FFmpeg  4.4
mpegaudiodec_template.c
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1 /*
2  * MPEG Audio decoder
3  * Copyright (c) 2001, 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * MPEG Audio decoder
25  */
26 
27 #include "libavutil/attributes.h"
28 #include "libavutil/avassert.h"
30 #include "libavutil/crc.h"
31 #include "libavutil/float_dsp.h"
32 #include "libavutil/libm.h"
33 #include "libavutil/mem_internal.h"
34 #include "libavutil/thread.h"
35 
36 #include "avcodec.h"
37 #include "get_bits.h"
38 #include "internal.h"
39 #include "mathops.h"
40 #include "mpegaudiodsp.h"
41 
42 /*
43  * TODO:
44  * - test lsf / mpeg25 extensively.
45  */
46 
47 #include "mpegaudio.h"
48 #include "mpegaudiodecheader.h"
49 
50 #define BACKSTEP_SIZE 512
51 #define EXTRABYTES 24
52 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
53 
54 /* layer 3 "granule" */
55 typedef struct GranuleDef {
63  int table_select[3];
64  int subblock_gain[3];
67  int region_size[3]; /* number of huffman codes in each region */
68  int preflag;
69  int short_start, long_end; /* long/short band indexes */
71  DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
72 } GranuleDef;
73 
74 typedef struct MPADecodeContext {
78  int extrasize;
79  /* next header (used in free format parsing) */
86  INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
87  GranuleDef granules[2][2]; /* Used in Layer 3 */
88  int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
93  void (*butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len);
95  uint32_t crc;
97 
98 #define HEADER_SIZE 4
99 
100 #include "mpegaudiodata.h"
101 
102 #include "mpegaudio_tablegen.h"
103 /* intensity stereo coef table */
104 static INTFLOAT is_table_lsf[2][2][16];
105 
106 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
108 /* mult table for layer 2 group quantization */
109 
110 #define SCALE_GEN(v) \
111 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
112 
113 static const int32_t scale_factor_mult2[3][3] = {
114  SCALE_GEN(4.0 / 3.0), /* 3 steps */
115  SCALE_GEN(4.0 / 5.0), /* 5 steps */
116  SCALE_GEN(4.0 / 9.0), /* 9 steps */
117 };
118 
119 /**
120  * Convert region offsets to region sizes and truncate
121  * size to big_values.
122  */
124 {
125  int i, k, j = 0;
126  g->region_size[2] = 576 / 2;
127  for (i = 0; i < 3; i++) {
128  k = FFMIN(g->region_size[i], g->big_values);
129  g->region_size[i] = k - j;
130  j = k;
131  }
132 }
133 
135 {
136  if (g->block_type == 2) {
137  if (s->sample_rate_index != 8)
138  g->region_size[0] = (36 / 2);
139  else
140  g->region_size[0] = (72 / 2);
141  } else {
142  if (s->sample_rate_index <= 2)
143  g->region_size[0] = (36 / 2);
144  else if (s->sample_rate_index != 8)
145  g->region_size[0] = (54 / 2);
146  else
147  g->region_size[0] = (108 / 2);
148  }
149  g->region_size[1] = (576 / 2);
150 }
151 
153  int ra1, int ra2)
154 {
155  int l;
156  g->region_size[0] = ff_band_index_long[s->sample_rate_index][ra1 + 1];
157  /* should not overflow */
158  l = FFMIN(ra1 + ra2 + 2, 22);
159  g->region_size[1] = ff_band_index_long[s->sample_rate_index][ l];
160 }
161 
163 {
164  if (g->block_type == 2) {
165  if (g->switch_point) {
166  if(s->sample_rate_index == 8)
167  avpriv_request_sample(s->avctx, "switch point in 8khz");
168  /* if switched mode, we handle the 36 first samples as
169  long blocks. For 8000Hz, we handle the 72 first
170  exponents as long blocks */
171  if (s->sample_rate_index <= 2)
172  g->long_end = 8;
173  else
174  g->long_end = 6;
175 
176  g->short_start = 3;
177  } else {
178  g->long_end = 0;
179  g->short_start = 0;
180  }
181  } else {
182  g->short_start = 13;
183  g->long_end = 22;
184  }
185 }
186 
187 /* layer 1 unscaling */
188 /* n = number of bits of the mantissa minus 1 */
189 static inline int l1_unscale(int n, int mant, int scale_factor)
190 {
191  int shift, mod;
192  int64_t val;
193 
194  shift = ff_scale_factor_modshift[scale_factor];
195  mod = shift & 3;
196  shift >>= 2;
197  val = MUL64((int)(mant + (-1U << n) + 1), scale_factor_mult[n-1][mod]);
198  shift += n;
199  /* NOTE: at this point, 1 <= shift >= 21 + 15 */
200  return (int)((val + (1LL << (shift - 1))) >> shift);
201 }
202 
203 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
204 {
205  int shift, mod, val;
206 
207  shift = ff_scale_factor_modshift[scale_factor];
208  mod = shift & 3;
209  shift >>= 2;
210 
211  val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
212  /* NOTE: at this point, 0 <= shift <= 21 */
213  if (shift > 0)
214  val = (val + (1 << (shift - 1))) >> shift;
215  return val;
216 }
217 
218 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
219 static inline int l3_unscale(int value, int exponent)
220 {
221  unsigned int m;
222  int e;
223 
224  e = ff_table_4_3_exp [4 * value + (exponent & 3)];
225  m = ff_table_4_3_value[4 * value + (exponent & 3)];
226  e -= exponent >> 2;
227 #ifdef DEBUG
228  if(e < 1)
229  av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
230 #endif
231  if (e > (SUINT)31)
232  return 0;
233  m = (m + ((1U << e) >> 1)) >> e;
234 
235  return m;
236 }
237 
238 static av_cold void decode_init_static(void)
239 {
240  int i, j;
241 
242  /* scale factor multiply for layer 1 */
243  for (i = 0; i < 15; i++) {
244  int n, norm;
245  n = i + 2;
246  norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
247  scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
248  scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
249  scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
250  ff_dlog(NULL, "%d: norm=%x s=%"PRIx32" %"PRIx32" %"PRIx32"\n", i,
251  (unsigned)norm,
252  scale_factor_mult[i][0],
253  scale_factor_mult[i][1],
254  scale_factor_mult[i][2]);
255  }
256 
257  /* compute n ^ (4/3) and store it in mantissa/exp format */
258 
260 
261  for (i = 0; i < 16; i++) {
262  double f;
263  int e, k;
264 
265  for (j = 0; j < 2; j++) {
266  e = -(j + 1) * ((i + 1) >> 1);
267  f = exp2(e / 4.0);
268  k = i & 1;
269  is_table_lsf[j][k ^ 1][i] = FIXR(f);
270  is_table_lsf[j][k ][i] = FIXR(1.0);
271  ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
272  i, j, (float) is_table_lsf[j][0][i],
273  (float) is_table_lsf[j][1][i]);
274  }
275  }
278 }
279 
280 static av_cold int decode_init(AVCodecContext * avctx)
281 {
282  static AVOnce init_static_once = AV_ONCE_INIT;
283  MPADecodeContext *s = avctx->priv_data;
284 
285  s->avctx = avctx;
286 
287 #if USE_FLOATS
288  {
289  AVFloatDSPContext *fdsp;
291  if (!fdsp)
292  return AVERROR(ENOMEM);
293  s->butterflies_float = fdsp->butterflies_float;
294  av_free(fdsp);
295  }
296 #endif
297 
298  ff_mpadsp_init(&s->mpadsp);
299 
300  if (avctx->request_sample_fmt == OUT_FMT &&
301  avctx->codec_id != AV_CODEC_ID_MP3ON4)
302  avctx->sample_fmt = OUT_FMT;
303  else
304  avctx->sample_fmt = OUT_FMT_P;
305  s->err_recognition = avctx->err_recognition;
306 
307  if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
308  s->adu_mode = 1;
309 
310  ff_thread_once(&init_static_once, decode_init_static);
311 
312  return 0;
313 }
314 
315 #define C3 FIXHR(0.86602540378443864676/2)
316 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
317 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
318 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
319 
320 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
321  cases. */
323 {
324  SUINTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
325 
326  in0 = in[0*3];
327  in1 = in[1*3] + in[0*3];
328  in2 = in[2*3] + in[1*3];
329  in3 = in[3*3] + in[2*3];
330  in4 = in[4*3] + in[3*3];
331  in5 = in[5*3] + in[4*3];
332  in5 += in3;
333  in3 += in1;
334 
335  in2 = MULH3(in2, C3, 2);
336  in3 = MULH3(in3, C3, 4);
337 
338  t1 = in0 - in4;
339  t2 = MULH3(in1 - in5, C4, 2);
340 
341  out[ 7] =
342  out[10] = t1 + t2;
343  out[ 1] =
344  out[ 4] = t1 - t2;
345 
346  in0 += SHR(in4, 1);
347  in4 = in0 + in2;
348  in5 += 2*in1;
349  in1 = MULH3(in5 + in3, C5, 1);
350  out[ 8] =
351  out[ 9] = in4 + in1;
352  out[ 2] =
353  out[ 3] = in4 - in1;
354 
355  in0 -= in2;
356  in5 = MULH3(in5 - in3, C6, 2);
357  out[ 0] =
358  out[ 5] = in0 - in5;
359  out[ 6] =
360  out[11] = in0 + in5;
361 }
362 
363 static int handle_crc(MPADecodeContext *s, int sec_len)
364 {
365  if (s->error_protection && (s->err_recognition & AV_EF_CRCCHECK)) {
366  const uint8_t *buf = s->gb.buffer - HEADER_SIZE;
367  int sec_byte_len = sec_len >> 3;
368  int sec_rem_bits = sec_len & 7;
369  const AVCRC *crc_tab = av_crc_get_table(AV_CRC_16_ANSI);
370  uint8_t tmp_buf[4];
371  uint32_t crc_val = av_crc(crc_tab, UINT16_MAX, &buf[2], 2);
372  crc_val = av_crc(crc_tab, crc_val, &buf[6], sec_byte_len);
373 
374  AV_WB32(tmp_buf,
375  ((buf[6 + sec_byte_len] & (0xFF00 >> sec_rem_bits)) << 24) +
376  ((s->crc << 16) >> sec_rem_bits));
377 
378  crc_val = av_crc(crc_tab, crc_val, tmp_buf, 3);
379 
380  if (crc_val) {
381  av_log(s->avctx, AV_LOG_ERROR, "CRC mismatch %X!\n", crc_val);
382  if (s->err_recognition & AV_EF_EXPLODE)
383  return AVERROR_INVALIDDATA;
384  }
385  }
386  return 0;
387 }
388 
389 /* return the number of decoded frames */
391 {
392  int bound, i, v, n, ch, j, mant;
393  uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
394  uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
395  int ret;
396 
397  ret = handle_crc(s, (s->nb_channels == 1) ? 8*16 : 8*32);
398  if (ret < 0)
399  return ret;
400 
401  if (s->mode == MPA_JSTEREO)
402  bound = (s->mode_ext + 1) * 4;
403  else
404  bound = SBLIMIT;
405 
406  /* allocation bits */
407  for (i = 0; i < bound; i++) {
408  for (ch = 0; ch < s->nb_channels; ch++) {
409  allocation[ch][i] = get_bits(&s->gb, 4);
410  }
411  }
412  for (i = bound; i < SBLIMIT; i++)
413  allocation[0][i] = get_bits(&s->gb, 4);
414 
415  /* scale factors */
416  for (i = 0; i < bound; i++) {
417  for (ch = 0; ch < s->nb_channels; ch++) {
418  if (allocation[ch][i])
419  scale_factors[ch][i] = get_bits(&s->gb, 6);
420  }
421  }
422  for (i = bound; i < SBLIMIT; i++) {
423  if (allocation[0][i]) {
424  scale_factors[0][i] = get_bits(&s->gb, 6);
425  scale_factors[1][i] = get_bits(&s->gb, 6);
426  }
427  }
428 
429  /* compute samples */
430  for (j = 0; j < 12; j++) {
431  for (i = 0; i < bound; i++) {
432  for (ch = 0; ch < s->nb_channels; ch++) {
433  n = allocation[ch][i];
434  if (n) {
435  mant = get_bits(&s->gb, n + 1);
436  v = l1_unscale(n, mant, scale_factors[ch][i]);
437  } else {
438  v = 0;
439  }
440  s->sb_samples[ch][j][i] = v;
441  }
442  }
443  for (i = bound; i < SBLIMIT; i++) {
444  n = allocation[0][i];
445  if (n) {
446  mant = get_bits(&s->gb, n + 1);
447  v = l1_unscale(n, mant, scale_factors[0][i]);
448  s->sb_samples[0][j][i] = v;
449  v = l1_unscale(n, mant, scale_factors[1][i]);
450  s->sb_samples[1][j][i] = v;
451  } else {
452  s->sb_samples[0][j][i] = 0;
453  s->sb_samples[1][j][i] = 0;
454  }
455  }
456  }
457  return 12;
458 }
459 
461 {
462  int sblimit; /* number of used subbands */
463  const unsigned char *alloc_table;
464  int table, bit_alloc_bits, i, j, ch, bound, v;
465  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
466  unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
467  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
468  int scale, qindex, bits, steps, k, l, m, b;
469  int ret;
470 
471  /* select decoding table */
472  table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
473  s->sample_rate, s->lsf);
474  sblimit = ff_mpa_sblimit_table[table];
476 
477  if (s->mode == MPA_JSTEREO)
478  bound = (s->mode_ext + 1) * 4;
479  else
480  bound = sblimit;
481 
482  ff_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
483 
484  /* sanity check */
485  if (bound > sblimit)
486  bound = sblimit;
487 
488  /* parse bit allocation */
489  j = 0;
490  for (i = 0; i < bound; i++) {
491  bit_alloc_bits = alloc_table[j];
492  for (ch = 0; ch < s->nb_channels; ch++)
493  bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
494  j += 1 << bit_alloc_bits;
495  }
496  for (i = bound; i < sblimit; i++) {
497  bit_alloc_bits = alloc_table[j];
498  v = get_bits(&s->gb, bit_alloc_bits);
499  bit_alloc[0][i] = v;
500  bit_alloc[1][i] = v;
501  j += 1 << bit_alloc_bits;
502  }
503 
504  /* scale codes */
505  for (i = 0; i < sblimit; i++) {
506  for (ch = 0; ch < s->nb_channels; ch++) {
507  if (bit_alloc[ch][i])
508  scale_code[ch][i] = get_bits(&s->gb, 2);
509  }
510  }
511 
512  ret = handle_crc(s, get_bits_count(&s->gb) - 16);
513  if (ret < 0)
514  return ret;
515 
516  /* scale factors */
517  for (i = 0; i < sblimit; i++) {
518  for (ch = 0; ch < s->nb_channels; ch++) {
519  if (bit_alloc[ch][i]) {
520  sf = scale_factors[ch][i];
521  switch (scale_code[ch][i]) {
522  default:
523  case 0:
524  sf[0] = get_bits(&s->gb, 6);
525  sf[1] = get_bits(&s->gb, 6);
526  sf[2] = get_bits(&s->gb, 6);
527  break;
528  case 2:
529  sf[0] = get_bits(&s->gb, 6);
530  sf[1] = sf[0];
531  sf[2] = sf[0];
532  break;
533  case 1:
534  sf[0] = get_bits(&s->gb, 6);
535  sf[2] = get_bits(&s->gb, 6);
536  sf[1] = sf[0];
537  break;
538  case 3:
539  sf[0] = get_bits(&s->gb, 6);
540  sf[2] = get_bits(&s->gb, 6);
541  sf[1] = sf[2];
542  break;
543  }
544  }
545  }
546  }
547 
548  /* samples */
549  for (k = 0; k < 3; k++) {
550  for (l = 0; l < 12; l += 3) {
551  j = 0;
552  for (i = 0; i < bound; i++) {
553  bit_alloc_bits = alloc_table[j];
554  for (ch = 0; ch < s->nb_channels; ch++) {
555  b = bit_alloc[ch][i];
556  if (b) {
557  scale = scale_factors[ch][i][k];
558  qindex = alloc_table[j+b];
559  bits = ff_mpa_quant_bits[qindex];
560  if (bits < 0) {
561  int v2;
562  /* 3 values at the same time */
563  v = get_bits(&s->gb, -bits);
564  v2 = ff_division_tabs[qindex][v];
565  steps = ff_mpa_quant_steps[qindex];
566 
567  s->sb_samples[ch][k * 12 + l + 0][i] =
568  l2_unscale_group(steps, v2 & 15, scale);
569  s->sb_samples[ch][k * 12 + l + 1][i] =
570  l2_unscale_group(steps, (v2 >> 4) & 15, scale);
571  s->sb_samples[ch][k * 12 + l + 2][i] =
572  l2_unscale_group(steps, v2 >> 8 , scale);
573  } else {
574  for (m = 0; m < 3; m++) {
575  v = get_bits(&s->gb, bits);
576  v = l1_unscale(bits - 1, v, scale);
577  s->sb_samples[ch][k * 12 + l + m][i] = v;
578  }
579  }
580  } else {
581  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
582  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
583  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
584  }
585  }
586  /* next subband in alloc table */
587  j += 1 << bit_alloc_bits;
588  }
589  /* XXX: find a way to avoid this duplication of code */
590  for (i = bound; i < sblimit; i++) {
591  bit_alloc_bits = alloc_table[j];
592  b = bit_alloc[0][i];
593  if (b) {
594  int mant, scale0, scale1;
595  scale0 = scale_factors[0][i][k];
596  scale1 = scale_factors[1][i][k];
597  qindex = alloc_table[j + b];
598  bits = ff_mpa_quant_bits[qindex];
599  if (bits < 0) {
600  /* 3 values at the same time */
601  v = get_bits(&s->gb, -bits);
602  steps = ff_mpa_quant_steps[qindex];
603  mant = v % steps;
604  v = v / steps;
605  s->sb_samples[0][k * 12 + l + 0][i] =
606  l2_unscale_group(steps, mant, scale0);
607  s->sb_samples[1][k * 12 + l + 0][i] =
608  l2_unscale_group(steps, mant, scale1);
609  mant = v % steps;
610  v = v / steps;
611  s->sb_samples[0][k * 12 + l + 1][i] =
612  l2_unscale_group(steps, mant, scale0);
613  s->sb_samples[1][k * 12 + l + 1][i] =
614  l2_unscale_group(steps, mant, scale1);
615  s->sb_samples[0][k * 12 + l + 2][i] =
616  l2_unscale_group(steps, v, scale0);
617  s->sb_samples[1][k * 12 + l + 2][i] =
618  l2_unscale_group(steps, v, scale1);
619  } else {
620  for (m = 0; m < 3; m++) {
621  mant = get_bits(&s->gb, bits);
622  s->sb_samples[0][k * 12 + l + m][i] =
623  l1_unscale(bits - 1, mant, scale0);
624  s->sb_samples[1][k * 12 + l + m][i] =
625  l1_unscale(bits - 1, mant, scale1);
626  }
627  }
628  } else {
629  s->sb_samples[0][k * 12 + l + 0][i] = 0;
630  s->sb_samples[0][k * 12 + l + 1][i] = 0;
631  s->sb_samples[0][k * 12 + l + 2][i] = 0;
632  s->sb_samples[1][k * 12 + l + 0][i] = 0;
633  s->sb_samples[1][k * 12 + l + 1][i] = 0;
634  s->sb_samples[1][k * 12 + l + 2][i] = 0;
635  }
636  /* next subband in alloc table */
637  j += 1 << bit_alloc_bits;
638  }
639  /* fill remaining samples to zero */
640  for (i = sblimit; i < SBLIMIT; i++) {
641  for (ch = 0; ch < s->nb_channels; ch++) {
642  s->sb_samples[ch][k * 12 + l + 0][i] = 0;
643  s->sb_samples[ch][k * 12 + l + 1][i] = 0;
644  s->sb_samples[ch][k * 12 + l + 2][i] = 0;
645  }
646  }
647  }
648  }
649  return 3 * 12;
650 }
651 
652 #define SPLIT(dst,sf,n) \
653  if (n == 3) { \
654  int m = (sf * 171) >> 9; \
655  dst = sf - 3 * m; \
656  sf = m; \
657  } else if (n == 4) { \
658  dst = sf & 3; \
659  sf >>= 2; \
660  } else if (n == 5) { \
661  int m = (sf * 205) >> 10; \
662  dst = sf - 5 * m; \
663  sf = m; \
664  } else if (n == 6) { \
665  int m = (sf * 171) >> 10; \
666  dst = sf - 6 * m; \
667  sf = m; \
668  } else { \
669  dst = 0; \
670  }
671 
672 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
673  int n3)
674 {
675  SPLIT(slen[3], sf, n3)
676  SPLIT(slen[2], sf, n2)
677  SPLIT(slen[1], sf, n1)
678  slen[0] = sf;
679 }
680 
682  int16_t *exponents)
683 {
684  const uint8_t *bstab, *pretab;
685  int len, i, j, k, l, v0, shift, gain, gains[3];
686  int16_t *exp_ptr;
687 
688  exp_ptr = exponents;
689  gain = g->global_gain - 210;
690  shift = g->scalefac_scale + 1;
691 
692  bstab = ff_band_size_long[s->sample_rate_index];
693  pretab = ff_mpa_pretab[g->preflag];
694  for (i = 0; i < g->long_end; i++) {
695  v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
696  len = bstab[i];
697  for (j = len; j > 0; j--)
698  *exp_ptr++ = v0;
699  }
700 
701  if (g->short_start < 13) {
702  bstab = ff_band_size_short[s->sample_rate_index];
703  gains[0] = gain - (g->subblock_gain[0] << 3);
704  gains[1] = gain - (g->subblock_gain[1] << 3);
705  gains[2] = gain - (g->subblock_gain[2] << 3);
706  k = g->long_end;
707  for (i = g->short_start; i < 13; i++) {
708  len = bstab[i];
709  for (l = 0; l < 3; l++) {
710  v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
711  for (j = len; j > 0; j--)
712  *exp_ptr++ = v0;
713  }
714  }
715  }
716 }
717 
718 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
719  int *end_pos2)
720 {
721  if (s->in_gb.buffer && *pos >= s->gb.size_in_bits - s->extrasize * 8) {
722  s->gb = s->in_gb;
723  s->in_gb.buffer = NULL;
724  s->extrasize = 0;
725  av_assert2((get_bits_count(&s->gb) & 7) == 0);
726  skip_bits_long(&s->gb, *pos - *end_pos);
727  *end_pos2 =
728  *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
729  *pos = get_bits_count(&s->gb);
730  }
731 }
732 
733 /* Following is an optimized code for
734  INTFLOAT v = *src
735  if(get_bits1(&s->gb))
736  v = -v;
737  *dst = v;
738 */
739 #if USE_FLOATS
740 #define READ_FLIP_SIGN(dst,src) \
741  v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
742  AV_WN32A(dst, v);
743 #else
744 #define READ_FLIP_SIGN(dst,src) \
745  v = -get_bits1(&s->gb); \
746  *(dst) = (*(src) ^ v) - v;
747 #endif
748 
750  int16_t *exponents, int end_pos2)
751 {
752  int s_index;
753  int i;
754  int last_pos, bits_left;
755  VLC *vlc;
756  int end_pos = FFMIN(end_pos2, s->gb.size_in_bits - s->extrasize * 8);
757 
758  /* low frequencies (called big values) */
759  s_index = 0;
760  for (i = 0; i < 3; i++) {
761  int j, k, l, linbits;
762  j = g->region_size[i];
763  if (j == 0)
764  continue;
765  /* select vlc table */
766  k = g->table_select[i];
767  l = ff_mpa_huff_data[k][0];
768  linbits = ff_mpa_huff_data[k][1];
769  vlc = &ff_huff_vlc[l];
770 
771  if (!l) {
772  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
773  s_index += 2 * j;
774  continue;
775  }
776 
777  /* read huffcode and compute each couple */
778  for (; j > 0; j--) {
779  int exponent, x, y;
780  int v;
781  int pos = get_bits_count(&s->gb);
782 
783  if (pos >= end_pos){
784  switch_buffer(s, &pos, &end_pos, &end_pos2);
785  if (pos >= end_pos)
786  break;
787  }
788  y = get_vlc2(&s->gb, vlc->table, 7, 3);
789 
790  if (!y) {
791  g->sb_hybrid[s_index ] =
792  g->sb_hybrid[s_index + 1] = 0;
793  s_index += 2;
794  continue;
795  }
796 
797  exponent= exponents[s_index];
798 
799  ff_dlog(s->avctx, "region=%d n=%d y=%d exp=%d\n",
800  i, g->region_size[i] - j, y, exponent);
801  if (y & 16) {
802  x = y >> 5;
803  y = y & 0x0f;
804  if (x < 15) {
805  READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
806  } else {
807  x += get_bitsz(&s->gb, linbits);
808  v = l3_unscale(x, exponent);
809  if (get_bits1(&s->gb))
810  v = -v;
811  g->sb_hybrid[s_index] = v;
812  }
813  if (y < 15) {
814  READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
815  } else {
816  y += get_bitsz(&s->gb, linbits);
817  v = l3_unscale(y, exponent);
818  if (get_bits1(&s->gb))
819  v = -v;
820  g->sb_hybrid[s_index + 1] = v;
821  }
822  } else {
823  x = y >> 5;
824  y = y & 0x0f;
825  x += y;
826  if (x < 15) {
827  READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
828  } else {
829  x += get_bitsz(&s->gb, linbits);
830  v = l3_unscale(x, exponent);
831  if (get_bits1(&s->gb))
832  v = -v;
833  g->sb_hybrid[s_index+!!y] = v;
834  }
835  g->sb_hybrid[s_index + !y] = 0;
836  }
837  s_index += 2;
838  }
839  }
840 
841  /* high frequencies */
842  vlc = &ff_huff_quad_vlc[g->count1table_select];
843  last_pos = 0;
844  while (s_index <= 572) {
845  int pos, code;
846  pos = get_bits_count(&s->gb);
847  if (pos >= end_pos) {
848  if (pos > end_pos2 && last_pos) {
849  /* some encoders generate an incorrect size for this
850  part. We must go back into the data */
851  s_index -= 4;
852  skip_bits_long(&s->gb, last_pos - pos);
853  av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
854  if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
855  s_index=0;
856  break;
857  }
858  switch_buffer(s, &pos, &end_pos, &end_pos2);
859  if (pos >= end_pos)
860  break;
861  }
862  last_pos = pos;
863 
864  code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
865  ff_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
866  g->sb_hybrid[s_index + 0] =
867  g->sb_hybrid[s_index + 1] =
868  g->sb_hybrid[s_index + 2] =
869  g->sb_hybrid[s_index + 3] = 0;
870  while (code) {
871  static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
872  int v;
873  int pos = s_index + idxtab[code];
874  code ^= 8 >> idxtab[code];
875  READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
876  }
877  s_index += 4;
878  }
879  /* skip extension bits */
880  bits_left = end_pos2 - get_bits_count(&s->gb);
881  if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
882  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
883  s_index=0;
884  } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
885  av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
886  s_index = 0;
887  }
888  memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
889  skip_bits_long(&s->gb, bits_left);
890 
891  i = get_bits_count(&s->gb);
892  switch_buffer(s, &i, &end_pos, &end_pos2);
893 
894  return 0;
895 }
896 
897 /* Reorder short blocks from bitstream order to interleaved order. It
898  would be faster to do it in parsing, but the code would be far more
899  complicated */
901 {
902  int i, j, len;
903  INTFLOAT *ptr, *dst, *ptr1;
904  INTFLOAT tmp[576];
905 
906  if (g->block_type != 2)
907  return;
908 
909  if (g->switch_point) {
910  if (s->sample_rate_index != 8)
911  ptr = g->sb_hybrid + 36;
912  else
913  ptr = g->sb_hybrid + 72;
914  } else {
915  ptr = g->sb_hybrid;
916  }
917 
918  for (i = g->short_start; i < 13; i++) {
919  len = ff_band_size_short[s->sample_rate_index][i];
920  ptr1 = ptr;
921  dst = tmp;
922  for (j = len; j > 0; j--) {
923  *dst++ = ptr[0*len];
924  *dst++ = ptr[1*len];
925  *dst++ = ptr[2*len];
926  ptr++;
927  }
928  ptr += 2 * len;
929  memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
930  }
931 }
932 
933 #define ISQRT2 FIXR(0.70710678118654752440)
934 
936 {
937  int i, j, k, l;
938  int sf_max, sf, len, non_zero_found;
939  INTFLOAT *tab0, *tab1, v1, v2;
940  const INTFLOAT (*is_tab)[16];
941  SUINTFLOAT tmp0, tmp1;
942  int non_zero_found_short[3];
943 
944  /* intensity stereo */
945  if (s->mode_ext & MODE_EXT_I_STEREO) {
946  if (!s->lsf) {
947  is_tab = is_table;
948  sf_max = 7;
949  } else {
950  is_tab = is_table_lsf[g1->scalefac_compress & 1];
951  sf_max = 16;
952  }
953 
954  tab0 = g0->sb_hybrid + 576;
955  tab1 = g1->sb_hybrid + 576;
956 
957  non_zero_found_short[0] = 0;
958  non_zero_found_short[1] = 0;
959  non_zero_found_short[2] = 0;
960  k = (13 - g1->short_start) * 3 + g1->long_end - 3;
961  for (i = 12; i >= g1->short_start; i--) {
962  /* for last band, use previous scale factor */
963  if (i != 11)
964  k -= 3;
965  len = ff_band_size_short[s->sample_rate_index][i];
966  for (l = 2; l >= 0; l--) {
967  tab0 -= len;
968  tab1 -= len;
969  if (!non_zero_found_short[l]) {
970  /* test if non zero band. if so, stop doing i-stereo */
971  for (j = 0; j < len; j++) {
972  if (tab1[j] != 0) {
973  non_zero_found_short[l] = 1;
974  goto found1;
975  }
976  }
977  sf = g1->scale_factors[k + l];
978  if (sf >= sf_max)
979  goto found1;
980 
981  v1 = is_tab[0][sf];
982  v2 = is_tab[1][sf];
983  for (j = 0; j < len; j++) {
984  tmp0 = tab0[j];
985  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
986  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
987  }
988  } else {
989 found1:
990  if (s->mode_ext & MODE_EXT_MS_STEREO) {
991  /* lower part of the spectrum : do ms stereo
992  if enabled */
993  for (j = 0; j < len; j++) {
994  tmp0 = tab0[j];
995  tmp1 = tab1[j];
996  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
997  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
998  }
999  }
1000  }
1001  }
1002  }
1003 
1004  non_zero_found = non_zero_found_short[0] |
1005  non_zero_found_short[1] |
1006  non_zero_found_short[2];
1007 
1008  for (i = g1->long_end - 1;i >= 0;i--) {
1009  len = ff_band_size_long[s->sample_rate_index][i];
1010  tab0 -= len;
1011  tab1 -= len;
1012  /* test if non zero band. if so, stop doing i-stereo */
1013  if (!non_zero_found) {
1014  for (j = 0; j < len; j++) {
1015  if (tab1[j] != 0) {
1016  non_zero_found = 1;
1017  goto found2;
1018  }
1019  }
1020  /* for last band, use previous scale factor */
1021  k = (i == 21) ? 20 : i;
1022  sf = g1->scale_factors[k];
1023  if (sf >= sf_max)
1024  goto found2;
1025  v1 = is_tab[0][sf];
1026  v2 = is_tab[1][sf];
1027  for (j = 0; j < len; j++) {
1028  tmp0 = tab0[j];
1029  tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1030  tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1031  }
1032  } else {
1033 found2:
1034  if (s->mode_ext & MODE_EXT_MS_STEREO) {
1035  /* lower part of the spectrum : do ms stereo
1036  if enabled */
1037  for (j = 0; j < len; j++) {
1038  tmp0 = tab0[j];
1039  tmp1 = tab1[j];
1040  tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1041  tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1042  }
1043  }
1044  }
1045  }
1046  } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1047  /* ms stereo ONLY */
1048  /* NOTE: the 1/sqrt(2) normalization factor is included in the
1049  global gain */
1050 #if USE_FLOATS
1051  s->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1052 #else
1053  tab0 = g0->sb_hybrid;
1054  tab1 = g1->sb_hybrid;
1055  for (i = 0; i < 576; i++) {
1056  tmp0 = tab0[i];
1057  tmp1 = tab1[i];
1058  tab0[i] = tmp0 + tmp1;
1059  tab1[i] = tmp0 - tmp1;
1060  }
1061 #endif
1062  }
1063 }
1064 
1065 #if USE_FLOATS
1066 #if HAVE_MIPSFPU
1068 #endif /* HAVE_MIPSFPU */
1069 #else
1070 #if HAVE_MIPSDSP
1072 #endif /* HAVE_MIPSDSP */
1073 #endif /* USE_FLOATS */
1074 
1075 #ifndef compute_antialias
1076 #if USE_FLOATS
1077 #define AA(j) do { \
1078  float tmp0 = ptr[-1-j]; \
1079  float tmp1 = ptr[ j]; \
1080  ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1081  ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1082  } while (0)
1083 #else
1084 #define AA(j) do { \
1085  SUINT tmp0 = ptr[-1-j]; \
1086  SUINT tmp1 = ptr[ j]; \
1087  SUINT tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1088  ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1089  ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1090  } while (0)
1091 #endif
1092 
1094 {
1095  INTFLOAT *ptr;
1096  int n, i;
1097 
1098  /* we antialias only "long" bands */
1099  if (g->block_type == 2) {
1100  if (!g->switch_point)
1101  return;
1102  /* XXX: check this for 8000Hz case */
1103  n = 1;
1104  } else {
1105  n = SBLIMIT - 1;
1106  }
1107 
1108  ptr = g->sb_hybrid + 18;
1109  for (i = n; i > 0; i--) {
1110  AA(0);
1111  AA(1);
1112  AA(2);
1113  AA(3);
1114  AA(4);
1115  AA(5);
1116  AA(6);
1117  AA(7);
1118 
1119  ptr += 18;
1120  }
1121 }
1122 #endif /* compute_antialias */
1123 
1125  INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1126 {
1127  INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1128  INTFLOAT out2[12];
1129  int i, j, mdct_long_end, sblimit;
1130 
1131  /* find last non zero block */
1132  ptr = g->sb_hybrid + 576;
1133  ptr1 = g->sb_hybrid + 2 * 18;
1134  while (ptr >= ptr1) {
1135  int32_t *p;
1136  ptr -= 6;
1137  p = (int32_t*)ptr;
1138  if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1139  break;
1140  }
1141  sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1142 
1143  if (g->block_type == 2) {
1144  /* XXX: check for 8000 Hz */
1145  if (g->switch_point)
1146  mdct_long_end = 2;
1147  else
1148  mdct_long_end = 0;
1149  } else {
1150  mdct_long_end = sblimit;
1151  }
1152 
1153  s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1154  mdct_long_end, g->switch_point,
1155  g->block_type);
1156 
1157  buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1158  ptr = g->sb_hybrid + 18 * mdct_long_end;
1159 
1160  for (j = mdct_long_end; j < sblimit; j++) {
1161  /* select frequency inversion */
1162  win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1163  out_ptr = sb_samples + j;
1164 
1165  for (i = 0; i < 6; i++) {
1166  *out_ptr = buf[4*i];
1167  out_ptr += SBLIMIT;
1168  }
1169  imdct12(out2, ptr + 0);
1170  for (i = 0; i < 6; i++) {
1171  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1172  buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1173  out_ptr += SBLIMIT;
1174  }
1175  imdct12(out2, ptr + 1);
1176  for (i = 0; i < 6; i++) {
1177  *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1178  buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1179  out_ptr += SBLIMIT;
1180  }
1181  imdct12(out2, ptr + 2);
1182  for (i = 0; i < 6; i++) {
1183  buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1184  buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1185  buf[4*(i + 6*2)] = 0;
1186  }
1187  ptr += 18;
1188  buf += (j&3) != 3 ? 1 : (4*18-3);
1189  }
1190  /* zero bands */
1191  for (j = sblimit; j < SBLIMIT; j++) {
1192  /* overlap */
1193  out_ptr = sb_samples + j;
1194  for (i = 0; i < 18; i++) {
1195  *out_ptr = buf[4*i];
1196  buf[4*i] = 0;
1197  out_ptr += SBLIMIT;
1198  }
1199  buf += (j&3) != 3 ? 1 : (4*18-3);
1200  }
1201 }
1202 
1203 /* main layer3 decoding function */
1205 {
1206  int nb_granules, main_data_begin;
1207  int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1208  GranuleDef *g;
1209  int16_t exponents[576]; //FIXME try INTFLOAT
1210  int ret;
1211 
1212  /* read side info */
1213  if (s->lsf) {
1214  ret = handle_crc(s, ((s->nb_channels == 1) ? 8*9 : 8*17));
1215  main_data_begin = get_bits(&s->gb, 8);
1216  skip_bits(&s->gb, s->nb_channels);
1217  nb_granules = 1;
1218  } else {
1219  ret = handle_crc(s, ((s->nb_channels == 1) ? 8*17 : 8*32));
1220  main_data_begin = get_bits(&s->gb, 9);
1221  if (s->nb_channels == 2)
1222  skip_bits(&s->gb, 3);
1223  else
1224  skip_bits(&s->gb, 5);
1225  nb_granules = 2;
1226  for (ch = 0; ch < s->nb_channels; ch++) {
1227  s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1228  s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1229  }
1230  }
1231  if (ret < 0)
1232  return ret;
1233 
1234  for (gr = 0; gr < nb_granules; gr++) {
1235  for (ch = 0; ch < s->nb_channels; ch++) {
1236  ff_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1237  g = &s->granules[ch][gr];
1238  g->part2_3_length = get_bits(&s->gb, 12);
1239  g->big_values = get_bits(&s->gb, 9);
1240  if (g->big_values > 288) {
1241  av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1242  return AVERROR_INVALIDDATA;
1243  }
1244 
1245  g->global_gain = get_bits(&s->gb, 8);
1246  /* if MS stereo only is selected, we precompute the
1247  1/sqrt(2) renormalization factor */
1248  if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1250  g->global_gain -= 2;
1251  if (s->lsf)
1252  g->scalefac_compress = get_bits(&s->gb, 9);
1253  else
1254  g->scalefac_compress = get_bits(&s->gb, 4);
1255  blocksplit_flag = get_bits1(&s->gb);
1256  if (blocksplit_flag) {
1257  g->block_type = get_bits(&s->gb, 2);
1258  if (g->block_type == 0) {
1259  av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1260  return AVERROR_INVALIDDATA;
1261  }
1262  g->switch_point = get_bits1(&s->gb);
1263  for (i = 0; i < 2; i++)
1264  g->table_select[i] = get_bits(&s->gb, 5);
1265  for (i = 0; i < 3; i++)
1266  g->subblock_gain[i] = get_bits(&s->gb, 3);
1267  init_short_region(s, g);
1268  } else {
1269  int region_address1, region_address2;
1270  g->block_type = 0;
1271  g->switch_point = 0;
1272  for (i = 0; i < 3; i++)
1273  g->table_select[i] = get_bits(&s->gb, 5);
1274  /* compute huffman coded region sizes */
1275  region_address1 = get_bits(&s->gb, 4);
1276  region_address2 = get_bits(&s->gb, 3);
1277  ff_dlog(s->avctx, "region1=%d region2=%d\n",
1278  region_address1, region_address2);
1279  init_long_region(s, g, region_address1, region_address2);
1280  }
1283 
1284  g->preflag = 0;
1285  if (!s->lsf)
1286  g->preflag = get_bits1(&s->gb);
1287  g->scalefac_scale = get_bits1(&s->gb);
1288  g->count1table_select = get_bits1(&s->gb);
1289  ff_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1290  g->block_type, g->switch_point);
1291  }
1292  }
1293 
1294  if (!s->adu_mode) {
1295  int skip;
1296  const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb) >> 3);
1297  s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0,
1298  FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
1299  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1300  /* now we get bits from the main_data_begin offset */
1301  ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1302  main_data_begin, s->last_buf_size);
1303 
1304  memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize);
1305  s->in_gb = s->gb;
1306  init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8);
1307  s->last_buf_size <<= 3;
1308  for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1309  for (ch = 0; ch < s->nb_channels; ch++) {
1310  g = &s->granules[ch][gr];
1311  s->last_buf_size += g->part2_3_length;
1312  memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1313  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1314  }
1315  }
1316  skip = s->last_buf_size - 8 * main_data_begin;
1317  if (skip >= s->gb.size_in_bits - s->extrasize * 8 && s->in_gb.buffer) {
1318  skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8);
1319  s->gb = s->in_gb;
1320  s->in_gb.buffer = NULL;
1321  s->extrasize = 0;
1322  } else {
1323  skip_bits_long(&s->gb, skip);
1324  }
1325  } else {
1326  gr = 0;
1327  s->extrasize = 0;
1328  }
1329 
1330  for (; gr < nb_granules; gr++) {
1331  for (ch = 0; ch < s->nb_channels; ch++) {
1332  g = &s->granules[ch][gr];
1333  bits_pos = get_bits_count(&s->gb);
1334 
1335  if (!s->lsf) {
1336  uint8_t *sc;
1337  int slen, slen1, slen2;
1338 
1339  /* MPEG-1 scale factors */
1340  slen1 = ff_slen_table[0][g->scalefac_compress];
1341  slen2 = ff_slen_table[1][g->scalefac_compress];
1342  ff_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1343  if (g->block_type == 2) {
1344  n = g->switch_point ? 17 : 18;
1345  j = 0;
1346  if (slen1) {
1347  for (i = 0; i < n; i++)
1348  g->scale_factors[j++] = get_bits(&s->gb, slen1);
1349  } else {
1350  for (i = 0; i < n; i++)
1351  g->scale_factors[j++] = 0;
1352  }
1353  if (slen2) {
1354  for (i = 0; i < 18; i++)
1355  g->scale_factors[j++] = get_bits(&s->gb, slen2);
1356  for (i = 0; i < 3; i++)
1357  g->scale_factors[j++] = 0;
1358  } else {
1359  for (i = 0; i < 21; i++)
1360  g->scale_factors[j++] = 0;
1361  }
1362  } else {
1363  sc = s->granules[ch][0].scale_factors;
1364  j = 0;
1365  for (k = 0; k < 4; k++) {
1366  n = k == 0 ? 6 : 5;
1367  if ((g->scfsi & (0x8 >> k)) == 0) {
1368  slen = (k < 2) ? slen1 : slen2;
1369  if (slen) {
1370  for (i = 0; i < n; i++)
1371  g->scale_factors[j++] = get_bits(&s->gb, slen);
1372  } else {
1373  for (i = 0; i < n; i++)
1374  g->scale_factors[j++] = 0;
1375  }
1376  } else {
1377  /* simply copy from last granule */
1378  for (i = 0; i < n; i++) {
1379  g->scale_factors[j] = sc[j];
1380  j++;
1381  }
1382  }
1383  }
1384  g->scale_factors[j++] = 0;
1385  }
1386  } else {
1387  int tindex, tindex2, slen[4], sl, sf;
1388 
1389  /* LSF scale factors */
1390  if (g->block_type == 2)
1391  tindex = g->switch_point ? 2 : 1;
1392  else
1393  tindex = 0;
1394 
1395  sf = g->scalefac_compress;
1396  if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1397  /* intensity stereo case */
1398  sf >>= 1;
1399  if (sf < 180) {
1400  lsf_sf_expand(slen, sf, 6, 6, 0);
1401  tindex2 = 3;
1402  } else if (sf < 244) {
1403  lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1404  tindex2 = 4;
1405  } else {
1406  lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1407  tindex2 = 5;
1408  }
1409  } else {
1410  /* normal case */
1411  if (sf < 400) {
1412  lsf_sf_expand(slen, sf, 5, 4, 4);
1413  tindex2 = 0;
1414  } else if (sf < 500) {
1415  lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1416  tindex2 = 1;
1417  } else {
1418  lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1419  tindex2 = 2;
1420  g->preflag = 1;
1421  }
1422  }
1423 
1424  j = 0;
1425  for (k = 0; k < 4; k++) {
1426  n = ff_lsf_nsf_table[tindex2][tindex][k];
1427  sl = slen[k];
1428  if (sl) {
1429  for (i = 0; i < n; i++)
1430  g->scale_factors[j++] = get_bits(&s->gb, sl);
1431  } else {
1432  for (i = 0; i < n; i++)
1433  g->scale_factors[j++] = 0;
1434  }
1435  }
1436  /* XXX: should compute exact size */
1437  for (; j < 40; j++)
1438  g->scale_factors[j] = 0;
1439  }
1440 
1441  exponents_from_scale_factors(s, g, exponents);
1442 
1443  /* read Huffman coded residue */
1444  huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1445  } /* ch */
1446 
1447  if (s->mode == MPA_JSTEREO)
1448  compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1449 
1450  for (ch = 0; ch < s->nb_channels; ch++) {
1451  g = &s->granules[ch][gr];
1452 
1453  reorder_block(s, g);
1454  compute_antialias(s, g);
1455  compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1456  }
1457  } /* gr */
1458  if (get_bits_count(&s->gb) < 0)
1459  skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1460  return nb_granules * 18;
1461 }
1462 
1464  const uint8_t *buf, int buf_size)
1465 {
1466  int i, nb_frames, ch, ret;
1467  OUT_INT *samples_ptr;
1468 
1469  init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1470  if (s->error_protection)
1471  s->crc = get_bits(&s->gb, 16);
1472 
1473  switch(s->layer) {
1474  case 1:
1475  s->avctx->frame_size = 384;
1476  nb_frames = mp_decode_layer1(s);
1477  break;
1478  case 2:
1479  s->avctx->frame_size = 1152;
1480  nb_frames = mp_decode_layer2(s);
1481  break;
1482  case 3:
1483  s->avctx->frame_size = s->lsf ? 576 : 1152;
1484  default:
1485  nb_frames = mp_decode_layer3(s);
1486 
1487  s->last_buf_size=0;
1488  if (s->in_gb.buffer) {
1489  align_get_bits(&s->gb);
1490  i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
1491  if (i >= 0 && i <= BACKSTEP_SIZE) {
1492  memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb) >> 3), i);
1493  s->last_buf_size=i;
1494  } else
1495  av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1496  s->gb = s->in_gb;
1497  s->in_gb.buffer = NULL;
1498  s->extrasize = 0;
1499  }
1500 
1501  align_get_bits(&s->gb);
1502  av_assert1((get_bits_count(&s->gb) & 7) == 0);
1503  i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
1504  if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1505  if (i < 0)
1506  av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1507  i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1508  }
1509  av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
1510  memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1511  s->last_buf_size += i;
1512  }
1513 
1514  if(nb_frames < 0)
1515  return nb_frames;
1516 
1517  /* get output buffer */
1518  if (!samples) {
1519  av_assert0(s->frame);
1520  s->frame->nb_samples = s->avctx->frame_size;
1521  if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
1522  return ret;
1523  samples = (OUT_INT **)s->frame->extended_data;
1524  }
1525 
1526  /* apply the synthesis filter */
1527  for (ch = 0; ch < s->nb_channels; ch++) {
1528  int sample_stride;
1529  if (s->avctx->sample_fmt == OUT_FMT_P) {
1530  samples_ptr = samples[ch];
1531  sample_stride = 1;
1532  } else {
1533  samples_ptr = samples[0] + ch;
1534  sample_stride = s->nb_channels;
1535  }
1536  for (i = 0; i < nb_frames; i++) {
1537  RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
1538  &(s->synth_buf_offset[ch]),
1539  RENAME(ff_mpa_synth_window),
1540  &s->dither_state, samples_ptr,
1541  sample_stride, s->sb_samples[ch][i]);
1542  samples_ptr += 32 * sample_stride;
1543  }
1544  }
1545 
1546  return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1547 }
1548 
1549 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1550  AVPacket *avpkt)
1551 {
1552  const uint8_t *buf = avpkt->data;
1553  int buf_size = avpkt->size;
1554  MPADecodeContext *s = avctx->priv_data;
1555  uint32_t header;
1556  int ret;
1557 
1558  int skipped = 0;
1559  while(buf_size && !*buf){
1560  buf++;
1561  buf_size--;
1562  skipped++;
1563  }
1564 
1565  if (buf_size < HEADER_SIZE)
1566  return AVERROR_INVALIDDATA;
1567 
1568  header = AV_RB32(buf);
1569  if (header >> 8 == AV_RB32("TAG") >> 8) {
1570  av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
1571  return buf_size + skipped;
1572  }
1574  if (ret < 0) {
1575  av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1576  return AVERROR_INVALIDDATA;
1577  } else if (ret == 1) {
1578  /* free format: prepare to compute frame size */
1579  s->frame_size = -1;
1580  return AVERROR_INVALIDDATA;
1581  }
1582  /* update codec info */
1583  avctx->channels = s->nb_channels;
1584  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1585  if (!avctx->bit_rate)
1586  avctx->bit_rate = s->bit_rate;
1587 
1588  if (s->frame_size <= 0) {
1589  av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1590  return AVERROR_INVALIDDATA;
1591  } else if (s->frame_size < buf_size) {
1592  av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
1593  buf_size= s->frame_size;
1594  }
1595 
1596  s->frame = data;
1597 
1598  ret = mp_decode_frame(s, NULL, buf, buf_size);
1599  if (ret >= 0) {
1600  s->frame->nb_samples = avctx->frame_size;
1601  *got_frame_ptr = 1;
1602  avctx->sample_rate = s->sample_rate;
1603  //FIXME maybe move the other codec info stuff from above here too
1604  } else {
1605  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1606  /* Only return an error if the bad frame makes up the whole packet or
1607  * the error is related to buffer management.
1608  * If there is more data in the packet, just consume the bad frame
1609  * instead of returning an error, which would discard the whole
1610  * packet. */
1611  *got_frame_ptr = 0;
1612  if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1613  return ret;
1614  }
1615  s->frame_size = 0;
1616  return buf_size + skipped;
1617 }
1618 
1620 {
1621  memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1622  memset(ctx->mdct_buf, 0, sizeof(ctx->mdct_buf));
1623  ctx->last_buf_size = 0;
1624  ctx->dither_state = 0;
1625 }
1626 
1627 static void flush(AVCodecContext *avctx)
1628 {
1629  mp_flush(avctx->priv_data);
1630 }
1631 
1632 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1633 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1634  int *got_frame_ptr, AVPacket *avpkt)
1635 {
1636  const uint8_t *buf = avpkt->data;
1637  int buf_size = avpkt->size;
1638  MPADecodeContext *s = avctx->priv_data;
1639  uint32_t header;
1640  int len, ret;
1641  int av_unused out_size;
1642 
1643  len = buf_size;
1644 
1645  // Discard too short frames
1646  if (buf_size < HEADER_SIZE) {
1647  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1648  return AVERROR_INVALIDDATA;
1649  }
1650 
1651 
1654 
1655  // Get header and restore sync word
1656  header = AV_RB32(buf) | 0xffe00000;
1657 
1659  if (ret < 0) {
1660  av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1661  return ret;
1662  }
1663  /* update codec info */
1664  avctx->sample_rate = s->sample_rate;
1665  avctx->channels = s->nb_channels;
1666  avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1667  if (!avctx->bit_rate)
1668  avctx->bit_rate = s->bit_rate;
1669 
1670  s->frame_size = len;
1671 
1672  s->frame = data;
1673 
1674  ret = mp_decode_frame(s, NULL, buf, buf_size);
1675  if (ret < 0) {
1676  av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1677  return ret;
1678  }
1679 
1680  *got_frame_ptr = 1;
1681 
1682  return buf_size;
1683 }
1684 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1685 
1686 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1687 
1688 /**
1689  * Context for MP3On4 decoder
1690  */
1691 typedef struct MP3On4DecodeContext {
1692  int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
1693  int syncword; ///< syncword patch
1694  const uint8_t *coff; ///< channel offsets in output buffer
1695  MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
1696 } MP3On4DecodeContext;
1697 
1698 #include "mpeg4audio.h"
1699 
1700 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1701 
1702 /* number of mp3 decoder instances */
1703 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1704 
1705 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1706 static const uint8_t chan_offset[8][5] = {
1707  { 0 },
1708  { 0 }, // C
1709  { 0 }, // FLR
1710  { 2, 0 }, // C FLR
1711  { 2, 0, 3 }, // C FLR BS
1712  { 2, 0, 3 }, // C FLR BLRS
1713  { 2, 0, 4, 3 }, // C FLR BLRS LFE
1714  { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1715 };
1716 
1717 /* mp3on4 channel layouts */
1718 static const int16_t chan_layout[8] = {
1719  0,
1727 };
1728 
1729 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1730 {
1731  MP3On4DecodeContext *s = avctx->priv_data;
1732  int i;
1733 
1734  for (i = 0; i < s->frames; i++)
1735  av_freep(&s->mp3decctx[i]);
1736 
1737  return 0;
1738 }
1739 
1740 
1741 static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
1742 {
1743  MP3On4DecodeContext *s = avctx->priv_data;
1744  MPEG4AudioConfig cfg;
1745  int i, ret;
1746 
1747  if ((avctx->extradata_size < 2) || !avctx->extradata) {
1748  av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1749  return AVERROR_INVALIDDATA;
1750  }
1751 
1753  avctx->extradata_size, 1, avctx);
1754  if (!cfg.chan_config || cfg.chan_config > 7) {
1755  av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1756  return AVERROR_INVALIDDATA;
1757  }
1758  s->frames = mp3Frames[cfg.chan_config];
1759  s->coff = chan_offset[cfg.chan_config];
1761  avctx->channel_layout = chan_layout[cfg.chan_config];
1762 
1763  if (cfg.sample_rate < 16000)
1764  s->syncword = 0xffe00000;
1765  else
1766  s->syncword = 0xfff00000;
1767 
1768  /* Init the first mp3 decoder in standard way, so that all tables get builded
1769  * We replace avctx->priv_data with the context of the first decoder so that
1770  * decode_init() does not have to be changed.
1771  * Other decoders will be initialized here copying data from the first context
1772  */
1773  // Allocate zeroed memory for the first decoder context
1774  s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1775  if (!s->mp3decctx[0])
1776  return AVERROR(ENOMEM);
1777  // Put decoder context in place to make init_decode() happy
1778  avctx->priv_data = s->mp3decctx[0];
1779  ret = decode_init(avctx);
1780  // Restore mp3on4 context pointer
1781  avctx->priv_data = s;
1782  if (ret < 0)
1783  return ret;
1784  s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1785 
1786  /* Create a separate codec/context for each frame (first is already ok).
1787  * Each frame is 1 or 2 channels - up to 5 frames allowed
1788  */
1789  for (i = 1; i < s->frames; i++) {
1790  s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1791  if (!s->mp3decctx[i])
1792  return AVERROR(ENOMEM);
1793  s->mp3decctx[i]->adu_mode = 1;
1794  s->mp3decctx[i]->avctx = avctx;
1795  s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1796  s->mp3decctx[i]->butterflies_float = s->mp3decctx[0]->butterflies_float;
1797  }
1798 
1799  return 0;
1800 }
1801 
1802 
1803 static void flush_mp3on4(AVCodecContext *avctx)
1804 {
1805  int i;
1806  MP3On4DecodeContext *s = avctx->priv_data;
1807 
1808  for (i = 0; i < s->frames; i++)
1809  mp_flush(s->mp3decctx[i]);
1810 }
1811 
1812 
1813 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1814  int *got_frame_ptr, AVPacket *avpkt)
1815 {
1816  AVFrame *frame = data;
1817  const uint8_t *buf = avpkt->data;
1818  int buf_size = avpkt->size;
1819  MP3On4DecodeContext *s = avctx->priv_data;
1820  MPADecodeContext *m;
1821  int fsize, len = buf_size, out_size = 0;
1822  uint32_t header;
1823  OUT_INT **out_samples;
1824  OUT_INT *outptr[2];
1825  int fr, ch, ret;
1826 
1827  /* get output buffer */
1829  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1830  return ret;
1831  out_samples = (OUT_INT **)frame->extended_data;
1832 
1833  // Discard too short frames
1834  if (buf_size < HEADER_SIZE)
1835  return AVERROR_INVALIDDATA;
1836 
1837  avctx->bit_rate = 0;
1838 
1839  ch = 0;
1840  for (fr = 0; fr < s->frames; fr++) {
1841  fsize = AV_RB16(buf) >> 4;
1843  m = s->mp3decctx[fr];
1844  av_assert1(m);
1845 
1846  if (fsize < HEADER_SIZE) {
1847  av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1848  return AVERROR_INVALIDDATA;
1849  }
1850  header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1851 
1853  if (ret < 0) {
1854  av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n");
1855  return AVERROR_INVALIDDATA;
1856  }
1857 
1858  if (ch + m->nb_channels > avctx->channels ||
1859  s->coff[fr] + m->nb_channels > avctx->channels) {
1860  av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1861  "channel count\n");
1862  return AVERROR_INVALIDDATA;
1863  }
1864  ch += m->nb_channels;
1865 
1866  outptr[0] = out_samples[s->coff[fr]];
1867  if (m->nb_channels > 1)
1868  outptr[1] = out_samples[s->coff[fr] + 1];
1869 
1870  if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0) {
1871  av_log(avctx, AV_LOG_ERROR, "failed to decode channel %d\n", ch);
1872  memset(outptr[0], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
1873  if (m->nb_channels > 1)
1874  memset(outptr[1], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
1875  ret = m->nb_channels * MPA_FRAME_SIZE*sizeof(OUT_INT);
1876  }
1877 
1878  out_size += ret;
1879  buf += fsize;
1880  len -= fsize;
1881 
1882  avctx->bit_rate += m->bit_rate;
1883  }
1884  if (ch != avctx->channels) {
1885  av_log(avctx, AV_LOG_ERROR, "failed to decode all channels\n");
1886  return AVERROR_INVALIDDATA;
1887  }
1888 
1889  /* update codec info */
1890  avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1891 
1892  frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1893  *got_frame_ptr = 1;
1894 
1895  return buf_size;
1896 }
1897 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
#define FIXR(x)
Definition: aac_defines.h:94
float INTFLOAT
Definition: aac_defines.h:88
static int bit_alloc(AC3EncodeContext *s, int snr_offset)
Run the bit allocation with a given SNR offset.
Definition: ac3enc.c:1131
static double val(void *priv, double ch)
Definition: aeval.c:76
static double bound(const double threshold, const double val)
static float win(SuperEqualizerContext *s, float n, int N)
Macro definitions for various function/variable attributes.
#define av_always_inline
Definition: attributes.h:45
#define av_unused
Definition: attributes.h:131
#define av_cold
Definition: attributes.h:88
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
int32_t
static int64_t fsize(FILE *f)
Definition: audiomatch.c:29
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Libavcodec external API header.
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:1654
#define AV_EF_CRCCHECK
Verify checksums embedded in the bitstream (could be of either encoded or decoded data,...
Definition: avcodec.h:1653
#define AV_EF_COMPLIANT
consider all spec non compliances as errors
Definition: avcodec.h:1660
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:1656
#define AV_EF_BUFFER
detect improper bitstream length
Definition: avcodec.h:1655
#define AV_EF_AGGRESSIVE
consider things that a sane encoder should not do as an error
Definition: avcodec.h:1661
#define AV_RB32
Definition: intreadwrite.h:130
#define AV_RB16
Definition: intreadwrite.h:53
static int alloc_table(VLC *vlc, int size, int use_static)
Definition: bitstream.c:115
#define s(width, name)
Definition: cbs_vp9.c:257
#define f(width, name)
Definition: cbs_vp9.c:255
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:105
#define av_clip
Definition: common.h:122
#define FFMAX(a, b)
Definition: common.h:103
#define FFMIN3(a, b, c)
Definition: common.h:106
Reference: libavcodec/mpegaudiodec.c.
Reference: libavcodec/mpegaudiodec.c.
#define av_restrict
Definition: config.h:11
#define NULL
Definition: coverity.c:32
Public header for CRC hash function implementation.
#define SUINT
#define INTFLOAT
#define SUINTFLOAT
#define MULH3(x, y, s)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1893
static AVFrame * frame
double value
Definition: eval.c:98
#define RENAME(name)
Definition: ffv1.h:196
#define FRAC_BITS
bitstream reader API header.
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:797
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:693
static av_always_inline int get_bitsz(GetBitContext *s, int n)
Read 0-25 bits.
Definition: get_bits.h:415
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
#define AV_CH_LAYOUT_4POINT0
#define AV_CH_LAYOUT_5POINT0
#define AV_CH_LAYOUT_7POINT1
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_SURROUND
#define AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_5POINT1
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
@ AV_CODEC_ID_MP3ON4
Definition: codec_id.h:438
@ AV_CODEC_ID_MP3ADU
Definition: codec_id.h:437
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
Definition: crc.c:374
uint32_t AVCRC
Definition: crc.h:47
uint32_t av_crc(const AVCRC *ctx, uint32_t crc, const uint8_t *buffer, size_t length)
Calculate the CRC of a block.
Definition: crc.c:392
@ AV_CRC_16_ANSI
Definition: crc.h:51
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#define AV_LOG_INFO
Standard information.
Definition: log.h:205
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:117
for(j=16;j >0;--j)
int i
Definition: input.c:407
#define AV_WB32(p, v)
Definition: intreadwrite.h:419
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define AVOnce
Definition: thread.h:172
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:175
#define AV_ONCE_INIT
Definition: thread.h:173
Replacements for frequently missing libm functions.
#define exp2(x)
Definition: libm.h:288
const int16_t * tab1
Definition: mace.c:144
#define MUL64(a, b)
Definition: mathops.h:54
const uint8_t ff_mpeg4audio_channels[14]
Definition: mpeg4audio.c:67
int avpriv_mpeg4audio_get_config2(MPEG4AudioConfig *c, const uint8_t *buf, int size, int sync_extension, void *logctx)
Parse MPEG-4 systems extradata from a raw buffer to retrieve audio configuration.
Definition: mpeg4audio.c:190
int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf)
Definition: mpegaudio.c:31
mpeg audio declarations for both encoder and decoder.
#define MPA_FRAME_SIZE
Definition: mpegaudio.h:37
#define FRAC_ONE
Definition: mpegaudio.h:58
#define MPA_MAX_CHANNELS
Definition: mpegaudio.h:42
#define MPA_JSTEREO
Definition: mpegaudio.h:47
int16_t OUT_INT
Definition: mpegaudio.h:76
#define SBLIMIT
Definition: mpegaudio.h:44
int32_t MPA_INT
Definition: mpegaudio.h:75
#define MPA_MAX_CODED_FRAME_SIZE
Definition: mpegaudio.h:40
static av_cold void mpegaudio_tableinit(void)
const int ff_mpa_quant_bits[17]
Definition: mpegaudiodata.c:55
const unsigned char *const ff_mpa_alloc_tables[5]
const int ff_mpa_sblimit_table[5]
Definition: mpegaudiodata.c:45
const int ff_mpa_quant_steps[17]
Definition: mpegaudiodata.c:47
mpeg audio layer common tables.
VLC ff_huff_quad_vlc[2]
#define MODE_EXT_MS_STEREO
Definition: mpegaudiodata.h:37
uint16_t ff_scale_factor_modshift[64]
const uint8_t ff_band_size_long[9][22]
const uint8_t ff_mpa_huff_data[32][2]
VLC ff_huff_vlc[16]
uint16_t ff_band_index_long[9][23]
const uint8_t ff_lsf_nsf_table[6][3][4]
uint32_t ff_table_4_3_value[TABLE_4_3_SIZE]
const uint8_t ff_slen_table[2][16]
const uint8_t ff_mpa_pretab[2][22]
#define MODE_EXT_I_STEREO
Definition: mpegaudiodata.h:38
int8_t ff_table_4_3_exp[TABLE_4_3_SIZE]
int16_t *const ff_division_tabs[4]
void ff_mpegaudiodec_common_init_static(void)
const uint8_t ff_band_size_short[9][13]
static const int32_t is_table[2][16]
#define OUT_FMT_P
#define MULLx(x, y, s)
#define SHR(a, b)
#define OUT_FMT
static int huffman_decode(MPADecodeContext *s, GranuleDef *g, int16_t *exponents, int end_pos2)
static INTFLOAT is_table_lsf[2][2][16]
static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g, int16_t *exponents)
#define ISQRT2
static av_cold void decode_init_static(void)
static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2)
static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2, int n3)
static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
static void init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
#define C5
#define C6
static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
#define HEADER_SIZE
#define SCALE_GEN(v)
static int l2_unscale_group(int steps, int mant, int scale_factor)
#define LAST_BUF_SIZE
#define C3
static void region_offset2size(GranuleDef *g)
Convert region offsets to region sizes and truncate size to big_values.
static av_cold int decode_init(AVCodecContext *avctx)
static int handle_crc(MPADecodeContext *s, int sec_len)
static int mp_decode_layer2(MPADecodeContext *s)
#define BACKSTEP_SIZE
static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples, const uint8_t *buf, int buf_size)
#define SPLIT(dst, sf, n)
static int32_t scale_factor_mult[15][3]
static int l3_unscale(int value, int exponent)
static int mp_decode_layer3(MPADecodeContext *s)
static int mp_decode_layer1(MPADecodeContext *s)
static void imdct12(INTFLOAT *out, SUINTFLOAT *in)
static void init_short_region(MPADecodeContext *s, GranuleDef *g)
#define READ_FLIP_SIGN(dst, src)
static void reorder_block(MPADecodeContext *s, GranuleDef *g)
#define C4
static int l1_unscale(int n, int mant, int scale_factor)
static const int32_t scale_factor_mult2[3][3]
static void mp_flush(MPADecodeContext *ctx)
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define AA(j)
static void compute_imdct(MPADecodeContext *s, GranuleDef *g, INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
static void flush(AVCodecContext *avctx)
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
MPEG Audio header decoder.
#define MPA_DECODE_HEADER
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:81
void RENAME() ff_mpa_synth_filter(MPADSPContext *s, MPA_INT *synth_buf_ptr, int *synth_buf_offset, MPA_INT *window, int *dither_state, OUT_INT *samples, ptrdiff_t incr, MPA_INT *sb_samples)
av_cold void RENAME() ff_mpa_synth_init(void)
const char data[16]
Definition: mxf.c:142
static const uint16_t table[]
Definition: prosumer.c:206
#define v0
Definition: regdef.h:26
#define t1
Definition: regdef.h:29
#define t2
Definition: regdef.h:30
typedef void(RENAME(mix_any_func_type))
static const uint8_t header[24]
Definition: sdr2.c:67
static int shift(int a, int b)
Definition: sonic.c:82
const uint8_t * code
Definition: spdifenc.c:413
unsigned int pos
Definition: spdifenc.c:412
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
enum AVSampleFormat request_sample_fmt
desired sample format
Definition: avcodec.h:1269
int sample_rate
samples per second
Definition: avcodec.h:1196
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:616
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int channels
number of audio channels
Definition: avcodec.h:1197
enum AVCodecID codec_id
Definition: avcodec.h:546
int extradata_size
Definition: avcodec.h:638
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
void * priv_data
Definition: avcodec.h:563
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:1645
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
Definition: float_dsp.h:164
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
uint8_t scale_factors[40]
int sb_hybrid[SBLIMIT *18]
uint8_t count1table_select
int adu_mode
0 for standard mp3, 1 for adu formatted mp3
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT *18]
AVCodecContext * avctx
int sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]
int synth_buf_offset[MPA_MAX_CHANNELS]
MPA_INT synth_buf[MPA_MAX_CHANNELS][512 *2]
MPA_DECODE_HEADER uint8_t last_buf[LAST_BUF_SIZE]
GranuleDef granules[2][2]
Definition: vlc.h:26
int bits
Definition: vlc.h:27
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
#define av_free(p)
#define ff_dlog(a,...)
#define avpriv_request_sample(...)
#define av_freep(p)
#define av_log(a,...)
static uint8_t tmp[11]
Definition: aes_ctr.c:27
FILE * out
Definition: movenc.c:54
int frames
Definition: movenc.c:66
int out_size
Definition: movenc.c:55
AVFormatContext * ctx
Definition: movenc.c:48
const char * b
Definition: vf_curves.c:118
const char * g
Definition: vf_curves.c:117
if(ret< 0)
Definition: vf_mcdeint.c:282
static int mod(int a, int b)
Modulo operation with only positive remainders.
Definition: vf_v360.c:747
int len
uint8_t bits
Definition: vp3data.h:141