC_CAP_DR1,
2072  .flush = flush_mp3on4,
2073  .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
2074  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2075  AV_SAMPLE_FMT_NONE },
2076 };
2077 #endif
2078 #endif
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