qdm2.c
Go to the documentation of this file.
1 /*
2  * QDM2 compatible decoder
3  * Copyright (c) 2003 Ewald Snel
4  * Copyright (c) 2005 Benjamin Larsson
5  * Copyright (c) 2005 Alex Beregszaszi
6  * Copyright (c) 2005 Roberto Togni
7  *
8  * This file is part of Libav.
9  *
10  * Libav is free software; you can redistribute it and/or
11  * modify it under the terms of the GNU Lesser General Public
12  * License as published by the Free Software Foundation; either
13  * version 2.1 of the License, or (at your option) any later version.
14  *
15  * Libav is distributed in the hope that it will be useful,
16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18  * Lesser General Public License for more details.
19  *
20  * You should have received a copy of the GNU Lesser General Public
21  * License along with Libav; if not, write to the Free Software
22  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23  */
24 
34 #include <math.h>
35 #include <stddef.h>
36 #include <stdio.h>
37 
38 #define BITSTREAM_READER_LE
39 #include "avcodec.h"
40 #include "get_bits.h"
41 #include "dsputil.h"
42 #include "rdft.h"
43 #include "mpegaudiodsp.h"
44 #include "mpegaudio.h"
45 
46 #include "qdm2data.h"
47 #include "qdm2_tablegen.h"
48 
49 #undef NDEBUG
50 #include <assert.h>
51 
52 
53 #define QDM2_LIST_ADD(list, size, packet) \
54 do { \
55  if (size > 0) { \
56  list[size - 1].next = &list[size]; \
57  } \
58  list[size].packet = packet; \
59  list[size].next = NULL; \
60  size++; \
61 } while(0)
62 
63 // Result is 8, 16 or 30
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
65 
66 #define FIX_NOISE_IDX(noise_idx) \
67  if ((noise_idx) >= 3840) \
68  (noise_idx) -= 3840; \
69 
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
71 
72 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
73 
74 #define SAMPLES_NEEDED \
75  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
76 
77 #define SAMPLES_NEEDED_2(why) \
78  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
79 
80 #define QDM2_MAX_FRAME_SIZE 512
81 
82 typedef int8_t sb_int8_array[2][30][64];
83 
87 typedef struct {
88  int type;
89  unsigned int size;
90  const uint8_t *data;
92 
96 typedef struct QDM2SubPNode {
98  struct QDM2SubPNode *next;
99 } QDM2SubPNode;
100 
101 typedef struct {
102  float re;
103  float im;
104 } QDM2Complex;
105 
106 typedef struct {
107  float level;
109  const float *table;
110  int phase;
112  int duration;
113  short time_index;
114  short cutoff;
115 } FFTTone;
116 
117 typedef struct {
118  int16_t sub_packet;
119  uint8_t channel;
120  int16_t offset;
121  int16_t exp;
122  uint8_t phase;
124 
125 typedef struct {
127 } QDM2FFT;
128 
132 typedef struct {
134 
137  int channels;
139  int fft_size;
141 
144  int fft_order;
151 
153  QDM2SubPacket sub_packets[16];
154  QDM2SubPNode sub_packet_list_A[16];
155  QDM2SubPNode sub_packet_list_B[16];
157  QDM2SubPNode sub_packet_list_C[16];
158  QDM2SubPNode sub_packet_list_D[16];
159 
161  FFTTone fft_tones[1000];
164  FFTCoefficient fft_coefs[1000];
166  int fft_coefs_min_index[5];
167  int fft_coefs_max_index[5];
168  int fft_level_exp[6];
171 
173  const uint8_t *compressed_data;
175  float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
176 
179  DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
180  int synth_buf_offset[MPA_MAX_CHANNELS];
181  DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
183 
185  float tone_level[MPA_MAX_CHANNELS][30][64];
186  int8_t coding_method[MPA_MAX_CHANNELS][30][64];
187  int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
188  int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
189  int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
190  int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
191  int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
192  int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
193  int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
194 
195  // Flags
199 
201  int noise_idx;
202 } QDM2Context;
203 
204 
206 
220 
221 static const uint16_t qdm2_vlc_offs[] = {
222  0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
223 };
224 
225 static av_cold void qdm2_init_vlc(void)
226 {
227  static int vlcs_initialized = 0;
228  static VLC_TYPE qdm2_table[3838][2];
229 
230  if (!vlcs_initialized) {
231 
232  vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
233  vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
234  init_vlc (&vlc_tab_level, 8, 24,
237 
238  vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
239  vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
240  init_vlc (&vlc_tab_diff, 8, 37,
241  vlc_tab_diff_huffbits, 1, 1,
243 
244  vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
245  vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
246  init_vlc (&vlc_tab_run, 5, 6,
247  vlc_tab_run_huffbits, 1, 1,
249 
250  fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
251  fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
252  init_vlc (&fft_level_exp_alt_vlc, 8, 28,
255 
256 
257  fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
258  fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
259  init_vlc (&fft_level_exp_vlc, 8, 20,
262 
263  fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
264  fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
265  init_vlc (&fft_stereo_exp_vlc, 6, 7,
268 
269  fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
270  fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
271  init_vlc (&fft_stereo_phase_vlc, 6, 9,
274 
275  vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
276  vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
277  init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
280 
281  vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
282  vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
283  init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
286 
287  vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
288  vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
289  init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
292 
293  vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
294  vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
295  init_vlc (&vlc_tab_type30, 6, 9,
298 
299  vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
300  vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
301  init_vlc (&vlc_tab_type34, 5, 10,
304 
305  vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
306  vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
307  init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
310 
311  vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
312  vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
313  init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
316 
317  vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
318  vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
319  init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
322 
323  vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
324  vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
325  init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
328 
329  vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
330  vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
331  init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
334 
335  vlcs_initialized=1;
336  }
337 }
338 
339 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
340 {
341  int value;
342 
343  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
344 
345  /* stage-2, 3 bits exponent escape sequence */
346  if (value-- == 0)
347  value = get_bits (gb, get_bits (gb, 3) + 1);
348 
349  /* stage-3, optional */
350  if (flag) {
351  int tmp = vlc_stage3_values[value];
352 
353  if ((value & ~3) > 0)
354  tmp += get_bits (gb, (value >> 2));
355  value = tmp;
356  }
357 
358  return value;
359 }
360 
361 
362 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
363 {
364  int value = qdm2_get_vlc (gb, vlc, 0, depth);
365 
366  return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
367 }
368 
369 
379 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
380  int i;
381 
382  for (i=0; i < length; i++)
383  value -= data[i];
384 
385  return (uint16_t)(value & 0xffff);
386 }
387 
388 
396 {
397  sub_packet->type = get_bits (gb, 8);
398 
399  if (sub_packet->type == 0) {
400  sub_packet->size = 0;
401  sub_packet->data = NULL;
402  } else {
403  sub_packet->size = get_bits (gb, 8);
404 
405  if (sub_packet->type & 0x80) {
406  sub_packet->size <<= 8;
407  sub_packet->size |= get_bits (gb, 8);
408  sub_packet->type &= 0x7f;
409  }
410 
411  if (sub_packet->type == 0x7f)
412  sub_packet->type |= (get_bits (gb, 8) << 8);
413 
414  sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
415  }
416 
417  av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
418  sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
419 }
420 
421 
430 {
431  while (list != NULL && list->packet != NULL) {
432  if (list->packet->type == type)
433  return list;
434  list = list->next;
435  }
436  return NULL;
437 }
438 
439 
447 {
448  int i, j, n, ch, sum;
449 
451 
452  for (ch = 0; ch < q->nb_channels; ch++)
453  for (i = 0; i < n; i++) {
454  sum = 0;
455 
456  for (j = 0; j < 8; j++)
457  sum += q->quantized_coeffs[ch][i][j];
458 
459  sum /= 8;
460  if (sum > 0)
461  sum--;
462 
463  for (j=0; j < 8; j++)
464  q->quantized_coeffs[ch][i][j] = sum;
465  }
466 }
467 
468 
476 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
477 {
478  int ch, j;
479 
481 
482  if (!q->nb_channels)
483  return;
484 
485  for (ch = 0; ch < q->nb_channels; ch++)
486  for (j = 0; j < 64; j++) {
487  q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
488  q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
489  }
490 }
491 
492 
501 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
502 {
503  int j,k;
504  int ch;
505  int run, case_val;
506  int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
507 
508  for (ch = 0; ch < channels; ch++) {
509  for (j = 0; j < 64; ) {
510  if((coding_method[ch][sb][j] - 8) > 22) {
511  run = 1;
512  case_val = 8;
513  } else {
514  switch (switchtable[coding_method[ch][sb][j]-8]) {
515  case 0: run = 10; case_val = 10; break;
516  case 1: run = 1; case_val = 16; break;
517  case 2: run = 5; case_val = 24; break;
518  case 3: run = 3; case_val = 30; break;
519  case 4: run = 1; case_val = 30; break;
520  case 5: run = 1; case_val = 8; break;
521  default: run = 1; case_val = 8; break;
522  }
523  }
524  for (k = 0; k < run; k++)
525  if (j + k < 128)
526  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
527  if (k > 0) {
529  //not debugged, almost never used
530  memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
531  memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
532  }
533  j += run;
534  }
535  }
536 }
537 
538 
546 static void fill_tone_level_array (QDM2Context *q, int flag)
547 {
548  int i, sb, ch, sb_used;
549  int tmp, tab;
550 
551  // This should never happen
552  if (q->nb_channels <= 0)
553  return;
554 
555  for (ch = 0; ch < q->nb_channels; ch++)
556  for (sb = 0; sb < 30; sb++)
557  for (i = 0; i < 8; i++) {
559  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
561  else
562  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
563  if(tmp < 0)
564  tmp += 0xff;
565  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
566  }
567 
568  sb_used = QDM2_SB_USED(q->sub_sampling);
569 
570  if ((q->superblocktype_2_3 != 0) && !flag) {
571  for (sb = 0; sb < sb_used; sb++)
572  for (ch = 0; ch < q->nb_channels; ch++)
573  for (i = 0; i < 64; i++) {
574  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
575  if (q->tone_level_idx[ch][sb][i] < 0)
576  q->tone_level[ch][sb][i] = 0;
577  else
578  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
579  }
580  } else {
581  tab = q->superblocktype_2_3 ? 0 : 1;
582  for (sb = 0; sb < sb_used; sb++) {
583  if ((sb >= 4) && (sb <= 23)) {
584  for (ch = 0; ch < q->nb_channels; ch++)
585  for (i = 0; i < 64; i++) {
586  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
587  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
588  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
589  q->tone_level_idx_hi2[ch][sb - 4];
590  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
591  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
592  q->tone_level[ch][sb][i] = 0;
593  else
594  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
595  }
596  } else {
597  if (sb > 4) {
598  for (ch = 0; ch < q->nb_channels; ch++)
599  for (i = 0; i < 64; i++) {
600  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
601  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
602  q->tone_level_idx_hi2[ch][sb - 4];
603  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
604  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
605  q->tone_level[ch][sb][i] = 0;
606  else
607  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
608  }
609  } else {
610  for (ch = 0; ch < q->nb_channels; ch++)
611  for (i = 0; i < 64; i++) {
612  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
613  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
614  q->tone_level[ch][sb][i] = 0;
615  else
616  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
617  }
618  }
619  }
620  }
621  }
622 
623  return;
624 }
625 
626 
641 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
642  sb_int8_array coding_method, int nb_channels,
643  int c, int superblocktype_2_3, int cm_table_select)
644 {
645  int ch, sb, j;
646  int tmp, acc, esp_40, comp;
647  int add1, add2, add3, add4;
648  int64_t multres;
649 
650  // This should never happen
651  if (nb_channels <= 0)
652  return;
653 
654  if (!superblocktype_2_3) {
655  /* This case is untested, no samples available */
657  for (ch = 0; ch < nb_channels; ch++)
658  for (sb = 0; sb < 30; sb++) {
659  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
660  add1 = tone_level_idx[ch][sb][j] - 10;
661  if (add1 < 0)
662  add1 = 0;
663  add2 = add3 = add4 = 0;
664  if (sb > 1) {
665  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
666  if (add2 < 0)
667  add2 = 0;
668  }
669  if (sb > 0) {
670  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
671  if (add3 < 0)
672  add3 = 0;
673  }
674  if (sb < 29) {
675  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
676  if (add4 < 0)
677  add4 = 0;
678  }
679  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
680  if (tmp < 0)
681  tmp = 0;
682  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
683  }
684  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
685  }
686  acc = 0;
687  for (ch = 0; ch < nb_channels; ch++)
688  for (sb = 0; sb < 30; sb++)
689  for (j = 0; j < 64; j++)
690  acc += tone_level_idx_temp[ch][sb][j];
691 
692  multres = 0x66666667 * (acc * 10);
693  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
694  for (ch = 0; ch < nb_channels; ch++)
695  for (sb = 0; sb < 30; sb++)
696  for (j = 0; j < 64; j++) {
697  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
698  if (comp < 0)
699  comp += 0xff;
700  comp /= 256; // signed shift
701  switch(sb) {
702  case 0:
703  if (comp < 30)
704  comp = 30;
705  comp += 15;
706  break;
707  case 1:
708  if (comp < 24)
709  comp = 24;
710  comp += 10;
711  break;
712  case 2:
713  case 3:
714  case 4:
715  if (comp < 16)
716  comp = 16;
717  }
718  if (comp <= 5)
719  tmp = 0;
720  else if (comp <= 10)
721  tmp = 10;
722  else if (comp <= 16)
723  tmp = 16;
724  else if (comp <= 24)
725  tmp = -1;
726  else
727  tmp = 0;
728  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
729  }
730  for (sb = 0; sb < 30; sb++)
731  fix_coding_method_array(sb, nb_channels, coding_method);
732  for (ch = 0; ch < nb_channels; ch++)
733  for (sb = 0; sb < 30; sb++)
734  for (j = 0; j < 64; j++)
735  if (sb >= 10) {
736  if (coding_method[ch][sb][j] < 10)
737  coding_method[ch][sb][j] = 10;
738  } else {
739  if (sb >= 2) {
740  if (coding_method[ch][sb][j] < 16)
741  coding_method[ch][sb][j] = 16;
742  } else {
743  if (coding_method[ch][sb][j] < 30)
744  coding_method[ch][sb][j] = 30;
745  }
746  }
747  } else { // superblocktype_2_3 != 0
748  for (ch = 0; ch < nb_channels; ch++)
749  for (sb = 0; sb < 30; sb++)
750  for (j = 0; j < 64; j++)
751  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
752  }
753 
754  return;
755 }
756 
757 
769 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
770 {
771  int sb, j, k, n, ch, run, channels;
772  int joined_stereo, zero_encoding, chs;
773  int type34_first;
774  float type34_div = 0;
775  float type34_predictor;
776  float samples[10], sign_bits[16];
777 
778  if (length == 0) {
779  // If no data use noise
780  for (sb=sb_min; sb < sb_max; sb++)
782 
783  return;
784  }
785 
786  for (sb = sb_min; sb < sb_max; sb++) {
788 
789  channels = q->nb_channels;
790 
791  if (q->nb_channels <= 1 || sb < 12)
792  joined_stereo = 0;
793  else if (sb >= 24)
794  joined_stereo = 1;
795  else
796  joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
797 
798  if (joined_stereo) {
799  if (BITS_LEFT(length,gb) >= 16)
800  for (j = 0; j < 16; j++)
801  sign_bits[j] = get_bits1 (gb);
802 
803  for (j = 0; j < 64; j++)
804  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
805  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
806 
808  channels = 1;
809  }
810 
811  for (ch = 0; ch < channels; ch++) {
812  zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
813  type34_predictor = 0.0;
814  type34_first = 1;
815 
816  for (j = 0; j < 128; ) {
817  switch (q->coding_method[ch][sb][j / 2]) {
818  case 8:
819  if (BITS_LEFT(length,gb) >= 10) {
820  if (zero_encoding) {
821  for (k = 0; k < 5; k++) {
822  if ((j + 2 * k) >= 128)
823  break;
824  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
825  }
826  } else {
827  n = get_bits(gb, 8);
828  for (k = 0; k < 5; k++)
829  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
830  }
831  for (k = 0; k < 5; k++)
832  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
833  } else {
834  for (k = 0; k < 10; k++)
835  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
836  }
837  run = 10;
838  break;
839 
840  case 10:
841  if (BITS_LEFT(length,gb) >= 1) {
842  float f = 0.81;
843 
844  if (get_bits1(gb))
845  f = -f;
846  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
847  samples[0] = f;
848  } else {
849  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
850  }
851  run = 1;
852  break;
853 
854  case 16:
855  if (BITS_LEFT(length,gb) >= 10) {
856  if (zero_encoding) {
857  for (k = 0; k < 5; k++) {
858  if ((j + k) >= 128)
859  break;
860  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
861  }
862  } else {
863  n = get_bits (gb, 8);
864  for (k = 0; k < 5; k++)
865  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
866  }
867  } else {
868  for (k = 0; k < 5; k++)
869  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
870  }
871  run = 5;
872  break;
873 
874  case 24:
875  if (BITS_LEFT(length,gb) >= 7) {
876  n = get_bits(gb, 7);
877  for (k = 0; k < 3; k++)
878  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
879  } else {
880  for (k = 0; k < 3; k++)
881  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
882  }
883  run = 3;
884  break;
885 
886  case 30:
887  if (BITS_LEFT(length,gb) >= 4) {
888  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
889  if (index < FF_ARRAY_ELEMS(type30_dequant)) {
890  samples[0] = type30_dequant[index];
891  } else
892  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
893  } else
894  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
895 
896  run = 1;
897  break;
898 
899  case 34:
900  if (BITS_LEFT(length,gb) >= 7) {
901  if (type34_first) {
902  type34_div = (float)(1 << get_bits(gb, 2));
903  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
904  type34_predictor = samples[0];
905  type34_first = 0;
906  } else {
907  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
908  if (index < FF_ARRAY_ELEMS(type34_delta)) {
909  samples[0] = type34_delta[index] / type34_div + type34_predictor;
910  type34_predictor = samples[0];
911  } else
912  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
913  }
914  } else {
915  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
916  }
917  run = 1;
918  break;
919 
920  default:
921  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
922  run = 1;
923  break;
924  }
925 
926  if (joined_stereo) {
927  float tmp[10][MPA_MAX_CHANNELS];
928 
929  for (k = 0; k < run; k++) {
930  tmp[k][0] = samples[k];
931  tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
932  }
933  for (chs = 0; chs < q->nb_channels; chs++)
934  for (k = 0; k < run; k++)
935  if ((j + k) < 128)
936  q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
937  } else {
938  for (k = 0; k < run; k++)
939  if ((j + k) < 128)
940  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
941  }
942 
943  j += run;
944  } // j loop
945  } // channel loop
946  } // subband loop
947 }
948 
949 
959 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
960 {
961  int i, k, run, level, diff;
962 
963  if (BITS_LEFT(length,gb) < 16)
964  return;
965  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
966 
967  quantized_coeffs[0] = level;
968 
969  for (i = 0; i < 7; ) {
970  if (BITS_LEFT(length,gb) < 16)
971  break;
972  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
973 
974  if (BITS_LEFT(length,gb) < 16)
975  break;
976  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
977 
978  for (k = 1; k <= run; k++)
979  quantized_coeffs[i + k] = (level + ((k * diff) / run));
980 
981  level += diff;
982  i += run;
983  }
984 }
985 
986 
997 {
998  int sb, j, k, n, ch;
999 
1000  for (ch = 0; ch < q->nb_channels; ch++) {
1001  init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1002 
1003  if (BITS_LEFT(length,gb) < 16) {
1004  memset(q->quantized_coeffs[ch][0], 0, 8);
1005  break;
1006  }
1007  }
1008 
1009  n = q->sub_sampling + 1;
1010 
1011  for (sb = 0; sb < n; sb++)
1012  for (ch = 0; ch < q->nb_channels; ch++)
1013  for (j = 0; j < 8; j++) {
1014  if (BITS_LEFT(length,gb) < 1)
1015  break;
1016  if (get_bits1(gb)) {
1017  for (k=0; k < 8; k++) {
1018  if (BITS_LEFT(length,gb) < 16)
1019  break;
1020  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1021  }
1022  } else {
1023  for (k=0; k < 8; k++)
1024  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1025  }
1026  }
1027 
1028  n = QDM2_SB_USED(q->sub_sampling) - 4;
1029 
1030  for (sb = 0; sb < n; sb++)
1031  for (ch = 0; ch < q->nb_channels; ch++) {
1032  if (BITS_LEFT(length,gb) < 16)
1033  break;
1034  q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1035  if (sb > 19)
1036  q->tone_level_idx_hi2[ch][sb] -= 16;
1037  else
1038  for (j = 0; j < 8; j++)
1039  q->tone_level_idx_mid[ch][sb][j] = -16;
1040  }
1041 
1042  n = QDM2_SB_USED(q->sub_sampling) - 5;
1043 
1044  for (sb = 0; sb < n; sb++)
1045  for (ch = 0; ch < q->nb_channels; ch++)
1046  for (j = 0; j < 8; j++) {
1047  if (BITS_LEFT(length,gb) < 16)
1048  break;
1049  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1050  }
1051 }
1052 
1060 {
1061  GetBitContext gb;
1062  int i, j, k, n, ch, run, level, diff;
1063 
1064  init_get_bits(&gb, node->packet->data, node->packet->size*8);
1065 
1066  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1067 
1068  for (i = 1; i < n; i++)
1069  for (ch=0; ch < q->nb_channels; ch++) {
1070  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1071  q->quantized_coeffs[ch][i][0] = level;
1072 
1073  for (j = 0; j < (8 - 1); ) {
1074  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1075  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1076 
1077  for (k = 1; k <= run; k++)
1078  q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1079 
1080  level += diff;
1081  j += run;
1082  }
1083  }
1084 
1085  for (ch = 0; ch < q->nb_channels; ch++)
1086  for (i = 0; i < 8; i++)
1087  q->quantized_coeffs[ch][0][i] = 0;
1088 }
1089 
1090 
1098 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1099 {
1100  GetBitContext gb;
1101 
1102  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1103 
1104  if (length != 0) {
1105  init_tone_level_dequantization(q, &gb, length);
1106  fill_tone_level_array(q, 1);
1107  } else {
1108  fill_tone_level_array(q, 0);
1109  }
1110 }
1111 
1112 
1120 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1121 {
1122  GetBitContext gb;
1123 
1124  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1125  if (length >= 32) {
1126  int c = get_bits (&gb, 13);
1127 
1128  if (c > 3)
1131  }
1132 
1133  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1134 }
1135 
1136 
1144 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1145 {
1146  GetBitContext gb;
1147 
1148  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1149  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1150 }
1151 
1152 /*
1153  * Process new subpackets for synthesis filter
1154  *
1155  * @param q context
1156  * @param list list with synthesis filter packets (list D)
1157  */
1159 {
1160  QDM2SubPNode *nodes[4];
1161 
1162  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1163  if (nodes[0] != NULL)
1164  process_subpacket_9(q, nodes[0]);
1165 
1166  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1167  if (nodes[1] != NULL)
1168  process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1169  else
1170  process_subpacket_10(q, NULL, 0);
1171 
1172  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1173  if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1174  process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1175  else
1176  process_subpacket_11(q, NULL, 0);
1177 
1178  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1179  if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1180  process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1181  else
1182  process_subpacket_12(q, NULL, 0);
1183 }
1184 
1185 
1186 /*
1187  * Decode superblock, fill packet lists.
1188  *
1189  * @param q context
1190  */
1192 {
1193  GetBitContext gb;
1194  QDM2SubPacket header, *packet;
1195  int i, packet_bytes, sub_packet_size, sub_packets_D;
1196  unsigned int next_index = 0;
1197 
1198  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1199  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1200  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1201 
1202  q->sub_packets_B = 0;
1203  sub_packets_D = 0;
1204 
1205  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1206 
1208  qdm2_decode_sub_packet_header(&gb, &header);
1209 
1210  if (header.type < 2 || header.type >= 8) {
1211  q->has_errors = 1;
1212  av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1213  return;
1214  }
1215 
1216  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1217  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1218 
1219  init_get_bits(&gb, header.data, header.size*8);
1220 
1221  if (header.type == 2 || header.type == 4 || header.type == 5) {
1222  int csum = 257 * get_bits(&gb, 8);
1223  csum += 2 * get_bits(&gb, 8);
1224 
1225  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1226 
1227  if (csum != 0) {
1228  q->has_errors = 1;
1229  av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1230  return;
1231  }
1232  }
1233 
1234  q->sub_packet_list_B[0].packet = NULL;
1235  q->sub_packet_list_D[0].packet = NULL;
1236 
1237  for (i = 0; i < 6; i++)
1238  if (--q->fft_level_exp[i] < 0)
1239  q->fft_level_exp[i] = 0;
1240 
1241  for (i = 0; packet_bytes > 0; i++) {
1242  int j;
1243 
1244  q->sub_packet_list_A[i].next = NULL;
1245 
1246  if (i > 0) {
1247  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1248 
1249  /* seek to next block */
1250  init_get_bits(&gb, header.data, header.size*8);
1251  skip_bits(&gb, next_index*8);
1252 
1253  if (next_index >= header.size)
1254  break;
1255  }
1256 
1257  /* decode subpacket */
1258  packet = &q->sub_packets[i];
1259  qdm2_decode_sub_packet_header(&gb, packet);
1260  next_index = packet->size + get_bits_count(&gb) / 8;
1261  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1262 
1263  if (packet->type == 0)
1264  break;
1265 
1266  if (sub_packet_size > packet_bytes) {
1267  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1268  break;
1269  packet->size += packet_bytes - sub_packet_size;
1270  }
1271 
1272  packet_bytes -= sub_packet_size;
1273 
1274  /* add subpacket to 'all subpackets' list */
1275  q->sub_packet_list_A[i].packet = packet;
1276 
1277  /* add subpacket to related list */
1278  if (packet->type == 8) {
1279  SAMPLES_NEEDED_2("packet type 8");
1280  return;
1281  } else if (packet->type >= 9 && packet->type <= 12) {
1282  /* packets for MPEG Audio like Synthesis Filter */
1283  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1284  } else if (packet->type == 13) {
1285  for (j = 0; j < 6; j++)
1286  q->fft_level_exp[j] = get_bits(&gb, 6);
1287  } else if (packet->type == 14) {
1288  for (j = 0; j < 6; j++)
1289  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1290  } else if (packet->type == 15) {
1291  SAMPLES_NEEDED_2("packet type 15")
1292  return;
1293  } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1294  /* packets for FFT */
1296  }
1297  } // Packet bytes loop
1298 
1299 /* **************************************************************** */
1300  if (q->sub_packet_list_D[0].packet != NULL) {
1302  q->do_synth_filter = 1;
1303  } else if (q->do_synth_filter) {
1304  process_subpacket_10(q, NULL, 0);
1305  process_subpacket_11(q, NULL, 0);
1306  process_subpacket_12(q, NULL, 0);
1307  }
1308 /* **************************************************************** */
1309 }
1310 
1311 
1312 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1313  int offset, int duration, int channel,
1314  int exp, int phase)
1315 {
1316  if (q->fft_coefs_min_index[duration] < 0)
1318 
1319  q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1320  q->fft_coefs[q->fft_coefs_index].channel = channel;
1321  q->fft_coefs[q->fft_coefs_index].offset = offset;
1322  q->fft_coefs[q->fft_coefs_index].exp = exp;
1323  q->fft_coefs[q->fft_coefs_index].phase = phase;
1324  q->fft_coefs_index++;
1325 }
1326 
1327 
1329 {
1330  int channel, stereo, phase, exp;
1331  int local_int_4, local_int_8, stereo_phase, local_int_10;
1332  int local_int_14, stereo_exp, local_int_20, local_int_28;
1333  int n, offset;
1334 
1335  local_int_4 = 0;
1336  local_int_28 = 0;
1337  local_int_20 = 2;
1338  local_int_8 = (4 - duration);
1339  local_int_10 = 1 << (q->group_order - duration - 1);
1340  offset = 1;
1341 
1342  while (1) {
1343  if (q->superblocktype_2_3) {
1344  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1345  offset = 1;
1346  if (n == 0) {
1347  local_int_4 += local_int_10;
1348  local_int_28 += (1 << local_int_8);
1349  } else {
1350  local_int_4 += 8*local_int_10;
1351  local_int_28 += (8 << local_int_8);
1352  }
1353  }
1354  offset += (n - 2);
1355  } else {
1356  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1357  while (offset >= (local_int_10 - 1)) {
1358  offset += (1 - (local_int_10 - 1));
1359  local_int_4 += local_int_10;
1360  local_int_28 += (1 << local_int_8);
1361  }
1362  }
1363 
1364  if (local_int_4 >= q->group_size)
1365  return;
1366 
1367  local_int_14 = (offset >> local_int_8);
1368  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1369  return;
1370 
1371  if (q->nb_channels > 1) {
1372  channel = get_bits1(gb);
1373  stereo = get_bits1(gb);
1374  } else {
1375  channel = 0;
1376  stereo = 0;
1377  }
1378 
1379  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1380  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1381  exp = (exp < 0) ? 0 : exp;
1382 
1383  phase = get_bits(gb, 3);
1384  stereo_exp = 0;
1385  stereo_phase = 0;
1386 
1387  if (stereo) {
1388  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1389  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1390  if (stereo_phase < 0)
1391  stereo_phase += 8;
1392  }
1393 
1394  if (q->frequency_range > (local_int_14 + 1)) {
1395  int sub_packet = (local_int_20 + local_int_28);
1396 
1397  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1398  if (stereo)
1399  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1400  }
1401 
1402  offset++;
1403  }
1404 }
1405 
1406 
1408 {
1409  int i, j, min, max, value, type, unknown_flag;
1410  GetBitContext gb;
1411 
1412  if (q->sub_packet_list_B[0].packet == NULL)
1413  return;
1414 
1415  /* reset minimum indexes for FFT coefficients */
1416  q->fft_coefs_index = 0;
1417  for (i=0; i < 5; i++)
1418  q->fft_coefs_min_index[i] = -1;
1419 
1420  /* process subpackets ordered by type, largest type first */
1421  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1422  QDM2SubPacket *packet= NULL;
1423 
1424  /* find subpacket with largest type less than max */
1425  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1426  value = q->sub_packet_list_B[j].packet->type;
1427  if (value > min && value < max) {
1428  min = value;
1429  packet = q->sub_packet_list_B[j].packet;
1430  }
1431  }
1432 
1433  max = min;
1434 
1435  /* check for errors (?) */
1436  if (!packet)
1437  return;
1438 
1439  if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1440  return;
1441 
1442  /* decode FFT tones */
1443  init_get_bits (&gb, packet->data, packet->size*8);
1444 
1445  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1446  unknown_flag = 1;
1447  else
1448  unknown_flag = 0;
1449 
1450  type = packet->type;
1451 
1452  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1453  int duration = q->sub_sampling + 5 - (type & 15);
1454 
1455  if (duration >= 0 && duration < 4)
1456  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1457  } else if (type == 31) {
1458  for (j=0; j < 4; j++)
1459  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1460  } else if (type == 46) {
1461  for (j=0; j < 6; j++)
1462  q->fft_level_exp[j] = get_bits(&gb, 6);
1463  for (j=0; j < 4; j++)
1464  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1465  }
1466  } // Loop on B packets
1467 
1468  /* calculate maximum indexes for FFT coefficients */
1469  for (i = 0, j = -1; i < 5; i++)
1470  if (q->fft_coefs_min_index[i] >= 0) {
1471  if (j >= 0)
1473  j = i;
1474  }
1475  if (j >= 0)
1477 }
1478 
1479 
1481 {
1482  float level, f[6];
1483  int i;
1484  QDM2Complex c;
1485  const double iscale = 2.0*M_PI / 512.0;
1486 
1487  tone->phase += tone->phase_shift;
1488 
1489  /* calculate current level (maximum amplitude) of tone */
1490  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1491  c.im = level * sin(tone->phase*iscale);
1492  c.re = level * cos(tone->phase*iscale);
1493 
1494  /* generate FFT coefficients for tone */
1495  if (tone->duration >= 3 || tone->cutoff >= 3) {
1496  tone->complex[0].im += c.im;
1497  tone->complex[0].re += c.re;
1498  tone->complex[1].im -= c.im;
1499  tone->complex[1].re -= c.re;
1500  } else {
1501  f[1] = -tone->table[4];
1502  f[0] = tone->table[3] - tone->table[0];
1503  f[2] = 1.0 - tone->table[2] - tone->table[3];
1504  f[3] = tone->table[1] + tone->table[4] - 1.0;
1505  f[4] = tone->table[0] - tone->table[1];
1506  f[5] = tone->table[2];
1507  for (i = 0; i < 2; i++) {
1508  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1509  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1510  }
1511  for (i = 0; i < 4; i++) {
1512  tone->complex[i].re += c.re * f[i+2];
1513  tone->complex[i].im += c.im * f[i+2];
1514  }
1515  }
1516 
1517  /* copy the tone if it has not yet died out */
1518  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1519  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1520  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1521  }
1522 }
1523 
1524 
1525 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1526 {
1527  int i, j, ch;
1528  const double iscale = 0.25 * M_PI;
1529 
1530  for (ch = 0; ch < q->channels; ch++) {
1531  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1532  }
1533 
1534 
1535  /* apply FFT tones with duration 4 (1 FFT period) */
1536  if (q->fft_coefs_min_index[4] >= 0)
1537  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1538  float level;
1539  QDM2Complex c;
1540 
1541  if (q->fft_coefs[i].sub_packet != sub_packet)
1542  break;
1543 
1544  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1545  level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63];
1546 
1547  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1548  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1549  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1550  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1551  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1552  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1553  }
1554 
1555  /* generate existing FFT tones */
1556  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1558  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1559  }
1560 
1561  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1562  for (i = 0; i < 4; i++)
1563  if (q->fft_coefs_min_index[i] >= 0) {
1564  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1565  int offset, four_i;
1566  FFTTone tone;
1567 
1568  if (q->fft_coefs[j].sub_packet != sub_packet)
1569  break;
1570 
1571  four_i = (4 - i);
1572  offset = q->fft_coefs[j].offset >> four_i;
1573  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1574 
1575  if (offset < q->frequency_range) {
1576  if (offset < 2)
1577  tone.cutoff = offset;
1578  else
1579  tone.cutoff = (offset >= 60) ? 3 : 2;
1580 
1581  tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
1582  tone.complex = &q->fft.complex[ch][offset];
1583  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1584  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1585  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1586  tone.duration = i;
1587  tone.time_index = 0;
1588 
1589  qdm2_fft_generate_tone(q, &tone);
1590  }
1591  }
1592  q->fft_coefs_min_index[i] = j;
1593  }
1594 }
1595 
1596 
1597 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1598 {
1599  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1600  int i;
1601  q->fft.complex[channel][0].re *= 2.0f;
1602  q->fft.complex[channel][0].im = 0.0f;
1603  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1604  /* add samples to output buffer */
1605  for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1606  q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1607 }
1608 
1609 
1615 {
1616  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1617 
1618  /* copy sb_samples */
1619  sb_used = QDM2_SB_USED(q->sub_sampling);
1620 
1621  for (ch = 0; ch < q->channels; ch++)
1622  for (i = 0; i < 8; i++)
1623  for (k=sb_used; k < SBLIMIT; k++)
1624  q->sb_samples[ch][(8 * index) + i][k] = 0;
1625 
1626  for (ch = 0; ch < q->nb_channels; ch++) {
1627  float *samples_ptr = q->samples + ch;
1628 
1629  for (i = 0; i < 8; i++) {
1631  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1632  ff_mpa_synth_window_float, &dither_state,
1633  samples_ptr, q->nb_channels,
1634  q->sb_samples[ch][(8 * index) + i]);
1635  samples_ptr += 32 * q->nb_channels;
1636  }
1637  }
1638 
1639  /* add samples to output buffer */
1640  sub_sampling = (4 >> q->sub_sampling);
1641 
1642  for (ch = 0; ch < q->channels; ch++)
1643  for (i = 0; i < q->frame_size; i++)
1644  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1645 }
1646 
1647 
1653 static av_cold void qdm2_init(QDM2Context *q) {
1654  static int initialized = 0;
1655 
1656  if (initialized != 0)
1657  return;
1658  initialized = 1;
1659 
1660  qdm2_init_vlc();
1663  rnd_table_init();
1665 
1666  av_log(NULL, AV_LOG_DEBUG, "init done\n");
1667 }
1668 
1669 
1670 #if 0
1671 static void dump_context(QDM2Context *q)
1672 {
1673  int i;
1674 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1675  PRINT("compressed_data",q->compressed_data);
1676  PRINT("compressed_size",q->compressed_size);
1677  PRINT("frame_size",q->frame_size);
1678  PRINT("checksum_size",q->checksum_size);
1679  PRINT("channels",q->channels);
1680  PRINT("nb_channels",q->nb_channels);
1681  PRINT("fft_frame_size",q->fft_frame_size);
1682  PRINT("fft_size",q->fft_size);
1683  PRINT("sub_sampling",q->sub_sampling);
1684  PRINT("fft_order",q->fft_order);
1685  PRINT("group_order",q->group_order);
1686  PRINT("group_size",q->group_size);
1687  PRINT("sub_packet",q->sub_packet);
1688  PRINT("frequency_range",q->frequency_range);
1689  PRINT("has_errors",q->has_errors);
1690  PRINT("fft_tone_end",q->fft_tone_end);
1691  PRINT("fft_tone_start",q->fft_tone_start);
1692  PRINT("fft_coefs_index",q->fft_coefs_index);
1693  PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1694  PRINT("cm_table_select",q->cm_table_select);
1695  PRINT("noise_idx",q->noise_idx);
1696 
1697  for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1698  {
1699  FFTTone *t = &q->fft_tones[i];
1700 
1701  av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1702  av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1703 // PRINT(" level", t->level);
1704  PRINT(" phase", t->phase);
1705  PRINT(" phase_shift", t->phase_shift);
1706  PRINT(" duration", t->duration);
1707  PRINT(" samples_im", t->samples_im);
1708  PRINT(" samples_re", t->samples_re);
1709  PRINT(" table", t->table);
1710  }
1711 
1712 }
1713 #endif
1714 
1715 
1720 {
1721  QDM2Context *s = avctx->priv_data;
1722  uint8_t *extradata;
1723  int extradata_size;
1724  int tmp_val, tmp, size;
1725 
1726  /* extradata parsing
1727 
1728  Structure:
1729  wave {
1730  frma (QDM2)
1731  QDCA
1732  QDCP
1733  }
1734 
1735  32 size (including this field)
1736  32 tag (=frma)
1737  32 type (=QDM2 or QDMC)
1738 
1739  32 size (including this field, in bytes)
1740  32 tag (=QDCA) // maybe mandatory parameters
1741  32 unknown (=1)
1742  32 channels (=2)
1743  32 samplerate (=44100)
1744  32 bitrate (=96000)
1745  32 block size (=4096)
1746  32 frame size (=256) (for one channel)
1747  32 packet size (=1300)
1748 
1749  32 size (including this field, in bytes)
1750  32 tag (=QDCP) // maybe some tuneable parameters
1751  32 float1 (=1.0)
1752  32 zero ?
1753  32 float2 (=1.0)
1754  32 float3 (=1.0)
1755  32 unknown (27)
1756  32 unknown (8)
1757  32 zero ?
1758  */
1759 
1760  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1761  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1762  return -1;
1763  }
1764 
1765  extradata = avctx->extradata;
1766  extradata_size = avctx->extradata_size;
1767 
1768  while (extradata_size > 7) {
1769  if (!memcmp(extradata, "frmaQDM", 7))
1770  break;
1771  extradata++;
1772  extradata_size--;
1773  }
1774 
1775  if (extradata_size < 12) {
1776  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1777  extradata_size);
1778  return -1;
1779  }
1780 
1781  if (memcmp(extradata, "frmaQDM", 7)) {
1782  av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1783  return -1;
1784  }
1785 
1786  if (extradata[7] == 'C') {
1787 // s->is_qdmc = 1;
1788  av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1789  return -1;
1790  }
1791 
1792  extradata += 8;
1793  extradata_size -= 8;
1794 
1795  size = AV_RB32(extradata);
1796 
1797  if(size > extradata_size){
1798  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1799  extradata_size, size);
1800  return -1;
1801  }
1802 
1803  extradata += 4;
1804  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1805  if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1806  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1807  return -1;
1808  }
1809 
1810  extradata += 8;
1811 
1812  avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1813  extradata += 4;
1814  if (s->channels > MPA_MAX_CHANNELS)
1815  return AVERROR_INVALIDDATA;
1816 
1817  avctx->sample_rate = AV_RB32(extradata);
1818  extradata += 4;
1819 
1820  avctx->bit_rate = AV_RB32(extradata);
1821  extradata += 4;
1822 
1823  s->group_size = AV_RB32(extradata);
1824  extradata += 4;
1825 
1826  s->fft_size = AV_RB32(extradata);
1827  extradata += 4;
1828 
1829  s->checksum_size = AV_RB32(extradata);
1830  if (s->checksum_size >= 1U << 28) {
1831  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1832  return AVERROR_INVALIDDATA;
1833  }
1834 
1835  s->fft_order = av_log2(s->fft_size) + 1;
1836  s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1837 
1838  // something like max decodable tones
1839  s->group_order = av_log2(s->group_size) + 1;
1840  s->frame_size = s->group_size / 16; // 16 iterations per super block
1842  return AVERROR_INVALIDDATA;
1843 
1844  s->sub_sampling = s->fft_order - 7;
1845  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1846 
1847  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1848  case 0: tmp = 40; break;
1849  case 1: tmp = 48; break;
1850  case 2: tmp = 56; break;
1851  case 3: tmp = 72; break;
1852  case 4: tmp = 80; break;
1853  case 5: tmp = 100;break;
1854  default: tmp=s->sub_sampling; break;
1855  }
1856  tmp_val = 0;
1857  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1858  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1859  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1860  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1861  s->cm_table_select = tmp_val;
1862 
1863  if (s->sub_sampling == 0)
1864  tmp = 7999;
1865  else
1866  tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1867  /*
1868  0: 7999 -> 0
1869  1: 20000 -> 2
1870  2: 28000 -> 2
1871  */
1872  if (tmp < 8000)
1873  s->coeff_per_sb_select = 0;
1874  else if (tmp <= 16000)
1875  s->coeff_per_sb_select = 1;
1876  else
1877  s->coeff_per_sb_select = 2;
1878 
1879  // Fail on unknown fft order
1880  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1881  av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1882  return -1;
1883  }
1884 
1886  ff_mpadsp_init(&s->mpadsp);
1887 
1888  qdm2_init(s);
1889 
1890  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1891 
1893  avctx->coded_frame = &s->frame;
1894 
1895 // dump_context(s);
1896  return 0;
1897 }
1898 
1899 
1901 {
1902  QDM2Context *s = avctx->priv_data;
1903 
1904  ff_rdft_end(&s->rdft_ctx);
1905 
1906  return 0;
1907 }
1908 
1909 
1910 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1911 {
1912  int ch, i;
1913  const int frame_size = (q->frame_size * q->channels);
1914 
1915  /* select input buffer */
1916  q->compressed_data = in;
1918 
1919 // dump_context(q);
1920 
1921  /* copy old block, clear new block of output samples */
1922  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1923  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1924 
1925  /* decode block of QDM2 compressed data */
1926  if (q->sub_packet == 0) {
1927  q->has_errors = 0; // zero it for a new super block
1928  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1930  }
1931 
1932  /* parse subpackets */
1933  if (!q->has_errors) {
1934  if (q->sub_packet == 2)
1936 
1938  }
1939 
1940  /* sound synthesis stage 1 (FFT) */
1941  for (ch = 0; ch < q->channels; ch++) {
1942  qdm2_calculate_fft(q, ch, q->sub_packet);
1943 
1944  if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1945  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1946  return -1;
1947  }
1948  }
1949 
1950  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1951  if (!q->has_errors && q->do_synth_filter)
1953 
1954  q->sub_packet = (q->sub_packet + 1) % 16;
1955 
1956  /* clip and convert output float[] to 16bit signed samples */
1957  for (i = 0; i < frame_size; i++) {
1958  int value = (int)q->output_buffer[i];
1959 
1960  if (value > SOFTCLIP_THRESHOLD)
1961  value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1962  else if (value < -SOFTCLIP_THRESHOLD)
1963  value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1964 
1965  out[i] = value;
1966  }
1967 
1968  return 0;
1969 }
1970 
1971 
1972 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1973  int *got_frame_ptr, AVPacket *avpkt)
1974 {
1975  const uint8_t *buf = avpkt->data;
1976  int buf_size = avpkt->size;
1977  QDM2Context *s = avctx->priv_data;
1978  int16_t *out;
1979  int i, ret;
1980 
1981  if(!buf)
1982  return 0;
1983  if(buf_size < s->checksum_size)
1984  return -1;
1985 
1986  /* get output buffer */
1987  s->frame.nb_samples = 16 * s->frame_size;
1988  if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
1989  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1990  return ret;
1991  }
1992  out = (int16_t *)s->frame.data[0];
1993 
1994  for (i = 0; i < 16; i++) {
1995  if (qdm2_decode(s, buf, out) < 0)
1996  return -1;
1997  out += s->channels * s->frame_size;
1998  }
1999 
2000  *got_frame_ptr = 1;
2001  *(AVFrame *)data = s->frame;
2002 
2003  return s->checksum_size;
2004 }
2005 
2007 {
2008  .name = "qdm2",
2009  .type = AVMEDIA_TYPE_AUDIO,
2010  .id = CODEC_ID_QDM2,
2011  .priv_data_size = sizeof(QDM2Context),
2015  .capabilities = CODEC_CAP_DR1,
2016  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2017 };