qdm2.c
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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 int fix_coding_method_array(int sb, int channels,
502  sb_int8_array coding_method)
503 {
504  int j,k;
505  int ch;
506  int run, case_val;
507  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};
508 
509  for (ch = 0; ch < channels; ch++) {
510  for (j = 0; j < 64; ) {
511  if (coding_method[ch][sb][j] < 8)
512  return -1;
513  if ((coding_method[ch][sb][j] - 8) > 22) {
514  run = 1;
515  case_val = 8;
516  } else {
517  switch (switchtable[coding_method[ch][sb][j]-8]) {
518  case 0: run = 10; case_val = 10; break;
519  case 1: run = 1; case_val = 16; break;
520  case 2: run = 5; case_val = 24; break;
521  case 3: run = 3; case_val = 30; break;
522  case 4: run = 1; case_val = 30; break;
523  case 5: run = 1; case_val = 8; break;
524  default: run = 1; case_val = 8; break;
525  }
526  }
527  for (k = 0; k < run; k++)
528  if (j + k < 128)
529  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
530  if (k > 0) {
532  //not debugged, almost never used
533  memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
534  memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
535  }
536  j += run;
537  }
538  }
539  return 0;
540 }
541 
542 
550 static void fill_tone_level_array (QDM2Context *q, int flag)
551 {
552  int i, sb, ch, sb_used;
553  int tmp, tab;
554 
555  // This should never happen
556  if (q->nb_channels <= 0)
557  return;
558 
559  for (ch = 0; ch < q->nb_channels; ch++)
560  for (sb = 0; sb < 30; sb++)
561  for (i = 0; i < 8; i++) {
563  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
565  else
566  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
567  if(tmp < 0)
568  tmp += 0xff;
569  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
570  }
571 
572  sb_used = QDM2_SB_USED(q->sub_sampling);
573 
574  if ((q->superblocktype_2_3 != 0) && !flag) {
575  for (sb = 0; sb < sb_used; sb++)
576  for (ch = 0; ch < q->nb_channels; ch++)
577  for (i = 0; i < 64; i++) {
578  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
579  if (q->tone_level_idx[ch][sb][i] < 0)
580  q->tone_level[ch][sb][i] = 0;
581  else
582  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
583  }
584  } else {
585  tab = q->superblocktype_2_3 ? 0 : 1;
586  for (sb = 0; sb < sb_used; sb++) {
587  if ((sb >= 4) && (sb <= 23)) {
588  for (ch = 0; ch < q->nb_channels; ch++)
589  for (i = 0; i < 64; i++) {
590  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
591  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
592  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
593  q->tone_level_idx_hi2[ch][sb - 4];
594  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
595  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
596  q->tone_level[ch][sb][i] = 0;
597  else
598  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
599  }
600  } else {
601  if (sb > 4) {
602  for (ch = 0; ch < q->nb_channels; ch++)
603  for (i = 0; i < 64; i++) {
604  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
605  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
606  q->tone_level_idx_hi2[ch][sb - 4];
607  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
608  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
609  q->tone_level[ch][sb][i] = 0;
610  else
611  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
612  }
613  } else {
614  for (ch = 0; ch < q->nb_channels; ch++)
615  for (i = 0; i < 64; i++) {
616  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
617  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
618  q->tone_level[ch][sb][i] = 0;
619  else
620  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
621  }
622  }
623  }
624  }
625  }
626 
627  return;
628 }
629 
630 
645 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
646  sb_int8_array coding_method, int nb_channels,
647  int c, int superblocktype_2_3, int cm_table_select)
648 {
649  int ch, sb, j;
650  int tmp, acc, esp_40, comp;
651  int add1, add2, add3, add4;
652  int64_t multres;
653 
654  // This should never happen
655  if (nb_channels <= 0)
656  return;
657 
658  if (!superblocktype_2_3) {
659  /* This case is untested, no samples available */
661  for (ch = 0; ch < nb_channels; ch++)
662  for (sb = 0; sb < 30; sb++) {
663  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
664  add1 = tone_level_idx[ch][sb][j] - 10;
665  if (add1 < 0)
666  add1 = 0;
667  add2 = add3 = add4 = 0;
668  if (sb > 1) {
669  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
670  if (add2 < 0)
671  add2 = 0;
672  }
673  if (sb > 0) {
674  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
675  if (add3 < 0)
676  add3 = 0;
677  }
678  if (sb < 29) {
679  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
680  if (add4 < 0)
681  add4 = 0;
682  }
683  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
684  if (tmp < 0)
685  tmp = 0;
686  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
687  }
688  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
689  }
690  acc = 0;
691  for (ch = 0; ch < nb_channels; ch++)
692  for (sb = 0; sb < 30; sb++)
693  for (j = 0; j < 64; j++)
694  acc += tone_level_idx_temp[ch][sb][j];
695 
696  multres = 0x66666667 * (acc * 10);
697  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
698  for (ch = 0; ch < nb_channels; ch++)
699  for (sb = 0; sb < 30; sb++)
700  for (j = 0; j < 64; j++) {
701  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
702  if (comp < 0)
703  comp += 0xff;
704  comp /= 256; // signed shift
705  switch(sb) {
706  case 0:
707  if (comp < 30)
708  comp = 30;
709  comp += 15;
710  break;
711  case 1:
712  if (comp < 24)
713  comp = 24;
714  comp += 10;
715  break;
716  case 2:
717  case 3:
718  case 4:
719  if (comp < 16)
720  comp = 16;
721  }
722  if (comp <= 5)
723  tmp = 0;
724  else if (comp <= 10)
725  tmp = 10;
726  else if (comp <= 16)
727  tmp = 16;
728  else if (comp <= 24)
729  tmp = -1;
730  else
731  tmp = 0;
732  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
733  }
734  for (sb = 0; sb < 30; sb++)
735  fix_coding_method_array(sb, nb_channels, coding_method);
736  for (ch = 0; ch < nb_channels; ch++)
737  for (sb = 0; sb < 30; sb++)
738  for (j = 0; j < 64; j++)
739  if (sb >= 10) {
740  if (coding_method[ch][sb][j] < 10)
741  coding_method[ch][sb][j] = 10;
742  } else {
743  if (sb >= 2) {
744  if (coding_method[ch][sb][j] < 16)
745  coding_method[ch][sb][j] = 16;
746  } else {
747  if (coding_method[ch][sb][j] < 30)
748  coding_method[ch][sb][j] = 30;
749  }
750  }
751  } else { // superblocktype_2_3 != 0
752  for (ch = 0; ch < nb_channels; ch++)
753  for (sb = 0; sb < 30; sb++)
754  for (j = 0; j < 64; j++)
755  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
756  }
757 
758  return;
759 }
760 
761 
773 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
774 {
775  int sb, j, k, n, ch, run, channels;
776  int joined_stereo, zero_encoding;
777  int type34_first;
778  float type34_div = 0;
779  float type34_predictor;
780  float samples[10], sign_bits[16];
781 
782  if (length == 0) {
783  // If no data use noise
784  for (sb=sb_min; sb < sb_max; sb++)
786 
787  return;
788  }
789 
790  for (sb = sb_min; sb < sb_max; sb++) {
791  channels = q->nb_channels;
792 
793  if (q->nb_channels <= 1 || sb < 12)
794  joined_stereo = 0;
795  else if (sb >= 24)
796  joined_stereo = 1;
797  else
798  joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
799 
800  if (joined_stereo) {
801  if (BITS_LEFT(length,gb) >= 16)
802  for (j = 0; j < 16; j++)
803  sign_bits[j] = get_bits1 (gb);
804 
805  for (j = 0; j < 64; j++)
806  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
807  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
808 
810  q->coding_method)) {
812  continue;
813  }
814  channels = 1;
815  }
816 
817  for (ch = 0; ch < channels; ch++) {
819  zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
820  type34_predictor = 0.0;
821  type34_first = 1;
822 
823  for (j = 0; j < 128; ) {
824  switch (q->coding_method[ch][sb][j / 2]) {
825  case 8:
826  if (BITS_LEFT(length,gb) >= 10) {
827  if (zero_encoding) {
828  for (k = 0; k < 5; k++) {
829  if ((j + 2 * k) >= 128)
830  break;
831  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
832  }
833  } else {
834  n = get_bits(gb, 8);
835  for (k = 0; k < 5; k++)
836  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
837  }
838  for (k = 0; k < 5; k++)
839  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
840  } else {
841  for (k = 0; k < 10; k++)
842  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
843  }
844  run = 10;
845  break;
846 
847  case 10:
848  if (BITS_LEFT(length,gb) >= 1) {
849  float f = 0.81;
850 
851  if (get_bits1(gb))
852  f = -f;
853  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
854  samples[0] = f;
855  } else {
856  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
857  }
858  run = 1;
859  break;
860 
861  case 16:
862  if (BITS_LEFT(length,gb) >= 10) {
863  if (zero_encoding) {
864  for (k = 0; k < 5; k++) {
865  if ((j + k) >= 128)
866  break;
867  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
868  }
869  } else {
870  n = get_bits (gb, 8);
871  for (k = 0; k < 5; k++)
872  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
873  }
874  } else {
875  for (k = 0; k < 5; k++)
876  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
877  }
878  run = 5;
879  break;
880 
881  case 24:
882  if (BITS_LEFT(length,gb) >= 7) {
883  n = get_bits(gb, 7);
884  for (k = 0; k < 3; k++)
885  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
886  } else {
887  for (k = 0; k < 3; k++)
888  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
889  }
890  run = 3;
891  break;
892 
893  case 30:
894  if (BITS_LEFT(length,gb) >= 4) {
895  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
896  if (index < FF_ARRAY_ELEMS(type30_dequant)) {
897  samples[0] = type30_dequant[index];
898  } else
899  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
900  } else
901  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
902 
903  run = 1;
904  break;
905 
906  case 34:
907  if (BITS_LEFT(length,gb) >= 7) {
908  if (type34_first) {
909  type34_div = (float)(1 << get_bits(gb, 2));
910  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
911  type34_predictor = samples[0];
912  type34_first = 0;
913  } else {
914  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
915  if (index < FF_ARRAY_ELEMS(type34_delta)) {
916  samples[0] = type34_delta[index] / type34_div + type34_predictor;
917  type34_predictor = samples[0];
918  } else
919  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
920  }
921  } else {
922  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
923  }
924  run = 1;
925  break;
926 
927  default:
928  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
929  run = 1;
930  break;
931  }
932 
933  if (joined_stereo) {
934  for (k = 0; k < run && j + k < 128; k++) {
935  q->sb_samples[0][j + k][sb] =
936  q->tone_level[0][sb][(j + k) / 2] * samples[k];
937  if (q->nb_channels == 2) {
938  if (sign_bits[(j + k) / 8])
939  q->sb_samples[1][j + k][sb] =
940  q->tone_level[1][sb][(j + k) / 2] * -samples[k];
941  else
942  q->sb_samples[1][j + k][sb] =
943  q->tone_level[1][sb][(j + k) / 2] * samples[k];
944  }
945  }
946  } else {
947  for (k = 0; k < run; k++)
948  if ((j + k) < 128)
949  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
950  }
951 
952  j += run;
953  } // j loop
954  } // channel loop
955  } // subband loop
956 }
957 
958 
968 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
969 {
970  int i, k, run, level, diff;
971 
972  if (BITS_LEFT(length,gb) < 16)
973  return;
974  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
975 
976  quantized_coeffs[0] = level;
977 
978  for (i = 0; i < 7; ) {
979  if (BITS_LEFT(length,gb) < 16)
980  break;
981  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
982 
983  if (BITS_LEFT(length,gb) < 16)
984  break;
985  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
986 
987  for (k = 1; k <= run; k++)
988  quantized_coeffs[i + k] = (level + ((k * diff) / run));
989 
990  level += diff;
991  i += run;
992  }
993 }
994 
995 
1006 {
1007  int sb, j, k, n, ch;
1008 
1009  for (ch = 0; ch < q->nb_channels; ch++) {
1010  init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1011 
1012  if (BITS_LEFT(length,gb) < 16) {
1013  memset(q->quantized_coeffs[ch][0], 0, 8);
1014  break;
1015  }
1016  }
1017 
1018  n = q->sub_sampling + 1;
1019 
1020  for (sb = 0; sb < n; sb++)
1021  for (ch = 0; ch < q->nb_channels; ch++)
1022  for (j = 0; j < 8; j++) {
1023  if (BITS_LEFT(length,gb) < 1)
1024  break;
1025  if (get_bits1(gb)) {
1026  for (k=0; k < 8; k++) {
1027  if (BITS_LEFT(length,gb) < 16)
1028  break;
1029  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1030  }
1031  } else {
1032  for (k=0; k < 8; k++)
1033  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1034  }
1035  }
1036 
1037  n = QDM2_SB_USED(q->sub_sampling) - 4;
1038 
1039  for (sb = 0; sb < n; sb++)
1040  for (ch = 0; ch < q->nb_channels; ch++) {
1041  if (BITS_LEFT(length,gb) < 16)
1042  break;
1043  q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1044  if (sb > 19)
1045  q->tone_level_idx_hi2[ch][sb] -= 16;
1046  else
1047  for (j = 0; j < 8; j++)
1048  q->tone_level_idx_mid[ch][sb][j] = -16;
1049  }
1050 
1051  n = QDM2_SB_USED(q->sub_sampling) - 5;
1052 
1053  for (sb = 0; sb < n; sb++)
1054  for (ch = 0; ch < q->nb_channels; ch++)
1055  for (j = 0; j < 8; j++) {
1056  if (BITS_LEFT(length,gb) < 16)
1057  break;
1058  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1059  }
1060 }
1061 
1069 {
1070  GetBitContext gb;
1071  int i, j, k, n, ch, run, level, diff;
1072 
1073  init_get_bits(&gb, node->packet->data, node->packet->size*8);
1074 
1075  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1076 
1077  for (i = 1; i < n; i++)
1078  for (ch=0; ch < q->nb_channels; ch++) {
1079  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1080  q->quantized_coeffs[ch][i][0] = level;
1081 
1082  for (j = 0; j < (8 - 1); ) {
1083  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1084  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1085 
1086  for (k = 1; k <= run; k++)
1087  q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1088 
1089  level += diff;
1090  j += run;
1091  }
1092  }
1093 
1094  for (ch = 0; ch < q->nb_channels; ch++)
1095  for (i = 0; i < 8; i++)
1096  q->quantized_coeffs[ch][0][i] = 0;
1097 }
1098 
1099 
1107 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1108 {
1109  GetBitContext gb;
1110 
1111  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1112 
1113  if (length != 0) {
1114  init_tone_level_dequantization(q, &gb, length);
1115  fill_tone_level_array(q, 1);
1116  } else {
1117  fill_tone_level_array(q, 0);
1118  }
1119 }
1120 
1121 
1129 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1130 {
1131  GetBitContext gb;
1132 
1133  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1134  if (length >= 32) {
1135  int c = get_bits (&gb, 13);
1136 
1137  if (c > 3)
1140  }
1141 
1142  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1143 }
1144 
1145 
1153 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1154 {
1155  GetBitContext gb;
1156 
1157  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1158  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1159 }
1160 
1161 /*
1162  * Process new subpackets for synthesis filter
1163  *
1164  * @param q context
1165  * @param list list with synthesis filter packets (list D)
1166  */
1168 {
1169  QDM2SubPNode *nodes[4];
1170 
1171  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1172  if (nodes[0] != NULL)
1173  process_subpacket_9(q, nodes[0]);
1174 
1175  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1176  if (nodes[1] != NULL)
1177  process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1178  else
1179  process_subpacket_10(q, NULL, 0);
1180 
1181  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1182  if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1183  process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1184  else
1185  process_subpacket_11(q, NULL, 0);
1186 
1187  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1188  if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1189  process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1190  else
1191  process_subpacket_12(q, NULL, 0);
1192 }
1193 
1194 
1195 /*
1196  * Decode superblock, fill packet lists.
1197  *
1198  * @param q context
1199  */
1201 {
1202  GetBitContext gb;
1203  QDM2SubPacket header, *packet;
1204  int i, packet_bytes, sub_packet_size, sub_packets_D;
1205  unsigned int next_index = 0;
1206 
1207  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1208  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1209  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1210 
1211  q->sub_packets_B = 0;
1212  sub_packets_D = 0;
1213 
1214  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1215 
1217  qdm2_decode_sub_packet_header(&gb, &header);
1218 
1219  if (header.type < 2 || header.type >= 8) {
1220  q->has_errors = 1;
1221  av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1222  return;
1223  }
1224 
1225  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1226  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1227 
1228  init_get_bits(&gb, header.data, header.size*8);
1229 
1230  if (header.type == 2 || header.type == 4 || header.type == 5) {
1231  int csum = 257 * get_bits(&gb, 8);
1232  csum += 2 * get_bits(&gb, 8);
1233 
1234  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1235 
1236  if (csum != 0) {
1237  q->has_errors = 1;
1238  av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1239  return;
1240  }
1241  }
1242 
1243  q->sub_packet_list_B[0].packet = NULL;
1244  q->sub_packet_list_D[0].packet = NULL;
1245 
1246  for (i = 0; i < 6; i++)
1247  if (--q->fft_level_exp[i] < 0)
1248  q->fft_level_exp[i] = 0;
1249 
1250  for (i = 0; packet_bytes > 0; i++) {
1251  int j;
1252 
1253  q->sub_packet_list_A[i].next = NULL;
1254 
1255  if (i > 0) {
1256  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1257 
1258  /* seek to next block */
1259  init_get_bits(&gb, header.data, header.size*8);
1260  skip_bits(&gb, next_index*8);
1261 
1262  if (next_index >= header.size)
1263  break;
1264  }
1265 
1266  /* decode subpacket */
1267  packet = &q->sub_packets[i];
1268  qdm2_decode_sub_packet_header(&gb, packet);
1269  next_index = packet->size + get_bits_count(&gb) / 8;
1270  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1271 
1272  if (packet->type == 0)
1273  break;
1274 
1275  if (sub_packet_size > packet_bytes) {
1276  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1277  break;
1278  packet->size += packet_bytes - sub_packet_size;
1279  }
1280 
1281  packet_bytes -= sub_packet_size;
1282 
1283  /* add subpacket to 'all subpackets' list */
1284  q->sub_packet_list_A[i].packet = packet;
1285 
1286  /* add subpacket to related list */
1287  if (packet->type == 8) {
1288  SAMPLES_NEEDED_2("packet type 8");
1289  return;
1290  } else if (packet->type >= 9 && packet->type <= 12) {
1291  /* packets for MPEG Audio like Synthesis Filter */
1292  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1293  } else if (packet->type == 13) {
1294  for (j = 0; j < 6; j++)
1295  q->fft_level_exp[j] = get_bits(&gb, 6);
1296  } else if (packet->type == 14) {
1297  for (j = 0; j < 6; j++)
1298  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1299  } else if (packet->type == 15) {
1300  SAMPLES_NEEDED_2("packet type 15")
1301  return;
1302  } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1303  /* packets for FFT */
1305  }
1306  } // Packet bytes loop
1307 
1308 /* **************************************************************** */
1309  if (q->sub_packet_list_D[0].packet != NULL) {
1311  q->do_synth_filter = 1;
1312  } else if (q->do_synth_filter) {
1313  process_subpacket_10(q, NULL, 0);
1314  process_subpacket_11(q, NULL, 0);
1315  process_subpacket_12(q, NULL, 0);
1316  }
1317 /* **************************************************************** */
1318 }
1319 
1320 
1321 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1322  int offset, int duration, int channel,
1323  int exp, int phase)
1324 {
1325  if (q->fft_coefs_min_index[duration] < 0)
1327 
1328  q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1329  q->fft_coefs[q->fft_coefs_index].channel = channel;
1330  q->fft_coefs[q->fft_coefs_index].offset = offset;
1331  q->fft_coefs[q->fft_coefs_index].exp = exp;
1332  q->fft_coefs[q->fft_coefs_index].phase = phase;
1333  q->fft_coefs_index++;
1334 }
1335 
1336 
1338 {
1339  int channel, stereo, phase, exp;
1340  int local_int_4, local_int_8, stereo_phase, local_int_10;
1341  int local_int_14, stereo_exp, local_int_20, local_int_28;
1342  int n, offset;
1343 
1344  local_int_4 = 0;
1345  local_int_28 = 0;
1346  local_int_20 = 2;
1347  local_int_8 = (4 - duration);
1348  local_int_10 = 1 << (q->group_order - duration - 1);
1349  offset = 1;
1350 
1351  while (1) {
1352  if (q->superblocktype_2_3) {
1353  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1354  offset = 1;
1355  if (n == 0) {
1356  local_int_4 += local_int_10;
1357  local_int_28 += (1 << local_int_8);
1358  } else {
1359  local_int_4 += 8*local_int_10;
1360  local_int_28 += (8 << local_int_8);
1361  }
1362  }
1363  offset += (n - 2);
1364  } else {
1365  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1366  while (offset >= (local_int_10 - 1)) {
1367  offset += (1 - (local_int_10 - 1));
1368  local_int_4 += local_int_10;
1369  local_int_28 += (1 << local_int_8);
1370  }
1371  }
1372 
1373  if (local_int_4 >= q->group_size)
1374  return;
1375 
1376  local_int_14 = (offset >> local_int_8);
1377  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1378  return;
1379 
1380  if (q->nb_channels > 1) {
1381  channel = get_bits1(gb);
1382  stereo = get_bits1(gb);
1383  } else {
1384  channel = 0;
1385  stereo = 0;
1386  }
1387 
1388  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1389  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1390  exp = (exp < 0) ? 0 : exp;
1391 
1392  phase = get_bits(gb, 3);
1393  stereo_exp = 0;
1394  stereo_phase = 0;
1395 
1396  if (stereo) {
1397  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1398  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1399  if (stereo_phase < 0)
1400  stereo_phase += 8;
1401  }
1402 
1403  if (q->frequency_range > (local_int_14 + 1)) {
1404  int sub_packet = (local_int_20 + local_int_28);
1405 
1406  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1407  if (stereo)
1408  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1409  }
1410 
1411  offset++;
1412  }
1413 }
1414 
1415 
1417 {
1418  int i, j, min, max, value, type, unknown_flag;
1419  GetBitContext gb;
1420 
1421  if (q->sub_packet_list_B[0].packet == NULL)
1422  return;
1423 
1424  /* reset minimum indexes for FFT coefficients */
1425  q->fft_coefs_index = 0;
1426  for (i=0; i < 5; i++)
1427  q->fft_coefs_min_index[i] = -1;
1428 
1429  /* process subpackets ordered by type, largest type first */
1430  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1431  QDM2SubPacket *packet= NULL;
1432 
1433  /* find subpacket with largest type less than max */
1434  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1435  value = q->sub_packet_list_B[j].packet->type;
1436  if (value > min && value < max) {
1437  min = value;
1438  packet = q->sub_packet_list_B[j].packet;
1439  }
1440  }
1441 
1442  max = min;
1443 
1444  /* check for errors (?) */
1445  if (!packet)
1446  return;
1447 
1448  if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1449  return;
1450 
1451  /* decode FFT tones */
1452  init_get_bits (&gb, packet->data, packet->size*8);
1453 
1454  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1455  unknown_flag = 1;
1456  else
1457  unknown_flag = 0;
1458 
1459  type = packet->type;
1460 
1461  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1462  int duration = q->sub_sampling + 5 - (type & 15);
1463 
1464  if (duration >= 0 && duration < 4)
1465  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1466  } else if (type == 31) {
1467  for (j=0; j < 4; j++)
1468  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1469  } else if (type == 46) {
1470  for (j=0; j < 6; j++)
1471  q->fft_level_exp[j] = get_bits(&gb, 6);
1472  for (j=0; j < 4; j++)
1473  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1474  }
1475  } // Loop on B packets
1476 
1477  /* calculate maximum indexes for FFT coefficients */
1478  for (i = 0, j = -1; i < 5; i++)
1479  if (q->fft_coefs_min_index[i] >= 0) {
1480  if (j >= 0)
1482  j = i;
1483  }
1484  if (j >= 0)
1486 }
1487 
1488 
1490 {
1491  float level, f[6];
1492  int i;
1493  QDM2Complex c;
1494  const double iscale = 2.0*M_PI / 512.0;
1495 
1496  tone->phase += tone->phase_shift;
1497 
1498  /* calculate current level (maximum amplitude) of tone */
1499  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1500  c.im = level * sin(tone->phase*iscale);
1501  c.re = level * cos(tone->phase*iscale);
1502 
1503  /* generate FFT coefficients for tone */
1504  if (tone->duration >= 3 || tone->cutoff >= 3) {
1505  tone->complex[0].im += c.im;
1506  tone->complex[0].re += c.re;
1507  tone->complex[1].im -= c.im;
1508  tone->complex[1].re -= c.re;
1509  } else {
1510  f[1] = -tone->table[4];
1511  f[0] = tone->table[3] - tone->table[0];
1512  f[2] = 1.0 - tone->table[2] - tone->table[3];
1513  f[3] = tone->table[1] + tone->table[4] - 1.0;
1514  f[4] = tone->table[0] - tone->table[1];
1515  f[5] = tone->table[2];
1516  for (i = 0; i < 2; i++) {
1517  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1518  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1519  }
1520  for (i = 0; i < 4; i++) {
1521  tone->complex[i].re += c.re * f[i+2];
1522  tone->complex[i].im += c.im * f[i+2];
1523  }
1524  }
1525 
1526  /* copy the tone if it has not yet died out */
1527  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1528  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1529  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1530  }
1531 }
1532 
1533 
1534 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1535 {
1536  int i, j, ch;
1537  const double iscale = 0.25 * M_PI;
1538 
1539  for (ch = 0; ch < q->channels; ch++) {
1540  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1541  }
1542 
1543 
1544  /* apply FFT tones with duration 4 (1 FFT period) */
1545  if (q->fft_coefs_min_index[4] >= 0)
1546  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1547  float level;
1548  QDM2Complex c;
1549 
1550  if (q->fft_coefs[i].sub_packet != sub_packet)
1551  break;
1552 
1553  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1554  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];
1555 
1556  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1557  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1558  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1559  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1560  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1561  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1562  }
1563 
1564  /* generate existing FFT tones */
1565  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1567  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1568  }
1569 
1570  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1571  for (i = 0; i < 4; i++)
1572  if (q->fft_coefs_min_index[i] >= 0) {
1573  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1574  int offset, four_i;
1575  FFTTone tone;
1576 
1577  if (q->fft_coefs[j].sub_packet != sub_packet)
1578  break;
1579 
1580  four_i = (4 - i);
1581  offset = q->fft_coefs[j].offset >> four_i;
1582  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1583 
1584  if (offset < q->frequency_range) {
1585  if (offset < 2)
1586  tone.cutoff = offset;
1587  else
1588  tone.cutoff = (offset >= 60) ? 3 : 2;
1589 
1590  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];
1591  tone.complex = &q->fft.complex[ch][offset];
1592  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1593  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1594  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1595  tone.duration = i;
1596  tone.time_index = 0;
1597 
1598  qdm2_fft_generate_tone(q, &tone);
1599  }
1600  }
1601  q->fft_coefs_min_index[i] = j;
1602  }
1603 }
1604 
1605 
1606 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1607 {
1608  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1609  int i;
1610  q->fft.complex[channel][0].re *= 2.0f;
1611  q->fft.complex[channel][0].im = 0.0f;
1612  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1613  /* add samples to output buffer */
1614  for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1615  q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1616 }
1617 
1618 
1624 {
1625  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1626 
1627  /* copy sb_samples */
1628  sb_used = QDM2_SB_USED(q->sub_sampling);
1629 
1630  for (ch = 0; ch < q->channels; ch++)
1631  for (i = 0; i < 8; i++)
1632  for (k=sb_used; k < SBLIMIT; k++)
1633  q->sb_samples[ch][(8 * index) + i][k] = 0;
1634 
1635  for (ch = 0; ch < q->nb_channels; ch++) {
1636  float *samples_ptr = q->samples + ch;
1637 
1638  for (i = 0; i < 8; i++) {
1640  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1641  ff_mpa_synth_window_float, &dither_state,
1642  samples_ptr, q->nb_channels,
1643  q->sb_samples[ch][(8 * index) + i]);
1644  samples_ptr += 32 * q->nb_channels;
1645  }
1646  }
1647 
1648  /* add samples to output buffer */
1649  sub_sampling = (4 >> q->sub_sampling);
1650 
1651  for (ch = 0; ch < q->channels; ch++)
1652  for (i = 0; i < q->frame_size; i++)
1653  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1654 }
1655 
1656 
1662 static av_cold void qdm2_init(QDM2Context *q) {
1663  static int initialized = 0;
1664 
1665  if (initialized != 0)
1666  return;
1667  initialized = 1;
1668 
1669  qdm2_init_vlc();
1672  rnd_table_init();
1674 
1675  av_log(NULL, AV_LOG_DEBUG, "init done\n");
1676 }
1677 
1678 
1679 #if 0
1680 static void dump_context(QDM2Context *q)
1681 {
1682  int i;
1683 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1684  PRINT("compressed_data",q->compressed_data);
1685  PRINT("compressed_size",q->compressed_size);
1686  PRINT("frame_size",q->frame_size);
1687  PRINT("checksum_size",q->checksum_size);
1688  PRINT("channels",q->channels);
1689  PRINT("nb_channels",q->nb_channels);
1690  PRINT("fft_frame_size",q->fft_frame_size);
1691  PRINT("fft_size",q->fft_size);
1692  PRINT("sub_sampling",q->sub_sampling);
1693  PRINT("fft_order",q->fft_order);
1694  PRINT("group_order",q->group_order);
1695  PRINT("group_size",q->group_size);
1696  PRINT("sub_packet",q->sub_packet);
1697  PRINT("frequency_range",q->frequency_range);
1698  PRINT("has_errors",q->has_errors);
1699  PRINT("fft_tone_end",q->fft_tone_end);
1700  PRINT("fft_tone_start",q->fft_tone_start);
1701  PRINT("fft_coefs_index",q->fft_coefs_index);
1702  PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1703  PRINT("cm_table_select",q->cm_table_select);
1704  PRINT("noise_idx",q->noise_idx);
1705 
1706  for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1707  {
1708  FFTTone *t = &q->fft_tones[i];
1709 
1710  av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1711  av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1712 // PRINT(" level", t->level);
1713  PRINT(" phase", t->phase);
1714  PRINT(" phase_shift", t->phase_shift);
1715  PRINT(" duration", t->duration);
1716  PRINT(" samples_im", t->samples_im);
1717  PRINT(" samples_re", t->samples_re);
1718  PRINT(" table", t->table);
1719  }
1720 
1721 }
1722 #endif
1723 
1724 
1729 {
1730  QDM2Context *s = avctx->priv_data;
1731  uint8_t *extradata;
1732  int extradata_size;
1733  int tmp_val, tmp, size;
1734 
1735  /* extradata parsing
1736 
1737  Structure:
1738  wave {
1739  frma (QDM2)
1740  QDCA
1741  QDCP
1742  }
1743 
1744  32 size (including this field)
1745  32 tag (=frma)
1746  32 type (=QDM2 or QDMC)
1747 
1748  32 size (including this field, in bytes)
1749  32 tag (=QDCA) // maybe mandatory parameters
1750  32 unknown (=1)
1751  32 channels (=2)
1752  32 samplerate (=44100)
1753  32 bitrate (=96000)
1754  32 block size (=4096)
1755  32 frame size (=256) (for one channel)
1756  32 packet size (=1300)
1757 
1758  32 size (including this field, in bytes)
1759  32 tag (=QDCP) // maybe some tuneable parameters
1760  32 float1 (=1.0)
1761  32 zero ?
1762  32 float2 (=1.0)
1763  32 float3 (=1.0)
1764  32 unknown (27)
1765  32 unknown (8)
1766  32 zero ?
1767  */
1768 
1769  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1770  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1771  return -1;
1772  }
1773 
1774  extradata = avctx->extradata;
1775  extradata_size = avctx->extradata_size;
1776 
1777  while (extradata_size > 7) {
1778  if (!memcmp(extradata, "frmaQDM", 7))
1779  break;
1780  extradata++;
1781  extradata_size--;
1782  }
1783 
1784  if (extradata_size < 12) {
1785  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1786  extradata_size);
1787  return -1;
1788  }
1789 
1790  if (memcmp(extradata, "frmaQDM", 7)) {
1791  av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1792  return -1;
1793  }
1794 
1795  if (extradata[7] == 'C') {
1796 // s->is_qdmc = 1;
1797  av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1798  return -1;
1799  }
1800 
1801  extradata += 8;
1802  extradata_size -= 8;
1803 
1804  size = AV_RB32(extradata);
1805 
1806  if(size > extradata_size){
1807  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1808  extradata_size, size);
1809  return -1;
1810  }
1811 
1812  extradata += 4;
1813  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1814  if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1815  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1816  return -1;
1817  }
1818 
1819  extradata += 8;
1820 
1821  avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1822  extradata += 4;
1823  if (s->channels > MPA_MAX_CHANNELS)
1824  return AVERROR_INVALIDDATA;
1825 
1826  avctx->sample_rate = AV_RB32(extradata);
1827  extradata += 4;
1828 
1829  avctx->bit_rate = AV_RB32(extradata);
1830  extradata += 4;
1831 
1832  s->group_size = AV_RB32(extradata);
1833  extradata += 4;
1834 
1835  s->fft_size = AV_RB32(extradata);
1836  extradata += 4;
1837 
1838  s->checksum_size = AV_RB32(extradata);
1839  if (s->checksum_size >= 1U << 28) {
1840  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1841  return AVERROR_INVALIDDATA;
1842  }
1843 
1844  s->fft_order = av_log2(s->fft_size) + 1;
1845  s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1846 
1847  // something like max decodable tones
1848  s->group_order = av_log2(s->group_size) + 1;
1849  s->frame_size = s->group_size / 16; // 16 iterations per super block
1851  return AVERROR_INVALIDDATA;
1852 
1853  s->sub_sampling = s->fft_order - 7;
1854  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1855 
1856  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1857  case 0: tmp = 40; break;
1858  case 1: tmp = 48; break;
1859  case 2: tmp = 56; break;
1860  case 3: tmp = 72; break;
1861  case 4: tmp = 80; break;
1862  case 5: tmp = 100;break;
1863  default: tmp=s->sub_sampling; break;
1864  }
1865  tmp_val = 0;
1866  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1867  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1868  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1869  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1870  s->cm_table_select = tmp_val;
1871 
1872  if (s->sub_sampling == 0)
1873  tmp = 7999;
1874  else
1875  tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1876  /*
1877  0: 7999 -> 0
1878  1: 20000 -> 2
1879  2: 28000 -> 2
1880  */
1881  if (tmp < 8000)
1882  s->coeff_per_sb_select = 0;
1883  else if (tmp <= 16000)
1884  s->coeff_per_sb_select = 1;
1885  else
1886  s->coeff_per_sb_select = 2;
1887 
1888  // Fail on unknown fft order
1889  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1890  av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1891  return -1;
1892  }
1893  if (s->fft_size != (1 << (s->fft_order - 1))) {
1894  av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1895  return AVERROR_INVALIDDATA;
1896  }
1897 
1899  ff_mpadsp_init(&s->mpadsp);
1900 
1901  qdm2_init(s);
1902 
1903  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1904 
1906  avctx->coded_frame = &s->frame;
1907 
1908 // dump_context(s);
1909  return 0;
1910 }
1911 
1912 
1914 {
1915  QDM2Context *s = avctx->priv_data;
1916 
1917  ff_rdft_end(&s->rdft_ctx);
1918 
1919  return 0;
1920 }
1921 
1922 
1923 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1924 {
1925  int ch, i;
1926  const int frame_size = (q->frame_size * q->channels);
1927 
1928  /* select input buffer */
1929  q->compressed_data = in;
1931 
1932 // dump_context(q);
1933 
1934  /* copy old block, clear new block of output samples */
1935  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1936  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1937 
1938  /* decode block of QDM2 compressed data */
1939  if (q->sub_packet == 0) {
1940  q->has_errors = 0; // zero it for a new super block
1941  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1943  }
1944 
1945  /* parse subpackets */
1946  if (!q->has_errors) {
1947  if (q->sub_packet == 2)
1949 
1951  }
1952 
1953  /* sound synthesis stage 1 (FFT) */
1954  for (ch = 0; ch < q->channels; ch++) {
1955  qdm2_calculate_fft(q, ch, q->sub_packet);
1956 
1957  if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1958  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1959  return -1;
1960  }
1961  }
1962 
1963  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1964  if (!q->has_errors && q->do_synth_filter)
1966 
1967  q->sub_packet = (q->sub_packet + 1) % 16;
1968 
1969  /* clip and convert output float[] to 16bit signed samples */
1970  for (i = 0; i < frame_size; i++) {
1971  int value = (int)q->output_buffer[i];
1972 
1973  if (value > SOFTCLIP_THRESHOLD)
1974  value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1975  else if (value < -SOFTCLIP_THRESHOLD)
1976  value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1977 
1978  out[i] = value;
1979  }
1980 
1981  return 0;
1982 }
1983 
1984 
1985 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1986  int *got_frame_ptr, AVPacket *avpkt)
1987 {
1988  const uint8_t *buf = avpkt->data;
1989  int buf_size = avpkt->size;
1990  QDM2Context *s = avctx->priv_data;
1991  int16_t *out;
1992  int i, ret;
1993 
1994  if(!buf)
1995  return 0;
1996  if(buf_size < s->checksum_size)
1997  return -1;
1998 
1999  /* get output buffer */
2000  s->frame.nb_samples = 16 * s->frame_size;
2001  if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
2002  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
2003  return ret;
2004  }
2005  out = (int16_t *)s->frame.data[0];
2006 
2007  for (i = 0; i < 16; i++) {
2008  if (qdm2_decode(s, buf, out) < 0)
2009  return -1;
2010  out += s->channels * s->frame_size;
2011  }
2012 
2013  *got_frame_ptr = 1;
2014  *(AVFrame *)data = s->frame;
2015 
2016  return s->checksum_size;
2017 }
2018 
2020 {
2021  .name = "qdm2",
2022  .type = AVMEDIA_TYPE_AUDIO,
2023  .id = CODEC_ID_QDM2,
2024  .priv_data_size = sizeof(QDM2Context),
2028  .capabilities = CODEC_CAP_DR1,
2029  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2030 };