FFmpeg
 All Data Structures Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
mpegaudiodec.c
Go to the documentation of this file.
1 /*
2  * MPEG Audio decoder
3  * Copyright (c) 2001, 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * MPEG Audio decoder
25  */
26 
27 #include "libavutil/avassert.h"
28 #include "libavutil/channel_layout.h"
29 #include "libavutil/libm.h"
30 #include "avcodec.h"
31 #include "get_bits.h"
32 #include "internal.h"
33 #include "mathops.h"
34 #include "mpegaudiodsp.h"
35 #include "dsputil.h"
36 
37 /*
38  * TODO:
39  * - test lsf / mpeg25 extensively.
40  */
41 
42 #include "mpegaudio.h"
43 #include "mpegaudiodecheader.h"
44 
45 #define BACKSTEP_SIZE 512
46 #define EXTRABYTES 24
47 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
48 
49 /* layer 3 "granule" */
50 typedef struct GranuleDef {
51  uint8_t scfsi;
52  int part2_3_length;
53  int big_values;
54  int global_gain;
55  int scalefac_compress;
56  uint8_t block_type;
57  uint8_t switch_point;
58  int table_select[3];
59  int subblock_gain[3];
60  uint8_t scalefac_scale;
61  uint8_t count1table_select;
62  int region_size[3]; /* number of huffman codes in each region */
63  int preflag;
64  int short_start, long_end; /* long/short band indexes */
65  uint8_t scale_factors[40];
66  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
67 } GranuleDef;
68 
69 typedef struct MPADecodeContext {
70  MPA_DECODE_HEADER
71  uint8_t last_buf[LAST_BUF_SIZE];
72  int last_buf_size;
73  /* next header (used in free format parsing) */
74  uint32_t free_format_next_header;
75  GetBitContext gb;
76  GetBitContext in_gb;
77  DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
78  int synth_buf_offset[MPA_MAX_CHANNELS];
79  DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
80  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
81  GranuleDef granules[2][2]; /* Used in Layer 3 */
82  int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
83  int dither_state;
84  int err_recognition;
85  AVCodecContext* avctx;
86  MPADSPContext mpadsp;
87  DSPContext dsp;
88  AVFrame frame;
89 } MPADecodeContext;
90 
91 #if CONFIG_FLOAT
92 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
93 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
94 # define FIXR(x) ((float)(x))
95 # define FIXHR(x) ((float)(x))
96 # define MULH3(x, y, s) ((s)*(y)*(x))
97 # define MULLx(x, y, s) ((y)*(x))
98 # define RENAME(a) a ## _float
99 # define OUT_FMT AV_SAMPLE_FMT_FLT
100 # define OUT_FMT_P AV_SAMPLE_FMT_FLTP
101 #else
102 # define SHR(a,b) ((a)>>(b))
103 /* WARNING: only correct for positive numbers */
104 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
105 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
106 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
107 # define MULH3(x, y, s) MULH((s)*(x), y)
108 # define MULLx(x, y, s) MULL(x,y,s)
109 # define RENAME(a) a ## _fixed
110 # define OUT_FMT AV_SAMPLE_FMT_S16
111 # define OUT_FMT_P AV_SAMPLE_FMT_S16P
112 #endif
113 
114 /****************/
115 
116 #define HEADER_SIZE 4
117 
118 #include "mpegaudiodata.h"
119 #include "mpegaudiodectab.h"
120 
121 /* vlc structure for decoding layer 3 huffman tables */
122 static VLC huff_vlc[16];
123 static VLC_TYPE huff_vlc_tables[
124  0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
125  142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
126  ][2];
127 static const int huff_vlc_tables_sizes[16] = {
128  0, 128, 128, 128, 130, 128, 154, 166,
129  142, 204, 190, 170, 542, 460, 662, 414
130 };
131 static VLC huff_quad_vlc[2];
132 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
133 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
134 /* computed from band_size_long */
135 static uint16_t band_index_long[9][23];
136 #include "mpegaudio_tablegen.h"
137 /* intensity stereo coef table */
138 static INTFLOAT is_table[2][16];
139 static INTFLOAT is_table_lsf[2][2][16];
140 static INTFLOAT csa_table[8][4];
141 
142 static int16_t division_tab3[1<<6 ];
143 static int16_t division_tab5[1<<8 ];
144 static int16_t division_tab9[1<<11];
145 
146 static int16_t * const division_tabs[4] = {
147  division_tab3, division_tab5, NULL, division_tab9
148 };
149 
150 /* lower 2 bits: modulo 3, higher bits: shift */
151 static uint16_t scale_factor_modshift[64];
152 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
153 static int32_t scale_factor_mult[15][3];
154 /* mult table for layer 2 group quantization */
155 
156 #define SCALE_GEN(v) \
157 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
158 
159 static const int32_t scale_factor_mult2[3][3] = {
160  SCALE_GEN(4.0 / 3.0), /* 3 steps */
161  SCALE_GEN(4.0 / 5.0), /* 5 steps */
162  SCALE_GEN(4.0 / 9.0), /* 9 steps */
163 };
164 
165 /**
166  * Convert region offsets to region sizes and truncate
167  * size to big_values.
168  */
169 static void ff_region_offset2size(GranuleDef *g)
170 {
171  int i, k, j = 0;
172  g->region_size[2] = 576 / 2;
173  for (i = 0; i < 3; i++) {
174  k = FFMIN(g->region_size[i], g->big_values);
175  g->region_size[i] = k - j;
176  j = k;
177  }
178 }
179 
180 static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)
181 {
182  if (g->block_type == 2) {
183  if (s->sample_rate_index != 8)
184  g->region_size[0] = (36 / 2);
185  else
186  g->region_size[0] = (72 / 2);
187  } else {
188  if (s->sample_rate_index <= 2)
189  g->region_size[0] = (36 / 2);
190  else if (s->sample_rate_index != 8)
191  g->region_size[0] = (54 / 2);
192  else
193  g->region_size[0] = (108 / 2);
194  }
195  g->region_size[1] = (576 / 2);
196 }
197 
198 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
199 {
200  int l;
201  g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
202  /* should not overflow */
203  l = FFMIN(ra1 + ra2 + 2, 22);
204  g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
205 }
206 
207 static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
208 {
209  if (g->block_type == 2) {
210  if (g->switch_point) {
211  if(s->sample_rate_index == 8)
212  av_log_ask_for_sample(s->avctx, "switch point in 8khz\n");
213  /* if switched mode, we handle the 36 first samples as
214  long blocks. For 8000Hz, we handle the 72 first
215  exponents as long blocks */
216  if (s->sample_rate_index <= 2)
217  g->long_end = 8;
218  else
219  g->long_end = 6;
220 
221  g->short_start = 3;
222  } else {
223  g->long_end = 0;
224  g->short_start = 0;
225  }
226  } else {
227  g->short_start = 13;
228  g->long_end = 22;
229  }
230 }
231 
232 /* layer 1 unscaling */
233 /* n = number of bits of the mantissa minus 1 */
234 static inline int l1_unscale(int n, int mant, int scale_factor)
235 {
236  int shift, mod;
237  int64_t val;
238 
239  shift = scale_factor_modshift[scale_factor];
240  mod = shift & 3;
241  shift >>= 2;
242  val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
243  shift += n;
244  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
245  return (int)((val + (1LL << (shift - 1))) >> shift);
246 }
247 
248 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
249 {
250  int shift, mod, val;
251 
252  shift = scale_factor_modshift[scale_factor];
253  mod = shift & 3;
254  shift >>= 2;
255 
256  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
257  /* NOTE: at this point, 0 <= shift <= 21 */
258  if (shift > 0)
259  val = (val + (1 << (shift - 1))) >> shift;
260  return val;
261 }
262 
263 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
264 static inline int l3_unscale(int value, int exponent)
265 {
266  unsigned int m;
267  int e;
268 
269  e = table_4_3_exp [4 * value + (exponent & 3)];
270  m = table_4_3_value[4 * value + (exponent & 3)];
271  e -= exponent >> 2;
272 #ifdef DEBUG
273  if(e < 1)
274  av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
275 #endif
276  if (e > 31)
277  return 0;
278  m = (m + (1 << (e - 1))) >> e;
279 
280  return m;
281 }
282 
283 static av_cold void decode_init_static(void)
284 {
285  int i, j, k;
286  int offset;
287 
288  /* scale factors table for layer 1/2 */
289  for (i = 0; i < 64; i++) {
290  int shift, mod;
291  /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
292  shift = i / 3;
293  mod = i % 3;
294  scale_factor_modshift[i] = mod | (shift << 2);
295  }
296 
297  /* scale factor multiply for layer 1 */
298  for (i = 0; i < 15; i++) {
299  int n, norm;
300  n = i + 2;
301  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
302  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
303  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
304  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
305  av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
306  scale_factor_mult[i][0],
307  scale_factor_mult[i][1],
308  scale_factor_mult[i][2]);
309  }
310 
311  RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
312 
313  /* huffman decode tables */
314  offset = 0;
315  for (i = 1; i < 16; i++) {
316  const HuffTable *h = &mpa_huff_tables[i];
317  int xsize, x, y;
318  uint8_t tmp_bits [512] = { 0 };
319  uint16_t tmp_codes[512] = { 0 };
320 
321  xsize = h->xsize;
322 
323  j = 0;
324  for (x = 0; x < xsize; x++) {
325  for (y = 0; y < xsize; y++) {
326  tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
327  tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
328  }
329  }
330 
331  /* XXX: fail test */
332  huff_vlc[i].table = huff_vlc_tables+offset;
333  huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
334  init_vlc(&huff_vlc[i], 7, 512,
335  tmp_bits, 1, 1, tmp_codes, 2, 2,
336  INIT_VLC_USE_NEW_STATIC);
337  offset += huff_vlc_tables_sizes[i];
338  }
339  av_assert0(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
340 
341  offset = 0;
342  for (i = 0; i < 2; i++) {
343  huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
344  huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
345  init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
346  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
347  INIT_VLC_USE_NEW_STATIC);
348  offset += huff_quad_vlc_tables_sizes[i];
349  }
350  av_assert0(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
351 
352  for (i = 0; i < 9; i++) {
353  k = 0;
354  for (j = 0; j < 22; j++) {
355  band_index_long[i][j] = k;
356  k += band_size_long[i][j];
357  }
358  band_index_long[i][22] = k;
359  }
360 
361  /* compute n ^ (4/3) and store it in mantissa/exp format */
362 
363  mpegaudio_tableinit();
364 
365  for (i = 0; i < 4; i++) {
366  if (ff_mpa_quant_bits[i] < 0) {
367  for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
368  int val1, val2, val3, steps;
369  int val = j;
370  steps = ff_mpa_quant_steps[i];
371  val1 = val % steps;
372  val /= steps;
373  val2 = val % steps;
374  val3 = val / steps;
375  division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
376  }
377  }
378  }
379 
380 
381  for (i = 0; i < 7; i++) {
382  float f;
383  INTFLOAT v;
384  if (i != 6) {
385  f = tan((double)i * M_PI / 12.0);
386  v = FIXR(f / (1.0 + f));
387  } else {
388  v = FIXR(1.0);
389  }
390  is_table[0][ i] = v;
391  is_table[1][6 - i] = v;
392  }
393  /* invalid values */
394  for (i = 7; i < 16; i++)
395  is_table[0][i] = is_table[1][i] = 0.0;
396 
397  for (i = 0; i < 16; i++) {
398  double f;
399  int e, k;
400 
401  for (j = 0; j < 2; j++) {
402  e = -(j + 1) * ((i + 1) >> 1);
403  f = exp2(e / 4.0);
404  k = i & 1;
405  is_table_lsf[j][k ^ 1][i] = FIXR(f);
406  is_table_lsf[j][k ][i] = FIXR(1.0);
407  av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
408  i, j, (float) is_table_lsf[j][0][i],
409  (float) is_table_lsf[j][1][i]);
410  }
411  }
412 
413  for (i = 0; i < 8; i++) {
414  float ci, cs, ca;
415  ci = ci_table[i];
416  cs = 1.0 / sqrt(1.0 + ci * ci);
417  ca = cs * ci;
418 #if !CONFIG_FLOAT
419  csa_table[i][0] = FIXHR(cs/4);
420  csa_table[i][1] = FIXHR(ca/4);
421  csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
422  csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
423 #else
424  csa_table[i][0] = cs;
425  csa_table[i][1] = ca;
426  csa_table[i][2] = ca + cs;
427  csa_table[i][3] = ca - cs;
428 #endif
429  }
430 }
431 
432 static av_cold int decode_init(AVCodecContext * avctx)
433 {
434  static int initialized_tables = 0;
435  MPADecodeContext *s = avctx->priv_data;
436 
437  if (!initialized_tables) {
438  decode_init_static();
439  initialized_tables = 1;
440  }
441 
442  s->avctx = avctx;
443 
444  ff_mpadsp_init(&s->mpadsp);
445  ff_dsputil_init(&s->dsp, avctx);
446 
447  if (avctx->request_sample_fmt == OUT_FMT &&
448  avctx->codec_id != AV_CODEC_ID_MP3ON4)
449  avctx->sample_fmt = OUT_FMT;
450  else
451  avctx->sample_fmt = OUT_FMT_P;
452  s->err_recognition = avctx->err_recognition;
453 
454  if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
455  s->adu_mode = 1;
456 
457  avcodec_get_frame_defaults(&s->frame);
458  avctx->coded_frame = &s->frame;
459 
460  return 0;
461 }
462 
463 #define C3 FIXHR(0.86602540378443864676/2)
464 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
465 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
466 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
467 
468 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
469  cases. */
470 static void imdct12(INTFLOAT *out, INTFLOAT *in)
471 {
472  INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
473 
474  in0 = in[0*3];
475  in1 = in[1*3] + in[0*3];
476  in2 = in[2*3] + in[1*3];
477  in3 = in[3*3] + in[2*3];
478  in4 = in[4*3] + in[3*3];
479  in5 = in[5*3] + in[4*3];
480  in5 += in3;
481  in3 += in1;
482 
483  in2 = MULH3(in2, C3, 2);
484  in3 = MULH3(in3, C3, 4);
485 
486  t1 = in0 - in4;
487  t2 = MULH3(in1 - in5, C4, 2);
488 
489  out[ 7] =
490  out[10] = t1 + t2;
491  out[ 1] =
492  out[ 4] = t1 - t2;
493 
494  in0 += SHR(in4, 1);
495  in4 = in0 + in2;
496  in5 += 2*in1;
497  in1 = MULH3(in5 + in3, C5, 1);
498  out[ 8] =
499  out[ 9] = in4 + in1;
500  out[ 2] =
501  out[ 3] = in4 - in1;
502 
503  in0 -= in2;
504  in5 = MULH3(in5 - in3, C6, 2);
505  out[ 0] =
506  out[ 5] = in0 - in5;
507  out[ 6] =
508  out[11] = in0 + in5;
509 }
510 
511 /* return the number of decoded frames */
512 static int mp_decode_layer1(MPADecodeContext *s)
513 {
514  int bound, i, v, n, ch, j, mant;
515  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
516  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
517 
518  if (s->mode == MPA_JSTEREO)
519  bound = (s->mode_ext + 1) * 4;
520  else
521  bound = SBLIMIT;
522 
523  /* allocation bits */
524  for (i = 0; i < bound; i++) {
525  for (ch = 0; ch < s->nb_channels; ch++) {
526  allocation[ch][i] = get_bits(&s->gb, 4);
527  }
528  }
529  for (i = bound; i < SBLIMIT; i++)
530  allocation[0][i] = get_bits(&s->gb, 4);
531 
532  /* scale factors */
533  for (i = 0; i < bound; i++) {
534  for (ch = 0; ch < s->nb_channels; ch++) {
535  if (allocation[ch][i])
536  scale_factors[ch][i] = get_bits(&s->gb, 6);
537  }
538  }
539  for (i = bound; i < SBLIMIT; i++) {
540  if (allocation[0][i]) {
541  scale_factors[0][i] = get_bits(&s->gb, 6);
542  scale_factors[1][i] = get_bits(&s->gb, 6);
543  }
544  }
545 
546  /* compute samples */
547  for (j = 0; j < 12; j++) {
548  for (i = 0; i < bound; i++) {
549  for (ch = 0; ch < s->nb_channels; ch++) {
550  n = allocation[ch][i];
551  if (n) {
552  mant = get_bits(&s->gb, n + 1);
553  v = l1_unscale(n, mant, scale_factors[ch][i]);
554  } else {
555  v = 0;
556  }
557  s->sb_samples[ch][j][i] = v;
558  }
559  }
560  for (i = bound; i < SBLIMIT; i++) {
561  n = allocation[0][i];
562  if (n) {
563  mant = get_bits(&s->gb, n + 1);
564  v = l1_unscale(n, mant, scale_factors[0][i]);
565  s->sb_samples[0][j][i] = v;
566  v = l1_unscale(n, mant, scale_factors[1][i]);
567  s->sb_samples[1][j][i] = v;
568  } else {
569  s->sb_samples[0][j][i] = 0;
570  s->sb_samples[1][j][i] = 0;
571  }
572  }
573  }
574  return 12;
575 }
576 
577 static int mp_decode_layer2(MPADecodeContext *s)
578 {
579  int sblimit; /* number of used subbands */
580  const unsigned char *alloc_table;
581  int table, bit_alloc_bits, i, j, ch, bound, v;
582  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
583  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
584  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
585  int scale, qindex, bits, steps, k, l, m, b;
586 
587  /* select decoding table */
588  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
589  s->sample_rate, s->lsf);
590  sblimit = ff_mpa_sblimit_table[table];
591  alloc_table = ff_mpa_alloc_tables[table];
592 
593  if (s->mode == MPA_JSTEREO)
594  bound = (s->mode_ext + 1) * 4;
595  else
596  bound = sblimit;
597 
598  av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
599 
600  /* sanity check */
601  if (bound > sblimit)
602  bound = sblimit;
603 
604  /* parse bit allocation */
605  j = 0;
606  for (i = 0; i < bound; i++) {
607  bit_alloc_bits = alloc_table[j];
608  for (ch = 0; ch < s->nb_channels; ch++)
609  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
610  j += 1 << bit_alloc_bits;
611  }
612  for (i = bound; i < sblimit; i++) {
613  bit_alloc_bits = alloc_table[j];
614  v = get_bits(&s->gb, bit_alloc_bits);
615  bit_alloc[0][i] = v;
616  bit_alloc[1][i] = v;
617  j += 1 << bit_alloc_bits;
618  }
619 
620  /* scale codes */
621  for (i = 0; i < sblimit; i++) {
622  for (ch = 0; ch < s->nb_channels; ch++) {
623  if (bit_alloc[ch][i])
624  scale_code[ch][i] = get_bits(&s->gb, 2);
625  }
626  }
627 
628  /* scale factors */
629  for (i = 0; i < sblimit; i++) {
630  for (ch = 0; ch < s->nb_channels; ch++) {
631  if (bit_alloc[ch][i]) {
632  sf = scale_factors[ch][i];
633  switch (scale_code[ch][i]) {
634  default:
635  case 0:
636  sf[0] = get_bits(&s->gb, 6);
637  sf[1] = get_bits(&s->gb, 6);
638  sf[2] = get_bits(&s->gb, 6);
639  break;
640  case 2:
641  sf[0] = get_bits(&s->gb, 6);
642  sf[1] = sf[0];
643  sf[2] = sf[0];
644  break;
645  case 1:
646  sf[0] = get_bits(&s->gb, 6);
647  sf[2] = get_bits(&s->gb, 6);
648  sf[1] = sf[0];
649  break;
650  case 3:
651  sf[0] = get_bits(&s->gb, 6);
652  sf[2] = get_bits(&s->gb, 6);
653  sf[1] = sf[2];
654  break;
655  }
656  }
657  }
658  }
659 
660  /* samples */
661  for (k = 0; k < 3; k++) {
662  for (l = 0; l < 12; l += 3) {
663  j = 0;
664  for (i = 0; i < bound; i++) {
665  bit_alloc_bits = alloc_table[j];
666  for (ch = 0; ch < s->nb_channels; ch++) {
667  b = bit_alloc[ch][i];
668  if (b) {
669  scale = scale_factors[ch][i][k];
670  qindex = alloc_table[j+b];
671  bits = ff_mpa_quant_bits[qindex];
672  if (bits < 0) {
673  int v2;
674  /* 3 values at the same time */
675  v = get_bits(&s->gb, -bits);
676  v2 = division_tabs[qindex][v];
677  steps = ff_mpa_quant_steps[qindex];
678 
679  s->sb_samples[ch][k * 12 + l + 0][i] =
680  l2_unscale_group(steps, v2 & 15, scale);
681  s->sb_samples[ch][k * 12 + l + 1][i] =
682  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
683  s->sb_samples[ch][k * 12 + l + 2][i] =
684  l2_unscale_group(steps, v2 >> 8 , scale);
685  } else {
686  for (m = 0; m < 3; m++) {
687  v = get_bits(&s->gb, bits);
688  v = l1_unscale(bits - 1, v, scale);
689  s->sb_samples[ch][k * 12 + l + m][i] = v;
690  }
691  }
692  } else {
693  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
694  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
695  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
696  }
697  }
698  /* next subband in alloc table */
699  j += 1 << bit_alloc_bits;
700  }
701  /* XXX: find a way to avoid this duplication of code */
702  for (i = bound; i < sblimit; i++) {
703  bit_alloc_bits = alloc_table[j];
704  b = bit_alloc[0][i];
705  if (b) {
706  int mant, scale0, scale1;
707  scale0 = scale_factors[0][i][k];
708  scale1 = scale_factors[1][i][k];
709  qindex = alloc_table[j+b];
710  bits = ff_mpa_quant_bits[qindex];
711  if (bits < 0) {
712  /* 3 values at the same time */
713  v = get_bits(&s->gb, -bits);
714  steps = ff_mpa_quant_steps[qindex];
715  mant = v % steps;
716  v = v / steps;
717  s->sb_samples[0][k * 12 + l + 0][i] =
718  l2_unscale_group(steps, mant, scale0);
719  s->sb_samples[1][k * 12 + l + 0][i] =
720  l2_unscale_group(steps, mant, scale1);
721  mant = v % steps;
722  v = v / steps;
723  s->sb_samples[0][k * 12 + l + 1][i] =
724  l2_unscale_group(steps, mant, scale0);
725  s->sb_samples[1][k * 12 + l + 1][i] =
726  l2_unscale_group(steps, mant, scale1);
727  s->sb_samples[0][k * 12 + l + 2][i] =
728  l2_unscale_group(steps, v, scale0);
729  s->sb_samples[1][k * 12 + l + 2][i] =
730  l2_unscale_group(steps, v, scale1);
731  } else {
732  for (m = 0; m < 3; m++) {
733  mant = get_bits(&s->gb, bits);
734  s->sb_samples[0][k * 12 + l + m][i] =
735  l1_unscale(bits - 1, mant, scale0);
736  s->sb_samples[1][k * 12 + l + m][i] =
737  l1_unscale(bits - 1, mant, scale1);
738  }
739  }
740  } else {
741  s->sb_samples[0][k * 12 + l + 0][i] = 0;
742  s->sb_samples[0][k * 12 + l + 1][i] = 0;
743  s->sb_samples[0][k * 12 + l + 2][i] = 0;
744  s->sb_samples[1][k * 12 + l + 0][i] = 0;
745  s->sb_samples[1][k * 12 + l + 1][i] = 0;
746  s->sb_samples[1][k * 12 + l + 2][i] = 0;
747  }
748  /* next subband in alloc table */
749  j += 1 << bit_alloc_bits;
750  }
751  /* fill remaining samples to zero */
752  for (i = sblimit; i < SBLIMIT; i++) {
753  for (ch = 0; ch < s->nb_channels; ch++) {
754  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
755  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
756  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
757  }
758  }
759  }
760  }
761  return 3 * 12;
762 }
763 
764 #define SPLIT(dst,sf,n) \
765  if (n == 3) { \
766  int m = (sf * 171) >> 9; \
767  dst = sf - 3 * m; \
768  sf = m; \
769  } else if (n == 4) { \
770  dst = sf & 3; \
771  sf >>= 2; \
772  } else if (n == 5) { \
773  int m = (sf * 205) >> 10; \
774  dst = sf - 5 * m; \
775  sf = m; \
776  } else if (n == 6) { \
777  int m = (sf * 171) >> 10; \
778  dst = sf - 6 * m; \
779  sf = m; \
780  } else { \
781  dst = 0; \
782  }
783 
784 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
785  int n3)
786 {
787  SPLIT(slen[3], sf, n3)
788  SPLIT(slen[2], sf, n2)
789  SPLIT(slen[1], sf, n1)
790  slen[0] = sf;
791 }
792 
793 static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
794  int16_t *exponents)
795 {
796  const uint8_t *bstab, *pretab;
797  int len, i, j, k, l, v0, shift, gain, gains[3];
798  int16_t *exp_ptr;
799 
800  exp_ptr = exponents;
801  gain = g->global_gain - 210;
802  shift = g->scalefac_scale + 1;
803 
804  bstab = band_size_long[s->sample_rate_index];
805  pretab = mpa_pretab[g->preflag];
806  for (i = 0; i < g->long_end; i++) {
807  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
808  len = bstab[i];
809  for (j = len; j > 0; j--)
810  *exp_ptr++ = v0;
811  }
812 
813  if (g->short_start < 13) {
814  bstab = band_size_short[s->sample_rate_index];
815  gains[0] = gain - (g->subblock_gain[0] << 3);
816  gains[1] = gain - (g->subblock_gain[1] << 3);
817  gains[2] = gain - (g->subblock_gain[2] << 3);
818  k = g->long_end;
819  for (i = g->short_start; i < 13; i++) {
820  len = bstab[i];
821  for (l = 0; l < 3; l++) {
822  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
823  for (j = len; j > 0; j--)
824  *exp_ptr++ = v0;
825  }
826  }
827  }
828 }
829 
830 /* handle n = 0 too */
831 static inline int get_bitsz(GetBitContext *s, int n)
832 {
833  return n ? get_bits(s, n) : 0;
834 }
835 
836 
837 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
838  int *end_pos2)
839 {
840  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
841  s->gb = s->in_gb;
842  s->in_gb.buffer = NULL;
843  av_assert2((get_bits_count(&s->gb) & 7) == 0);
844  skip_bits_long(&s->gb, *pos - *end_pos);
845  *end_pos2 =
846  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
847  *pos = get_bits_count(&s->gb);
848  }
849 }
850 
851 /* Following is a optimized code for
852  INTFLOAT v = *src
853  if(get_bits1(&s->gb))
854  v = -v;
855  *dst = v;
856 */
857 #if CONFIG_FLOAT
858 #define READ_FLIP_SIGN(dst,src) \
859  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
860  AV_WN32A(dst, v);
861 #else
862 #define READ_FLIP_SIGN(dst,src) \
863  v = -get_bits1(&s->gb); \
864  *(dst) = (*(src) ^ v) - v;
865 #endif
866 
867 static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
868  int16_t *exponents, int end_pos2)
869 {
870  int s_index;
871  int i;
872  int last_pos, bits_left;
873  VLC *vlc;
874  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
875 
876  /* low frequencies (called big values) */
877  s_index = 0;
878  for (i = 0; i < 3; i++) {
879  int j, k, l, linbits;
880  j = g->region_size[i];
881  if (j == 0)
882  continue;
883  /* select vlc table */
884  k = g->table_select[i];
885  l = mpa_huff_data[k][0];
886  linbits = mpa_huff_data[k][1];
887  vlc = &huff_vlc[l];
888 
889  if (!l) {
890  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
891  s_index += 2 * j;
892  continue;
893  }
894 
895  /* read huffcode and compute each couple */
896  for (; j > 0; j--) {
897  int exponent, x, y;
898  int v;
899  int pos = get_bits_count(&s->gb);
900 
901  if (pos >= end_pos){
902  switch_buffer(s, &pos, &end_pos, &end_pos2);
903  if (pos >= end_pos)
904  break;
905  }
906  y = get_vlc2(&s->gb, vlc->table, 7, 3);
907 
908  if (!y) {
909  g->sb_hybrid[s_index ] =
910  g->sb_hybrid[s_index+1] = 0;
911  s_index += 2;
912  continue;
913  }
914 
915  exponent= exponents[s_index];
916 
917  av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
918  i, g->region_size[i] - j, x, y, exponent);
919  if (y & 16) {
920  x = y >> 5;
921  y = y & 0x0f;
922  if (x < 15) {
923  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
924  } else {
925  x += get_bitsz(&s->gb, linbits);
926  v = l3_unscale(x, exponent);
927  if (get_bits1(&s->gb))
928  v = -v;
929  g->sb_hybrid[s_index] = v;
930  }
931  if (y < 15) {
932  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
933  } else {
934  y += get_bitsz(&s->gb, linbits);
935  v = l3_unscale(y, exponent);
936  if (get_bits1(&s->gb))
937  v = -v;
938  g->sb_hybrid[s_index+1] = v;
939  }
940  } else {
941  x = y >> 5;
942  y = y & 0x0f;
943  x += y;
944  if (x < 15) {
945  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
946  } else {
947  x += get_bitsz(&s->gb, linbits);
948  v = l3_unscale(x, exponent);
949  if (get_bits1(&s->gb))
950  v = -v;
951  g->sb_hybrid[s_index+!!y] = v;
952  }
953  g->sb_hybrid[s_index + !y] = 0;
954  }
955  s_index += 2;
956  }
957  }
958 
959  /* high frequencies */
960  vlc = &huff_quad_vlc[g->count1table_select];
961  last_pos = 0;
962  while (s_index <= 572) {
963  int pos, code;
964  pos = get_bits_count(&s->gb);
965  if (pos >= end_pos) {
966  if (pos > end_pos2 && last_pos) {
967  /* some encoders generate an incorrect size for this
968  part. We must go back into the data */
969  s_index -= 4;
970  skip_bits_long(&s->gb, last_pos - pos);
971  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
972  if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
973  s_index=0;
974  break;
975  }
976  switch_buffer(s, &pos, &end_pos, &end_pos2);
977  if (pos >= end_pos)
978  break;
979  }
980  last_pos = pos;
981 
982  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
983  av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
984  g->sb_hybrid[s_index+0] =
985  g->sb_hybrid[s_index+1] =
986  g->sb_hybrid[s_index+2] =
987  g->sb_hybrid[s_index+3] = 0;
988  while (code) {
989  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
990  int v;
991  int pos = s_index + idxtab[code];
992  code ^= 8 >> idxtab[code];
993  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
994  }
995  s_index += 4;
996  }
997  /* skip extension bits */
998  bits_left = end_pos2 - get_bits_count(&s->gb);
999  if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
1000  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1001  s_index=0;
1002  } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
1003  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1004  s_index = 0;
1005  }
1006  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
1007  skip_bits_long(&s->gb, bits_left);
1008 
1009  i = get_bits_count(&s->gb);
1010  switch_buffer(s, &i, &end_pos, &end_pos2);
1011 
1012  return 0;
1013 }
1014 
1015 /* Reorder short blocks from bitstream order to interleaved order. It
1016  would be faster to do it in parsing, but the code would be far more
1017  complicated */
1018 static void reorder_block(MPADecodeContext *s, GranuleDef *g)
1019 {
1020  int i, j, len;
1021  INTFLOAT *ptr, *dst, *ptr1;
1022  INTFLOAT tmp[576];
1023 
1024  if (g->block_type != 2)
1025  return;
1026 
1027  if (g->switch_point) {
1028  if (s->sample_rate_index != 8)
1029  ptr = g->sb_hybrid + 36;
1030  else
1031  ptr = g->sb_hybrid + 72;
1032  } else {
1033  ptr = g->sb_hybrid;
1034  }
1035 
1036  for (i = g->short_start; i < 13; i++) {
1037  len = band_size_short[s->sample_rate_index][i];
1038  ptr1 = ptr;
1039  dst = tmp;
1040  for (j = len; j > 0; j--) {
1041  *dst++ = ptr[0*len];
1042  *dst++ = ptr[1*len];
1043  *dst++ = ptr[2*len];
1044  ptr++;
1045  }
1046  ptr += 2 * len;
1047  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1048  }
1049 }
1050 
1051 #define ISQRT2 FIXR(0.70710678118654752440)
1052 
1053 static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
1054 {
1055  int i, j, k, l;
1056  int sf_max, sf, len, non_zero_found;
1057  INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1058  int non_zero_found_short[3];
1059 
1060  /* intensity stereo */
1061  if (s->mode_ext & MODE_EXT_I_STEREO) {
1062  if (!s->lsf) {
1063  is_tab = is_table;
1064  sf_max = 7;
1065  } else {
1066  is_tab = is_table_lsf[g1->scalefac_compress & 1];
1067  sf_max = 16;
1068  }
1069 
1070  tab0 = g0->sb_hybrid + 576;
1071  tab1 = g1->sb_hybrid + 576;
1072 
1073  non_zero_found_short[0] = 0;
1074  non_zero_found_short[1] = 0;
1075  non_zero_found_short[2] = 0;
1076  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1077  for (i = 12; i >= g1->short_start; i--) {
1078  /* for last band, use previous scale factor */
1079  if (i != 11)
1080  k -= 3;
1081  len = band_size_short[s->sample_rate_index][i];
1082  for (l = 2; l >= 0; l--) {
1083  tab0 -= len;
1084  tab1 -= len;
1085  if (!non_zero_found_short[l]) {
1086  /* test if non zero band. if so, stop doing i-stereo */
1087  for (j = 0; j < len; j++) {
1088  if (tab1[j] != 0) {
1089  non_zero_found_short[l] = 1;
1090  goto found1;
1091  }
1092  }
1093  sf = g1->scale_factors[k + l];
1094  if (sf >= sf_max)
1095  goto found1;
1096 
1097  v1 = is_tab[0][sf];
1098  v2 = is_tab[1][sf];
1099  for (j = 0; j < len; j++) {
1100  tmp0 = tab0[j];
1101  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1102  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1103  }
1104  } else {
1105 found1:
1106  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1107  /* lower part of the spectrum : do ms stereo
1108  if enabled */
1109  for (j = 0; j < len; j++) {
1110  tmp0 = tab0[j];
1111  tmp1 = tab1[j];
1112  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1113  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1114  }
1115  }
1116  }
1117  }
1118  }
1119 
1120  non_zero_found = non_zero_found_short[0] |
1121  non_zero_found_short[1] |
1122  non_zero_found_short[2];
1123 
1124  for (i = g1->long_end - 1;i >= 0;i--) {
1125  len = band_size_long[s->sample_rate_index][i];
1126  tab0 -= len;
1127  tab1 -= len;
1128  /* test if non zero band. if so, stop doing i-stereo */
1129  if (!non_zero_found) {
1130  for (j = 0; j < len; j++) {
1131  if (tab1[j] != 0) {
1132  non_zero_found = 1;
1133  goto found2;
1134  }
1135  }
1136  /* for last band, use previous scale factor */
1137  k = (i == 21) ? 20 : i;
1138  sf = g1->scale_factors[k];
1139  if (sf >= sf_max)
1140  goto found2;
1141  v1 = is_tab[0][sf];
1142  v2 = is_tab[1][sf];
1143  for (j = 0; j < len; j++) {
1144  tmp0 = tab0[j];
1145  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1146  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1147  }
1148  } else {
1149 found2:
1150  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1151  /* lower part of the spectrum : do ms stereo
1152  if enabled */
1153  for (j = 0; j < len; j++) {
1154  tmp0 = tab0[j];
1155  tmp1 = tab1[j];
1156  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1157  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1158  }
1159  }
1160  }
1161  }
1162  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1163  /* ms stereo ONLY */
1164  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1165  global gain */
1166 #if CONFIG_FLOAT
1167  s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1168 #else
1169  tab0 = g0->sb_hybrid;
1170  tab1 = g1->sb_hybrid;
1171  for (i = 0; i < 576; i++) {
1172  tmp0 = tab0[i];
1173  tmp1 = tab1[i];
1174  tab0[i] = tmp0 + tmp1;
1175  tab1[i] = tmp0 - tmp1;
1176  }
1177 #endif
1178  }
1179 }
1180 
1181 #if CONFIG_FLOAT
1182 #if HAVE_MIPSFPU
1183 # include "mips/compute_antialias_float.h"
1184 #endif /* HAVE_MIPSFPU */
1185 #else
1186 #if HAVE_MIPSDSPR1
1187 # include "mips/compute_antialias_fixed.h"
1188 #endif /* HAVE_MIPSDSPR1 */
1189 #endif /* CONFIG_FLOAT */
1190 
1191 #ifndef compute_antialias
1192 #if CONFIG_FLOAT
1193 #define AA(j) do { \
1194  float tmp0 = ptr[-1-j]; \
1195  float tmp1 = ptr[ j]; \
1196  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1197  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1198  } while (0)
1199 #else
1200 #define AA(j) do { \
1201  int tmp0 = ptr[-1-j]; \
1202  int tmp1 = ptr[ j]; \
1203  int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1204  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1205  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1206  } while (0)
1207 #endif
1208 
1209 static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
1210 {
1211  INTFLOAT *ptr;
1212  int n, i;
1213 
1214  /* we antialias only "long" bands */
1215  if (g->block_type == 2) {
1216  if (!g->switch_point)
1217  return;
1218  /* XXX: check this for 8000Hz case */
1219  n = 1;
1220  } else {
1221  n = SBLIMIT - 1;
1222  }
1223 
1224  ptr = g->sb_hybrid + 18;
1225  for (i = n; i > 0; i--) {
1226  AA(0);
1227  AA(1);
1228  AA(2);
1229  AA(3);
1230  AA(4);
1231  AA(5);
1232  AA(6);
1233  AA(7);
1234 
1235  ptr += 18;
1236  }
1237 }
1238 #endif /* compute_antialias */
1239 
1240 static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
1241  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1242 {
1243  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1244  INTFLOAT out2[12];
1245  int i, j, mdct_long_end, sblimit;
1246 
1247  /* find last non zero block */
1248  ptr = g->sb_hybrid + 576;
1249  ptr1 = g->sb_hybrid + 2 * 18;
1250  while (ptr >= ptr1) {
1251  int32_t *p;
1252  ptr -= 6;
1253  p = (int32_t*)ptr;
1254  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1255  break;
1256  }
1257  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1258 
1259  if (g->block_type == 2) {
1260  /* XXX: check for 8000 Hz */
1261  if (g->switch_point)
1262  mdct_long_end = 2;
1263  else
1264  mdct_long_end = 0;
1265  } else {
1266  mdct_long_end = sblimit;
1267  }
1268 
1269  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1270  mdct_long_end, g->switch_point,
1271  g->block_type);
1272 
1273  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1274  ptr = g->sb_hybrid + 18 * mdct_long_end;
1275 
1276  for (j = mdct_long_end; j < sblimit; j++) {
1277  /* select frequency inversion */
1278  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1279  out_ptr = sb_samples + j;
1280 
1281  for (i = 0; i < 6; i++) {
1282  *out_ptr = buf[4*i];
1283  out_ptr += SBLIMIT;
1284  }
1285  imdct12(out2, ptr + 0);
1286  for (i = 0; i < 6; i++) {
1287  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1288  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1289  out_ptr += SBLIMIT;
1290  }
1291  imdct12(out2, ptr + 1);
1292  for (i = 0; i < 6; i++) {
1293  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1294  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1295  out_ptr += SBLIMIT;
1296  }
1297  imdct12(out2, ptr + 2);
1298  for (i = 0; i < 6; i++) {
1299  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1300  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1301  buf[4*(i + 6*2)] = 0;
1302  }
1303  ptr += 18;
1304  buf += (j&3) != 3 ? 1 : (4*18-3);
1305  }
1306  /* zero bands */
1307  for (j = sblimit; j < SBLIMIT; j++) {
1308  /* overlap */
1309  out_ptr = sb_samples + j;
1310  for (i = 0; i < 18; i++) {
1311  *out_ptr = buf[4*i];
1312  buf[4*i] = 0;
1313  out_ptr += SBLIMIT;
1314  }
1315  buf += (j&3) != 3 ? 1 : (4*18-3);
1316  }
1317 }
1318 
1319 /* main layer3 decoding function */
1320 static int mp_decode_layer3(MPADecodeContext *s)
1321 {
1322  int nb_granules, main_data_begin;
1323  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1324  GranuleDef *g;
1325  int16_t exponents[576]; //FIXME try INTFLOAT
1326 
1327  /* read side info */
1328  if (s->lsf) {
1329  main_data_begin = get_bits(&s->gb, 8);
1330  skip_bits(&s->gb, s->nb_channels);
1331  nb_granules = 1;
1332  } else {
1333  main_data_begin = get_bits(&s->gb, 9);
1334  if (s->nb_channels == 2)
1335  skip_bits(&s->gb, 3);
1336  else
1337  skip_bits(&s->gb, 5);
1338  nb_granules = 2;
1339  for (ch = 0; ch < s->nb_channels; ch++) {
1340  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1341  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1342  }
1343  }
1344 
1345  for (gr = 0; gr < nb_granules; gr++) {
1346  for (ch = 0; ch < s->nb_channels; ch++) {
1347  av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1348  g = &s->granules[ch][gr];
1349  g->part2_3_length = get_bits(&s->gb, 12);
1350  g->big_values = get_bits(&s->gb, 9);
1351  if (g->big_values > 288) {
1352  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1353  return AVERROR_INVALIDDATA;
1354  }
1355 
1356  g->global_gain = get_bits(&s->gb, 8);
1357  /* if MS stereo only is selected, we precompute the
1358  1/sqrt(2) renormalization factor */
1359  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1360  MODE_EXT_MS_STEREO)
1361  g->global_gain -= 2;
1362  if (s->lsf)
1363  g->scalefac_compress = get_bits(&s->gb, 9);
1364  else
1365  g->scalefac_compress = get_bits(&s->gb, 4);
1366  blocksplit_flag = get_bits1(&s->gb);
1367  if (blocksplit_flag) {
1368  g->block_type = get_bits(&s->gb, 2);
1369  if (g->block_type == 0) {
1370  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1371  return AVERROR_INVALIDDATA;
1372  }
1373  g->switch_point = get_bits1(&s->gb);
1374  for (i = 0; i < 2; i++)
1375  g->table_select[i] = get_bits(&s->gb, 5);
1376  for (i = 0; i < 3; i++)
1377  g->subblock_gain[i] = get_bits(&s->gb, 3);
1378  ff_init_short_region(s, g);
1379  } else {
1380  int region_address1, region_address2;
1381  g->block_type = 0;
1382  g->switch_point = 0;
1383  for (i = 0; i < 3; i++)
1384  g->table_select[i] = get_bits(&s->gb, 5);
1385  /* compute huffman coded region sizes */
1386  region_address1 = get_bits(&s->gb, 4);
1387  region_address2 = get_bits(&s->gb, 3);
1388  av_dlog(s->avctx, "region1=%d region2=%d\n",
1389  region_address1, region_address2);
1390  ff_init_long_region(s, g, region_address1, region_address2);
1391  }
1392  ff_region_offset2size(g);
1393  ff_compute_band_indexes(s, g);
1394 
1395  g->preflag = 0;
1396  if (!s->lsf)
1397  g->preflag = get_bits1(&s->gb);
1398  g->scalefac_scale = get_bits1(&s->gb);
1399  g->count1table_select = get_bits1(&s->gb);
1400  av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1401  g->block_type, g->switch_point);
1402  }
1403  }
1404 
1405  if (!s->adu_mode) {
1406  int skip;
1407  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1408  int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES);
1409  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1410  /* now we get bits from the main_data_begin offset */
1411  av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1412  main_data_begin, s->last_buf_size);
1413 
1414  memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
1415  s->in_gb = s->gb;
1416  init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1417 #if !UNCHECKED_BITSTREAM_READER
1418  s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8;
1419 #endif
1420  s->last_buf_size <<= 3;
1421  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1422  for (ch = 0; ch < s->nb_channels; ch++) {
1423  g = &s->granules[ch][gr];
1424  s->last_buf_size += g->part2_3_length;
1425  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1426  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1427  }
1428  }
1429  skip = s->last_buf_size - 8 * main_data_begin;
1430  if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
1431  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
1432  s->gb = s->in_gb;
1433  s->in_gb.buffer = NULL;
1434  } else {
1435  skip_bits_long(&s->gb, skip);
1436  }
1437  } else {
1438  gr = 0;
1439  }
1440 
1441  for (; gr < nb_granules; gr++) {
1442  for (ch = 0; ch < s->nb_channels; ch++) {
1443  g = &s->granules[ch][gr];
1444  bits_pos = get_bits_count(&s->gb);
1445 
1446  if (!s->lsf) {
1447  uint8_t *sc;
1448  int slen, slen1, slen2;
1449 
1450  /* MPEG1 scale factors */
1451  slen1 = slen_table[0][g->scalefac_compress];
1452  slen2 = slen_table[1][g->scalefac_compress];
1453  av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1454  if (g->block_type == 2) {
1455  n = g->switch_point ? 17 : 18;
1456  j = 0;
1457  if (slen1) {
1458  for (i = 0; i < n; i++)
1459  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1460  } else {
1461  for (i = 0; i < n; i++)
1462  g->scale_factors[j++] = 0;
1463  }
1464  if (slen2) {
1465  for (i = 0; i < 18; i++)
1466  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1467  for (i = 0; i < 3; i++)
1468  g->scale_factors[j++] = 0;
1469  } else {
1470  for (i = 0; i < 21; i++)
1471  g->scale_factors[j++] = 0;
1472  }
1473  } else {
1474  sc = s->granules[ch][0].scale_factors;
1475  j = 0;
1476  for (k = 0; k < 4; k++) {
1477  n = k == 0 ? 6 : 5;
1478  if ((g->scfsi & (0x8 >> k)) == 0) {
1479  slen = (k < 2) ? slen1 : slen2;
1480  if (slen) {
1481  for (i = 0; i < n; i++)
1482  g->scale_factors[j++] = get_bits(&s->gb, slen);
1483  } else {
1484  for (i = 0; i < n; i++)
1485  g->scale_factors[j++] = 0;
1486  }
1487  } else {
1488  /* simply copy from last granule */
1489  for (i = 0; i < n; i++) {
1490  g->scale_factors[j] = sc[j];
1491  j++;
1492  }
1493  }
1494  }
1495  g->scale_factors[j++] = 0;
1496  }
1497  } else {
1498  int tindex, tindex2, slen[4], sl, sf;
1499 
1500  /* LSF scale factors */
1501  if (g->block_type == 2)
1502  tindex = g->switch_point ? 2 : 1;
1503  else
1504  tindex = 0;
1505 
1506  sf = g->scalefac_compress;
1507  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1508  /* intensity stereo case */
1509  sf >>= 1;
1510  if (sf < 180) {
1511  lsf_sf_expand(slen, sf, 6, 6, 0);
1512  tindex2 = 3;
1513  } else if (sf < 244) {
1514  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1515  tindex2 = 4;
1516  } else {
1517  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1518  tindex2 = 5;
1519  }
1520  } else {
1521  /* normal case */
1522  if (sf < 400) {
1523  lsf_sf_expand(slen, sf, 5, 4, 4);
1524  tindex2 = 0;
1525  } else if (sf < 500) {
1526  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1527  tindex2 = 1;
1528  } else {
1529  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1530  tindex2 = 2;
1531  g->preflag = 1;
1532  }
1533  }
1534 
1535  j = 0;
1536  for (k = 0; k < 4; k++) {
1537  n = lsf_nsf_table[tindex2][tindex][k];
1538  sl = slen[k];
1539  if (sl) {
1540  for (i = 0; i < n; i++)
1541  g->scale_factors[j++] = get_bits(&s->gb, sl);
1542  } else {
1543  for (i = 0; i < n; i++)
1544  g->scale_factors[j++] = 0;
1545  }
1546  }
1547  /* XXX: should compute exact size */
1548  for (; j < 40; j++)
1549  g->scale_factors[j] = 0;
1550  }
1551 
1552  exponents_from_scale_factors(s, g, exponents);
1553 
1554  /* read Huffman coded residue */
1555  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1556  } /* ch */
1557 
1558  if (s->mode == MPA_JSTEREO)
1559  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1560 
1561  for (ch = 0; ch < s->nb_channels; ch++) {
1562  g = &s->granules[ch][gr];
1563 
1564  reorder_block(s, g);
1565  compute_antialias(s, g);
1566  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1567  }
1568  } /* gr */
1569  if (get_bits_count(&s->gb) < 0)
1570  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1571  return nb_granules * 18;
1572 }
1573 
1574 static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
1575  const uint8_t *buf, int buf_size)
1576 {
1577  int i, nb_frames, ch, ret;
1578  OUT_INT *samples_ptr;
1579 
1580  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1581 
1582  /* skip error protection field */
1583  if (s->error_protection)
1584  skip_bits(&s->gb, 16);
1585 
1586  switch(s->layer) {
1587  case 1:
1588  s->avctx->frame_size = 384;
1589  nb_frames = mp_decode_layer1(s);
1590  break;
1591  case 2:
1592  s->avctx->frame_size = 1152;
1593  nb_frames = mp_decode_layer2(s);
1594  break;
1595  case 3:
1596  s->avctx->frame_size = s->lsf ? 576 : 1152;
1597  default:
1598  nb_frames = mp_decode_layer3(s);
1599 
1600  s->last_buf_size=0;
1601  if (s->in_gb.buffer) {
1602  align_get_bits(&s->gb);
1603  i = get_bits_left(&s->gb)>>3;
1604  if (i >= 0 && i <= BACKSTEP_SIZE) {
1605  memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1606  s->last_buf_size=i;
1607  } else
1608  av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1609  s->gb = s->in_gb;
1610  s->in_gb.buffer = NULL;
1611  }
1612 
1613  align_get_bits(&s->gb);
1614  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1615  i = get_bits_left(&s->gb) >> 3;
1616 
1617  if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1618  if (i < 0)
1619  av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1620  i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1621  }
1622  av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
1623  memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1624  s->last_buf_size += i;
1625  }
1626 
1627  if(nb_frames < 0)
1628  return nb_frames;
1629 
1630  /* get output buffer */
1631  if (!samples) {
1632  s->frame.nb_samples = s->avctx->frame_size;
1633  if ((ret = ff_get_buffer(s->avctx, &s->frame)) < 0) {
1634  av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1635  return ret;
1636  }
1637  samples = (OUT_INT **)s->frame.extended_data;
1638  }
1639 
1640  /* apply the synthesis filter */
1641  for (ch = 0; ch < s->nb_channels; ch++) {
1642  int sample_stride;
1643  if (s->avctx->sample_fmt == OUT_FMT_P) {
1644  samples_ptr = samples[ch];
1645  sample_stride = 1;
1646  } else {
1647  samples_ptr = samples[0] + ch;
1648  sample_stride = s->nb_channels;
1649  }
1650  for (i = 0; i < nb_frames; i++) {
1651  RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
1652  &(s->synth_buf_offset[ch]),
1653  RENAME(ff_mpa_synth_window),
1654  &s->dither_state, samples_ptr,
1655  sample_stride, s->sb_samples[ch][i]);
1656  samples_ptr += 32 * sample_stride;
1657  }
1658  }
1659 
1660  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1661 }
1662 
1663 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1664  AVPacket *avpkt)
1665 {
1666  const uint8_t *buf = avpkt->data;
1667  int buf_size = avpkt->size;
1668  MPADecodeContext *s = avctx->priv_data;
1669  uint32_t header;
1670  int ret;
1671 
1672  while(buf_size && !*buf){
1673  buf++;
1674  buf_size--;
1675  }
1676 
1677  if (buf_size < HEADER_SIZE)
1678  return AVERROR_INVALIDDATA;
1679 
1680  header = AV_RB32(buf);
1681  if (header>>8 == AV_RB32("TAG")>>8) {
1682  av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
1683  return buf_size;
1684  }
1685  if (ff_mpa_check_header(header) < 0) {
1686  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1687  return AVERROR_INVALIDDATA;
1688  }
1689 
1690  if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1691  /* free format: prepare to compute frame size */
1692  s->frame_size = -1;
1693  return AVERROR_INVALIDDATA;
1694  }
1695  /* update codec info */
1696  avctx->channels = s->nb_channels;
1697  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1698  if (!avctx->bit_rate)
1699  avctx->bit_rate = s->bit_rate;
1700 
1701  if (s->frame_size <= 0 || s->frame_size > buf_size) {
1702  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1703  return AVERROR_INVALIDDATA;
1704  } else if (s->frame_size < buf_size) {
1705  av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
1706  buf_size= s->frame_size;
1707  }
1708 
1709  ret = mp_decode_frame(s, NULL, buf, buf_size);
1710  if (ret >= 0) {
1711  *got_frame_ptr = 1;
1712  *(AVFrame *)data = s->frame;
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  ret = mp_decode_frame(s, NULL, buf, buf_size);
1782  if (ret < 0) {
1783  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1784  return ret;
1785  }
1786 
1787  *got_frame_ptr = 1;
1788  *(AVFrame *)data = s->frame;
1789 
1790  return buf_size;
1791 }
1792 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1793 
1794 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1795 
1796 /**
1797  * Context for MP3On4 decoder
1798  */
1799 typedef struct MP3On4DecodeContext {
1800  AVFrame *frame;
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 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  s->frame = avctx->coded_frame;
1890  // Restore mp3on4 context pointer
1891  avctx->priv_data = s;
1892  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1893 
1894  /* Create a separate codec/context for each frame (first is already ok).
1895  * Each frame is 1 or 2 channels - up to 5 frames allowed
1896  */
1897  for (i = 1; i < s->frames; i++) {
1898  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1899  if (!s->mp3decctx[i])
1900  goto alloc_fail;
1901  s->mp3decctx[i]->adu_mode = 1;
1902  s->mp3decctx[i]->avctx = avctx;
1903  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1904  }
1905 
1906  return 0;
1907 alloc_fail:
1908  decode_close_mp3on4(avctx);
1909  return AVERROR(ENOMEM);
1910 }
1911 
1912 
1913 static void flush_mp3on4(AVCodecContext *avctx)
1914 {
1915  int i;
1916  MP3On4DecodeContext *s = avctx->priv_data;
1917 
1918  for (i = 0; i < s->frames; i++)
1919  mp_flush(s->mp3decctx[i]);
1920 }
1921 
1922 
1923 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1924  int *got_frame_ptr, AVPacket *avpkt)
1925 {
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  s->frame->nb_samples = MPA_FRAME_SIZE;
1938  if ((ret = ff_get_buffer(avctx, s->frame)) < 0) {
1939  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1940  return ret;
1941  }
1942  out_samples = (OUT_INT **)s->frame->extended_data;
1943 
1944  // Discard too short frames
1945  if (buf_size < HEADER_SIZE)
1946  return AVERROR_INVALIDDATA;
1947 
1948  avctx->bit_rate = 0;
1949 
1950  ch = 0;
1951  for (fr = 0; fr < s->frames; fr++) {
1952  fsize = AV_RB16(buf) >> 4;
1953  fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1954  m = s->mp3decctx[fr];
1955  av_assert1(m);
1956 
1957  if (fsize < HEADER_SIZE) {
1958  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1959  return AVERROR_INVALIDDATA;
1960  }
1961  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1962 
1963  if (ff_mpa_check_header(header) < 0) // Bad header, discard block
1964  break;
1965 
1966  avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
1967 
1968  if (ch + m->nb_channels > avctx->channels || s->coff[fr] + m->nb_channels > avctx->channels) {
1969  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1970  "channel count\n");
1971  return AVERROR_INVALIDDATA;
1972  }
1973  ch += m->nb_channels;
1974 
1975  outptr[0] = out_samples[s->coff[fr]];
1976  if (m->nb_channels > 1)
1977  outptr[1] = out_samples[s->coff[fr] + 1];
1978 
1979  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
1980  return ret;
1981 
1982  out_size += ret;
1983  buf += fsize;
1984  len -= fsize;
1985 
1986  avctx->bit_rate += m->bit_rate;
1987  }
1988 
1989  /* update codec info */
1990  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1991 
1992  s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1993  *got_frame_ptr = 1;
1994  *(AVFrame *)data = *s->frame;
1995 
1996  return buf_size;
1997 }
1998 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
1999 
2000 #if !CONFIG_FLOAT
2001 #if CONFIG_MP1_DECODER
2002 AVCodec ff_mp1_decoder = {
2003  .name = "mp1",
2004  .type = AVMEDIA_TYPE_AUDIO,
2005  .id = AV_CODEC_ID_MP1,
2006  .priv_data_size = sizeof(MPADecodeContext),
2007  .init = decode_init,
2008  .decode = decode_frame,
2009  .capabilities = CODEC_CAP_DR1,
2010  .flush = flush,
2011  .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
2012  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2013  AV_SAMPLE_FMT_S16,
2014  AV_SAMPLE_FMT_NONE },
2015 };
2016 #endif
2017 #if CONFIG_MP2_DECODER
2018 AVCodec ff_mp2_decoder = {
2019  .name = "mp2",
2020  .type = AVMEDIA_TYPE_AUDIO,
2021  .id = AV_CODEC_ID_MP2,
2022  .priv_data_size = sizeof(MPADecodeContext),
2023  .init = decode_init,
2024  .decode = decode_frame,
2025  .capabilities = CODEC_CAP_DR1,
2026  .flush = flush,
2027  .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2028  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2029  AV_SAMPLE_FMT_S16,
2030  AV_SAMPLE_FMT_NONE },
2031 };
2032 #endif
2033 #if CONFIG_MP3_DECODER
2034 AVCodec ff_mp3_decoder = {
2035  .name = "mp3",
2036  .type = AVMEDIA_TYPE_AUDIO,
2037  .id = AV_CODEC_ID_MP3,
2038  .priv_data_size = sizeof(MPADecodeContext),
2039  .init = decode_init,
2040  .decode = decode_frame,
2041  .capabilities = CODEC_CAP_DR1,
2042  .flush = flush,
2043  .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2044  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2045  AV_SAMPLE_FMT_S16,
2046  AV_SAMPLE_FMT_NONE },
2047 };
2048 #endif
2049 #if CONFIG_MP3ADU_DECODER
2050 AVCodec ff_mp3adu_decoder = {
2051  .name = "mp3adu",
2052  .type = AVMEDIA_TYPE_AUDIO,
2053  .id = AV_CODEC_ID_MP3ADU,
2054  .priv_data_size = sizeof(MPADecodeContext),
2055  .init = decode_init,
2056  .decode = decode_frame_adu,
2057  .capabilities = CODEC_CAP_DR1,
2058  .flush = flush,
2059  .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2060  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2061  AV_SAMPLE_FMT_S16,
2062  AV_SAMPLE_FMT_NONE },
2063 };
2064 #endif
2065 #if CONFIG_MP3ON4_DECODER
2066 AVCodec ff_mp3on4_decoder = {
2067  .name = "mp3on4",
2068  .type = AVMEDIA_TYPE_AUDIO,
2069  .id = AV_CODEC_ID_MP3ON4,
2070  .priv_data_size = sizeof(MP3On4DecodeContext),
2071  .init = decode_init_mp3on4,
2072  .close = decode_close_mp3on4,
2073  .decode = decode_frame_mp3on4,
2074  .capabilities = CODEC_CAP_DR1,
2075  .flush = flush_mp3on4,
2076  .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
2077  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2078  AV_SAMPLE_FMT_NONE },
2079 };
2080 #endif
2081 #endif
Generated on Sat May 25 2013 03:58:38 for FFmpeg by   doxygen 1.8.2