FFmpeg  1.2.4
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 FFmpeg.
9  *
10  * FFmpeg 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  * FFmpeg 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 FFmpeg; 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 "libavutil/channel_layout.h"
40 #include "avcodec.h"
41 #include "get_bits.h"
42 #include "internal.h"
43 #include "rdft.h"
44 #include "mpegaudiodsp.h"
45 #include "mpegaudio.h"
46 
47 #include "qdm2data.h"
48 #include "qdm2_tablegen.h"
49 
50 #undef NDEBUG
51 #include <assert.h>
52 
53 
54 #define QDM2_LIST_ADD(list, size, packet) \
55 do { \
56  if (size > 0) { \
57  list[size - 1].next = &list[size]; \
58  } \
59  list[size].packet = packet; \
60  list[size].next = NULL; \
61  size++; \
62 } while(0)
63 
64 // Result is 8, 16 or 30
65 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
66 
67 #define FIX_NOISE_IDX(noise_idx) \
68  if ((noise_idx) >= 3840) \
69  (noise_idx) -= 3840; \
70 
71 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
72 
73 #define SAMPLES_NEEDED \
74  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
75 
76 #define SAMPLES_NEEDED_2(why) \
77  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
78 
79 #define QDM2_MAX_FRAME_SIZE 512
80 
81 typedef int8_t sb_int8_array[2][30][64];
82 
86 typedef struct {
87  int type;
88  unsigned int size;
89  const uint8_t *data;
91 
95 typedef struct QDM2SubPNode {
97  struct QDM2SubPNode *next;
98 } QDM2SubPNode;
99 
100 typedef struct {
101  float re;
102  float im;
103 } QDM2Complex;
104 
105 typedef struct {
106  float level;
108  const float *table;
109  int phase;
111  int duration;
112  short time_index;
113  short cutoff;
114 } FFTTone;
115 
116 typedef struct {
117  int16_t sub_packet;
119  int16_t offset;
120  int16_t exp;
123 
124 typedef struct {
126 } QDM2FFT;
127 
131 typedef struct {
134  int channels;
136  int fft_size;
138 
141  int fft_order;
147 
149  QDM2SubPacket sub_packets[16];
150  QDM2SubPNode sub_packet_list_A[16];
151  QDM2SubPNode sub_packet_list_B[16];
153  QDM2SubPNode sub_packet_list_C[16];
154  QDM2SubPNode sub_packet_list_D[16];
155 
157  FFTTone fft_tones[1000];
160  FFTCoefficient fft_coefs[1000];
162  int fft_coefs_min_index[5];
163  int fft_coefs_max_index[5];
164  int fft_level_exp[6];
167 
171  float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
172 
175  DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
176  int synth_buf_offset[MPA_MAX_CHANNELS];
177  DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
179 
181  float tone_level[MPA_MAX_CHANNELS][30][64];
182  int8_t coding_method[MPA_MAX_CHANNELS][30][64];
183  int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
184  int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
185  int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
186  int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
187  int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
188  int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
189  int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
190 
191  // Flags
195 
197  int noise_idx;
198 } QDM2Context;
199 
200 
214 
215 static const uint16_t qdm2_vlc_offs[] = {
216  0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
217 };
218 
219 static av_cold void qdm2_init_vlc(void)
220 {
221  static int vlcs_initialized = 0;
222  static VLC_TYPE qdm2_table[3838][2];
223 
224  if (!vlcs_initialized) {
225 
226  vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
227  vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
228  init_vlc (&vlc_tab_level, 8, 24,
231 
232  vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
233  vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
234  init_vlc (&vlc_tab_diff, 8, 37,
235  vlc_tab_diff_huffbits, 1, 1,
237 
238  vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
239  vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
240  init_vlc (&vlc_tab_run, 5, 6,
241  vlc_tab_run_huffbits, 1, 1,
243 
244  fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
245  fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
246  init_vlc (&fft_level_exp_alt_vlc, 8, 28,
249 
250 
251  fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
252  fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
253  init_vlc (&fft_level_exp_vlc, 8, 20,
256 
257  fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
258  fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
259  init_vlc (&fft_stereo_exp_vlc, 6, 7,
262 
263  fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
264  fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
265  init_vlc (&fft_stereo_phase_vlc, 6, 9,
268 
269  vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
270  vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
271  init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
274 
275  vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
276  vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
277  init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
280 
281  vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
282  vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
283  init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
286 
287  vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
288  vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
289  init_vlc (&vlc_tab_type30, 6, 9,
292 
293  vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
294  vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
295  init_vlc (&vlc_tab_type34, 5, 10,
298 
299  vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
300  vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
301  init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
304 
305  vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
306  vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
307  init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
310 
311  vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
312  vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
313  init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
316 
317  vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
318  vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
319  init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
322 
323  vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
324  vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
325  init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
328 
329  vlcs_initialized=1;
330  }
331 }
332 
333 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
334 {
335  int value;
336 
337  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
338 
339  /* stage-2, 3 bits exponent escape sequence */
340  if (value-- == 0)
341  value = get_bits (gb, get_bits (gb, 3) + 1);
342 
343  /* stage-3, optional */
344  if (flag) {
345  int tmp;
346 
347  if (value >= 60) {
348  av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
349  return 0;
350  }
351 
352  tmp= vlc_stage3_values[value];
353 
354  if ((value & ~3) > 0)
355  tmp += get_bits (gb, (value >> 2));
356  value = tmp;
357  }
358 
359  return value;
360 }
361 
362 
363 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
364 {
365  int value = qdm2_get_vlc (gb, vlc, 0, depth);
366 
367  return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
368 }
369 
370 
380 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
381  int i;
382 
383  for (i=0; i < length; i++)
384  value -= data[i];
385 
386  return (uint16_t)(value & 0xffff);
387 }
388 
389 
397 {
398  sub_packet->type = get_bits (gb, 8);
399 
400  if (sub_packet->type == 0) {
401  sub_packet->size = 0;
402  sub_packet->data = NULL;
403  } else {
404  sub_packet->size = get_bits (gb, 8);
405 
406  if (sub_packet->type & 0x80) {
407  sub_packet->size <<= 8;
408  sub_packet->size |= get_bits (gb, 8);
409  sub_packet->type &= 0x7f;
410  }
411 
412  if (sub_packet->type == 0x7f)
413  sub_packet->type |= (get_bits (gb, 8) << 8);
414 
415  sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
416  }
417 
418  av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
419  sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
420 }
421 
422 
431 {
432  while (list != NULL && list->packet != NULL) {
433  if (list->packet->type == type)
434  return list;
435  list = list->next;
436  }
437  return NULL;
438 }
439 
440 
448 {
449  int i, j, n, ch, sum;
450 
452 
453  for (ch = 0; ch < q->nb_channels; ch++)
454  for (i = 0; i < n; i++) {
455  sum = 0;
456 
457  for (j = 0; j < 8; j++)
458  sum += q->quantized_coeffs[ch][i][j];
459 
460  sum /= 8;
461  if (sum > 0)
462  sum--;
463 
464  for (j=0; j < 8; j++)
465  q->quantized_coeffs[ch][i][j] = sum;
466  }
467 }
468 
469 
477 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
478 {
479  int ch, j;
480 
482 
483  if (!q->nb_channels)
484  return;
485 
486  for (ch = 0; ch < q->nb_channels; ch++)
487  for (j = 0; j < 64; j++) {
488  q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
489  q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
490  }
491 }
492 
493 
502 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
503 {
504  int j,k;
505  int ch;
506  int run, case_val;
507  static const 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) > 22) {
512  run = 1;
513  case_val = 8;
514  } else {
515  switch (switchtable[coding_method[ch][sb][j]-8]) {
516  case 0: run = 10; case_val = 10; break;
517  case 1: run = 1; case_val = 16; break;
518  case 2: run = 5; case_val = 24; break;
519  case 3: run = 3; case_val = 30; break;
520  case 4: run = 1; case_val = 30; break;
521  case 5: run = 1; case_val = 8; break;
522  default: run = 1; case_val = 8; break;
523  }
524  }
525  for (k = 0; k < run; k++)
526  if (j + k < 128)
527  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
528  if (k > 0) {
530  //not debugged, almost never used
531  memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
532  memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
533  }
534  j += run;
535  }
536  }
537 }
538 
539 
547 static void fill_tone_level_array (QDM2Context *q, int flag)
548 {
549  int i, sb, ch, sb_used;
550  int tmp, tab;
551 
552  for (ch = 0; ch < q->nb_channels; ch++)
553  for (sb = 0; sb < 30; sb++)
554  for (i = 0; i < 8; i++) {
556  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
558  else
559  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
560  if(tmp < 0)
561  tmp += 0xff;
562  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
563  }
564 
565  sb_used = QDM2_SB_USED(q->sub_sampling);
566 
567  if ((q->superblocktype_2_3 != 0) && !flag) {
568  for (sb = 0; sb < sb_used; sb++)
569  for (ch = 0; ch < q->nb_channels; ch++)
570  for (i = 0; i < 64; i++) {
571  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
572  if (q->tone_level_idx[ch][sb][i] < 0)
573  q->tone_level[ch][sb][i] = 0;
574  else
575  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
576  }
577  } else {
578  tab = q->superblocktype_2_3 ? 0 : 1;
579  for (sb = 0; sb < sb_used; sb++) {
580  if ((sb >= 4) && (sb <= 23)) {
581  for (ch = 0; ch < q->nb_channels; ch++)
582  for (i = 0; i < 64; i++) {
583  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
584  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
585  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
586  q->tone_level_idx_hi2[ch][sb - 4];
587  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
588  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
589  q->tone_level[ch][sb][i] = 0;
590  else
591  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
592  }
593  } else {
594  if (sb > 4) {
595  for (ch = 0; ch < q->nb_channels; ch++)
596  for (i = 0; i < 64; i++) {
597  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
598  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
599  q->tone_level_idx_hi2[ch][sb - 4];
600  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
601  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
602  q->tone_level[ch][sb][i] = 0;
603  else
604  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
605  }
606  } else {
607  for (ch = 0; ch < q->nb_channels; ch++)
608  for (i = 0; i < 64; i++) {
609  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
610  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
611  q->tone_level[ch][sb][i] = 0;
612  else
613  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
614  }
615  }
616  }
617  }
618  }
619 
620  return;
621 }
622 
623 
638 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
639  sb_int8_array coding_method, int nb_channels,
640  int c, int superblocktype_2_3, int cm_table_select)
641 {
642  int ch, sb, j;
643  int tmp, acc, esp_40, comp;
644  int add1, add2, add3, add4;
645  int64_t multres;
646 
647  if (!superblocktype_2_3) {
648  /* This case is untested, no samples available */
649  av_log_ask_for_sample(NULL, "!superblocktype_2_3");
650  return;
651  for (ch = 0; ch < nb_channels; ch++)
652  for (sb = 0; sb < 30; sb++) {
653  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
654  add1 = tone_level_idx[ch][sb][j] - 10;
655  if (add1 < 0)
656  add1 = 0;
657  add2 = add3 = add4 = 0;
658  if (sb > 1) {
659  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
660  if (add2 < 0)
661  add2 = 0;
662  }
663  if (sb > 0) {
664  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
665  if (add3 < 0)
666  add3 = 0;
667  }
668  if (sb < 29) {
669  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
670  if (add4 < 0)
671  add4 = 0;
672  }
673  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
674  if (tmp < 0)
675  tmp = 0;
676  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
677  }
678  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
679  }
680  acc = 0;
681  for (ch = 0; ch < nb_channels; ch++)
682  for (sb = 0; sb < 30; sb++)
683  for (j = 0; j < 64; j++)
684  acc += tone_level_idx_temp[ch][sb][j];
685 
686  multres = 0x66666667LL * (acc * 10);
687  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
688  for (ch = 0; ch < nb_channels; ch++)
689  for (sb = 0; sb < 30; sb++)
690  for (j = 0; j < 64; j++) {
691  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
692  if (comp < 0)
693  comp += 0xff;
694  comp /= 256; // signed shift
695  switch(sb) {
696  case 0:
697  if (comp < 30)
698  comp = 30;
699  comp += 15;
700  break;
701  case 1:
702  if (comp < 24)
703  comp = 24;
704  comp += 10;
705  break;
706  case 2:
707  case 3:
708  case 4:
709  if (comp < 16)
710  comp = 16;
711  }
712  if (comp <= 5)
713  tmp = 0;
714  else if (comp <= 10)
715  tmp = 10;
716  else if (comp <= 16)
717  tmp = 16;
718  else if (comp <= 24)
719  tmp = -1;
720  else
721  tmp = 0;
722  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
723  }
724  for (sb = 0; sb < 30; sb++)
725  fix_coding_method_array(sb, nb_channels, coding_method);
726  for (ch = 0; ch < nb_channels; ch++)
727  for (sb = 0; sb < 30; sb++)
728  for (j = 0; j < 64; j++)
729  if (sb >= 10) {
730  if (coding_method[ch][sb][j] < 10)
731  coding_method[ch][sb][j] = 10;
732  } else {
733  if (sb >= 2) {
734  if (coding_method[ch][sb][j] < 16)
735  coding_method[ch][sb][j] = 16;
736  } else {
737  if (coding_method[ch][sb][j] < 30)
738  coding_method[ch][sb][j] = 30;
739  }
740  }
741  } else { // superblocktype_2_3 != 0
742  for (ch = 0; ch < nb_channels; ch++)
743  for (sb = 0; sb < 30; sb++)
744  for (j = 0; j < 64; j++)
745  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
746  }
747 
748  return;
749 }
750 
751 
763 static int synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
764 {
765  int sb, j, k, n, ch, run, channels;
766  int joined_stereo, zero_encoding, chs;
767  int type34_first;
768  float type34_div = 0;
769  float type34_predictor;
770  float samples[10];
771  int sign_bits[16] = {0};
772 
773  if (length == 0) {
774  // If no data use noise
775  for (sb=sb_min; sb < sb_max; sb++)
777 
778  return 0;
779  }
780 
781  for (sb = sb_min; sb < sb_max; sb++) {
783 
784  channels = q->nb_channels;
785 
786  if (q->nb_channels <= 1 || sb < 12)
787  joined_stereo = 0;
788  else if (sb >= 24)
789  joined_stereo = 1;
790  else
791  joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
792 
793  if (joined_stereo) {
794  if (get_bits_left(gb) >= 16)
795  for (j = 0; j < 16; j++)
796  sign_bits[j] = get_bits1 (gb);
797 
798  if (q->coding_method[0][sb][0] <= 0) {
799  av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
800  return AVERROR_INVALIDDATA;
801  }
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 = (get_bits_left(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 (get_bits_left(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  if (n >= 243) {
829  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
830  return AVERROR_INVALIDDATA;
831  }
832 
833  for (k = 0; k < 5; k++)
834  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
835  }
836  for (k = 0; k < 5; k++)
837  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
838  } else {
839  for (k = 0; k < 10; k++)
840  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
841  }
842  run = 10;
843  break;
844 
845  case 10:
846  if (get_bits_left(gb) >= 1) {
847  float f = 0.81;
848 
849  if (get_bits1(gb))
850  f = -f;
851  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
852  samples[0] = f;
853  } else {
854  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
855  }
856  run = 1;
857  break;
858 
859  case 16:
860  if (get_bits_left(gb) >= 10) {
861  if (zero_encoding) {
862  for (k = 0; k < 5; k++) {
863  if ((j + k) >= 128)
864  break;
865  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
866  }
867  } else {
868  n = get_bits (gb, 8);
869  if (n >= 243) {
870  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
871  return AVERROR_INVALIDDATA;
872  }
873 
874  for (k = 0; k < 5; k++)
875  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
876  }
877  } else {
878  for (k = 0; k < 5; k++)
879  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
880  }
881  run = 5;
882  break;
883 
884  case 24:
885  if (get_bits_left(gb) >= 7) {
886  n = get_bits(gb, 7);
887  if (n >= 125) {
888  av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
889  return AVERROR_INVALIDDATA;
890  }
891 
892  for (k = 0; k < 3; k++)
893  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
894  } else {
895  for (k = 0; k < 3; k++)
896  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
897  }
898  run = 3;
899  break;
900 
901  case 30:
902  if (get_bits_left(gb) >= 4) {
903  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
904  if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
905  av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
906  return AVERROR_INVALIDDATA;
907  }
908  samples[0] = type30_dequant[index];
909  } else
910  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
911 
912  run = 1;
913  break;
914 
915  case 34:
916  if (get_bits_left(gb) >= 7) {
917  if (type34_first) {
918  type34_div = (float)(1 << get_bits(gb, 2));
919  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
920  type34_predictor = samples[0];
921  type34_first = 0;
922  } else {
923  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
924  if (index >= FF_ARRAY_ELEMS(type34_delta)) {
925  av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
926  return AVERROR_INVALIDDATA;
927  }
928  samples[0] = type34_delta[index] / type34_div + type34_predictor;
929  type34_predictor = samples[0];
930  }
931  } else {
932  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
933  }
934  run = 1;
935  break;
936 
937  default:
938  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
939  run = 1;
940  break;
941  }
942 
943  if (joined_stereo) {
944  float tmp[10][MPA_MAX_CHANNELS];
945  for (k = 0; k < run; k++) {
946  tmp[k][0] = samples[k];
947  if ((j + k) < 128)
948  tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
949  }
950  for (chs = 0; chs < q->nb_channels; chs++)
951  for (k = 0; k < run; k++)
952  if ((j + k) < 128)
953  q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
954  } else {
955  for (k = 0; k < run; k++)
956  if ((j + k) < 128)
957  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
958  }
959 
960  j += run;
961  } // j loop
962  } // channel loop
963  } // subband loop
964  return 0;
965 }
966 
967 
976 static int init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
977 {
978  int i, k, run, level, diff;
979 
980  if (get_bits_left(gb) < 16)
981  return -1;
982  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
983 
984  quantized_coeffs[0] = level;
985 
986  for (i = 0; i < 7; ) {
987  if (get_bits_left(gb) < 16)
988  return -1;
989  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
990 
991  if (i + run >= 8)
992  return -1;
993 
994  if (get_bits_left(gb) < 16)
995  return -1;
996  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
997 
998  for (k = 1; k <= run; k++)
999  quantized_coeffs[i + k] = (level + ((k * diff) / run));
1000 
1001  level += diff;
1002  i += run;
1003  }
1004  return 0;
1005 }
1006 
1007 
1017 {
1018  int sb, j, k, n, ch;
1019 
1020  for (ch = 0; ch < q->nb_channels; ch++) {
1022 
1023  if (get_bits_left(gb) < 16) {
1024  memset(q->quantized_coeffs[ch][0], 0, 8);
1025  break;
1026  }
1027  }
1028 
1029  n = q->sub_sampling + 1;
1030 
1031  for (sb = 0; sb < n; sb++)
1032  for (ch = 0; ch < q->nb_channels; ch++)
1033  for (j = 0; j < 8; j++) {
1034  if (get_bits_left(gb) < 1)
1035  break;
1036  if (get_bits1(gb)) {
1037  for (k=0; k < 8; k++) {
1038  if (get_bits_left(gb) < 16)
1039  break;
1040  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1041  }
1042  } else {
1043  for (k=0; k < 8; k++)
1044  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1045  }
1046  }
1047 
1048  n = QDM2_SB_USED(q->sub_sampling) - 4;
1049 
1050  for (sb = 0; sb < n; sb++)
1051  for (ch = 0; ch < q->nb_channels; ch++) {
1052  if (get_bits_left(gb) < 16)
1053  break;
1054  q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1055  if (sb > 19)
1056  q->tone_level_idx_hi2[ch][sb] -= 16;
1057  else
1058  for (j = 0; j < 8; j++)
1059  q->tone_level_idx_mid[ch][sb][j] = -16;
1060  }
1061 
1062  n = QDM2_SB_USED(q->sub_sampling) - 5;
1063 
1064  for (sb = 0; sb < n; sb++)
1065  for (ch = 0; ch < q->nb_channels; ch++)
1066  for (j = 0; j < 8; j++) {
1067  if (get_bits_left(gb) < 16)
1068  break;
1069  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1070  }
1071 }
1072 
1080 {
1081  GetBitContext gb;
1082  int i, j, k, n, ch, run, level, diff;
1083 
1084  init_get_bits(&gb, node->packet->data, node->packet->size*8);
1085 
1086  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1087 
1088  for (i = 1; i < n; i++)
1089  for (ch=0; ch < q->nb_channels; ch++) {
1090  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1091  q->quantized_coeffs[ch][i][0] = level;
1092 
1093  for (j = 0; j < (8 - 1); ) {
1094  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1095  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1096 
1097  if (j + run >= 8)
1098  return -1;
1099 
1100  for (k = 1; k <= run; k++)
1101  q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1102 
1103  level += diff;
1104  j += run;
1105  }
1106  }
1107 
1108  for (ch = 0; ch < q->nb_channels; ch++)
1109  for (i = 0; i < 8; i++)
1110  q->quantized_coeffs[ch][0][i] = 0;
1111 
1112  return 0;
1113 }
1114 
1115 
1123 {
1124  GetBitContext gb;
1125 
1126  if (node) {
1127  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1129  fill_tone_level_array(q, 1);
1130  } else {
1131  fill_tone_level_array(q, 0);
1132  }
1133 }
1134 
1135 
1143 {
1144  GetBitContext gb;
1145  int length = 0;
1146 
1147  if (node) {
1148  length = node->packet->size * 8;
1149  init_get_bits(&gb, node->packet->data, length);
1150  }
1151 
1152  if (length >= 32) {
1153  int c = get_bits (&gb, 13);
1154 
1155  if (c > 3)
1158  }
1159 
1160  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1161 }
1162 
1163 
1171 {
1172  GetBitContext gb;
1173  int length = 0;
1174 
1175  if (node) {
1176  length = node->packet->size * 8;
1177  init_get_bits(&gb, node->packet->data, length);
1178  }
1179 
1180  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1181 }
1182 
1190 {
1191  QDM2SubPNode *nodes[4];
1192 
1193  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1194  if (nodes[0] != NULL)
1195  process_subpacket_9(q, nodes[0]);
1196 
1197  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1198  if (nodes[1] != NULL)
1199  process_subpacket_10(q, nodes[1]);
1200  else
1202 
1203  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1204  if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1205  process_subpacket_11(q, nodes[2]);
1206  else
1208 
1209  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1210  if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1211  process_subpacket_12(q, nodes[3]);
1212  else
1214 }
1215 
1216 
1223 {
1224  GetBitContext gb;
1225  QDM2SubPacket header, *packet;
1226  int i, packet_bytes, sub_packet_size, sub_packets_D;
1227  unsigned int next_index = 0;
1228 
1229  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1230  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1231  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1232 
1233  q->sub_packets_B = 0;
1234  sub_packets_D = 0;
1235 
1236  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1237 
1239  qdm2_decode_sub_packet_header(&gb, &header);
1240 
1241  if (header.type < 2 || header.type >= 8) {
1242  q->has_errors = 1;
1243  av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1244  return;
1245  }
1246 
1247  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1248  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1249 
1250  init_get_bits(&gb, header.data, header.size*8);
1251 
1252  if (header.type == 2 || header.type == 4 || header.type == 5) {
1253  int csum = 257 * get_bits(&gb, 8);
1254  csum += 2 * get_bits(&gb, 8);
1255 
1256  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1257 
1258  if (csum != 0) {
1259  q->has_errors = 1;
1260  av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1261  return;
1262  }
1263  }
1264 
1265  q->sub_packet_list_B[0].packet = NULL;
1266  q->sub_packet_list_D[0].packet = NULL;
1267 
1268  for (i = 0; i < 6; i++)
1269  if (--q->fft_level_exp[i] < 0)
1270  q->fft_level_exp[i] = 0;
1271 
1272  for (i = 0; packet_bytes > 0; i++) {
1273  int j;
1274 
1275  if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1276  SAMPLES_NEEDED_2("too many packet bytes");
1277  return;
1278  }
1279 
1280  q->sub_packet_list_A[i].next = NULL;
1281 
1282  if (i > 0) {
1283  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1284 
1285  /* seek to next block */
1286  init_get_bits(&gb, header.data, header.size*8);
1287  skip_bits(&gb, next_index*8);
1288 
1289  if (next_index >= header.size)
1290  break;
1291  }
1292 
1293  /* decode subpacket */
1294  packet = &q->sub_packets[i];
1295  qdm2_decode_sub_packet_header(&gb, packet);
1296  next_index = packet->size + get_bits_count(&gb) / 8;
1297  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1298 
1299  if (packet->type == 0)
1300  break;
1301 
1302  if (sub_packet_size > packet_bytes) {
1303  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1304  break;
1305  packet->size += packet_bytes - sub_packet_size;
1306  }
1307 
1308  packet_bytes -= sub_packet_size;
1309 
1310  /* add subpacket to 'all subpackets' list */
1311  q->sub_packet_list_A[i].packet = packet;
1312 
1313  /* add subpacket to related list */
1314  if (packet->type == 8) {
1315  SAMPLES_NEEDED_2("packet type 8");
1316  return;
1317  } else if (packet->type >= 9 && packet->type <= 12) {
1318  /* packets for MPEG Audio like Synthesis Filter */
1319  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1320  } else if (packet->type == 13) {
1321  for (j = 0; j < 6; j++)
1322  q->fft_level_exp[j] = get_bits(&gb, 6);
1323  } else if (packet->type == 14) {
1324  for (j = 0; j < 6; j++)
1325  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1326  } else if (packet->type == 15) {
1327  SAMPLES_NEEDED_2("packet type 15")
1328  return;
1329  } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1330  /* packets for FFT */
1332  }
1333  } // Packet bytes loop
1334 
1335 /* **************************************************************** */
1336  if (q->sub_packet_list_D[0].packet != NULL) {
1338  q->do_synth_filter = 1;
1339  } else if (q->do_synth_filter) {
1343  }
1344 /* **************************************************************** */
1345 }
1346 
1347 
1348 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1349  int offset, int duration, int channel,
1350  int exp, int phase)
1351 {
1352  if (q->fft_coefs_min_index[duration] < 0)
1354 
1355  q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1356  q->fft_coefs[q->fft_coefs_index].channel = channel;
1358  q->fft_coefs[q->fft_coefs_index].exp = exp;
1359  q->fft_coefs[q->fft_coefs_index].phase = phase;
1360  q->fft_coefs_index++;
1361 }
1362 
1363 
1365 {
1366  int channel, stereo, phase, exp;
1367  int local_int_4, local_int_8, stereo_phase, local_int_10;
1368  int local_int_14, stereo_exp, local_int_20, local_int_28;
1369  int n, offset;
1370 
1371  local_int_4 = 0;
1372  local_int_28 = 0;
1373  local_int_20 = 2;
1374  local_int_8 = (4 - duration);
1375  local_int_10 = 1 << (q->group_order - duration - 1);
1376  offset = 1;
1377 
1378  while (get_bits_left(gb)>0) {
1379  if (q->superblocktype_2_3) {
1380  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1381  if (get_bits_left(gb)<0) {
1382  if(local_int_4 < q->group_size)
1383  av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1384  return;
1385  }
1386  offset = 1;
1387  if (n == 0) {
1388  local_int_4 += local_int_10;
1389  local_int_28 += (1 << local_int_8);
1390  } else {
1391  local_int_4 += 8*local_int_10;
1392  local_int_28 += (8 << local_int_8);
1393  }
1394  }
1395  offset += (n - 2);
1396  } else {
1397  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1398  while (offset >= (local_int_10 - 1)) {
1399  offset += (1 - (local_int_10 - 1));
1400  local_int_4 += local_int_10;
1401  local_int_28 += (1 << local_int_8);
1402  }
1403  }
1404 
1405  if (local_int_4 >= q->group_size)
1406  return;
1407 
1408  local_int_14 = (offset >> local_int_8);
1409  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1410  return;
1411 
1412  if (q->nb_channels > 1) {
1413  channel = get_bits1(gb);
1414  stereo = get_bits1(gb);
1415  } else {
1416  channel = 0;
1417  stereo = 0;
1418  }
1419 
1420  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1421  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1422  exp = (exp < 0) ? 0 : exp;
1423 
1424  phase = get_bits(gb, 3);
1425  stereo_exp = 0;
1426  stereo_phase = 0;
1427 
1428  if (stereo) {
1429  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1430  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1431  if (stereo_phase < 0)
1432  stereo_phase += 8;
1433  }
1434 
1435  if (q->frequency_range > (local_int_14 + 1)) {
1436  int sub_packet = (local_int_20 + local_int_28);
1437 
1438  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1439  if (stereo)
1440  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1441  }
1442 
1443  offset++;
1444  }
1445 }
1446 
1447 
1449 {
1450  int i, j, min, max, value, type, unknown_flag;
1451  GetBitContext gb;
1452 
1453  if (q->sub_packet_list_B[0].packet == NULL)
1454  return;
1455 
1456  /* reset minimum indexes for FFT coefficients */
1457  q->fft_coefs_index = 0;
1458  for (i=0; i < 5; i++)
1459  q->fft_coefs_min_index[i] = -1;
1460 
1461  /* process subpackets ordered by type, largest type first */
1462  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1463  QDM2SubPacket *packet= NULL;
1464 
1465  /* find subpacket with largest type less than max */
1466  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1467  value = q->sub_packet_list_B[j].packet->type;
1468  if (value > min && value < max) {
1469  min = value;
1470  packet = q->sub_packet_list_B[j].packet;
1471  }
1472  }
1473 
1474  max = min;
1475 
1476  /* check for errors (?) */
1477  if (!packet)
1478  return;
1479 
1480  if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1481  return;
1482 
1483  /* decode FFT tones */
1484  init_get_bits (&gb, packet->data, packet->size*8);
1485 
1486  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1487  unknown_flag = 1;
1488  else
1489  unknown_flag = 0;
1490 
1491  type = packet->type;
1492 
1493  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1494  int duration = q->sub_sampling + 5 - (type & 15);
1495 
1496  if (duration >= 0 && duration < 4)
1497  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1498  } else if (type == 31) {
1499  for (j=0; j < 4; j++)
1500  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1501  } else if (type == 46) {
1502  for (j=0; j < 6; j++)
1503  q->fft_level_exp[j] = get_bits(&gb, 6);
1504  for (j=0; j < 4; j++)
1505  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1506  }
1507  } // Loop on B packets
1508 
1509  /* calculate maximum indexes for FFT coefficients */
1510  for (i = 0, j = -1; i < 5; i++)
1511  if (q->fft_coefs_min_index[i] >= 0) {
1512  if (j >= 0)
1514  j = i;
1515  }
1516  if (j >= 0)
1518 }
1519 
1520 
1522 {
1523  float level, f[6];
1524  int i;
1525  QDM2Complex c;
1526  const double iscale = 2.0*M_PI / 512.0;
1527 
1528  tone->phase += tone->phase_shift;
1529 
1530  /* calculate current level (maximum amplitude) of tone */
1531  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1532  c.im = level * sin(tone->phase*iscale);
1533  c.re = level * cos(tone->phase*iscale);
1534 
1535  /* generate FFT coefficients for tone */
1536  if (tone->duration >= 3 || tone->cutoff >= 3) {
1537  tone->complex[0].im += c.im;
1538  tone->complex[0].re += c.re;
1539  tone->complex[1].im -= c.im;
1540  tone->complex[1].re -= c.re;
1541  } else {
1542  f[1] = -tone->table[4];
1543  f[0] = tone->table[3] - tone->table[0];
1544  f[2] = 1.0 - tone->table[2] - tone->table[3];
1545  f[3] = tone->table[1] + tone->table[4] - 1.0;
1546  f[4] = tone->table[0] - tone->table[1];
1547  f[5] = tone->table[2];
1548  for (i = 0; i < 2; i++) {
1549  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1550  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1551  }
1552  for (i = 0; i < 4; i++) {
1553  tone->complex[i].re += c.re * f[i+2];
1554  tone->complex[i].im += c.im * f[i+2];
1555  }
1556  }
1557 
1558  /* copy the tone if it has not yet died out */
1559  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1560  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1561  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1562  }
1563 }
1564 
1565 
1566 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1567 {
1568  int i, j, ch;
1569  const double iscale = 0.25 * M_PI;
1570 
1571  for (ch = 0; ch < q->channels; ch++) {
1572  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1573  }
1574 
1575 
1576  /* apply FFT tones with duration 4 (1 FFT period) */
1577  if (q->fft_coefs_min_index[4] >= 0)
1578  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1579  float level;
1580  QDM2Complex c;
1581 
1582  if (q->fft_coefs[i].sub_packet != sub_packet)
1583  break;
1584 
1585  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1586  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];
1587 
1588  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1589  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1590  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1591  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1592  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1593  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1594  }
1595 
1596  /* generate existing FFT tones */
1597  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1599  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1600  }
1601 
1602  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1603  for (i = 0; i < 4; i++)
1604  if (q->fft_coefs_min_index[i] >= 0) {
1605  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1606  int offset, four_i;
1607  FFTTone tone;
1608 
1609  if (q->fft_coefs[j].sub_packet != sub_packet)
1610  break;
1611 
1612  four_i = (4 - i);
1613  offset = q->fft_coefs[j].offset >> four_i;
1614  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1615 
1616  if (offset < q->frequency_range) {
1617  if (offset < 2)
1618  tone.cutoff = offset;
1619  else
1620  tone.cutoff = (offset >= 60) ? 3 : 2;
1621 
1622  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];
1623  tone.complex = &q->fft.complex[ch][offset];
1624  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1625  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1626  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1627  tone.duration = i;
1628  tone.time_index = 0;
1629 
1630  qdm2_fft_generate_tone(q, &tone);
1631  }
1632  }
1633  q->fft_coefs_min_index[i] = j;
1634  }
1635 }
1636 
1637 
1638 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1639 {
1640  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1641  float *out = q->output_buffer + channel;
1642  int i;
1643  q->fft.complex[channel][0].re *= 2.0f;
1644  q->fft.complex[channel][0].im = 0.0f;
1645  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1646  /* add samples to output buffer */
1647  for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1648  out[0] += q->fft.complex[channel][i].re * gain;
1649  out[q->channels] += q->fft.complex[channel][i].im * gain;
1650  out += 2 * q->channels;
1651  }
1652 }
1653 
1654 
1660 {
1661  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1662 
1663  /* copy sb_samples */
1664  sb_used = QDM2_SB_USED(q->sub_sampling);
1665 
1666  for (ch = 0; ch < q->channels; ch++)
1667  for (i = 0; i < 8; i++)
1668  for (k=sb_used; k < SBLIMIT; k++)
1669  q->sb_samples[ch][(8 * index) + i][k] = 0;
1670 
1671  for (ch = 0; ch < q->nb_channels; ch++) {
1672  float *samples_ptr = q->samples + ch;
1673 
1674  for (i = 0; i < 8; i++) {
1676  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1677  ff_mpa_synth_window_float, &dither_state,
1678  samples_ptr, q->nb_channels,
1679  q->sb_samples[ch][(8 * index) + i]);
1680  samples_ptr += 32 * q->nb_channels;
1681  }
1682  }
1683 
1684  /* add samples to output buffer */
1685  sub_sampling = (4 >> q->sub_sampling);
1686 
1687  for (ch = 0; ch < q->channels; ch++)
1688  for (i = 0; i < q->frame_size; i++)
1689  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1690 }
1691 
1692 
1698 static av_cold void qdm2_init(QDM2Context *q) {
1699  static int initialized = 0;
1700 
1701  if (initialized != 0)
1702  return;
1703  initialized = 1;
1704 
1705  qdm2_init_vlc();
1708  rnd_table_init();
1710 
1711  av_log(NULL, AV_LOG_DEBUG, "init done\n");
1712 }
1713 
1714 
1719 {
1720  QDM2Context *s = avctx->priv_data;
1721  uint8_t *extradata;
1722  int extradata_size;
1723  int tmp_val, tmp, size;
1724 
1725  /* extradata parsing
1726 
1727  Structure:
1728  wave {
1729  frma (QDM2)
1730  QDCA
1731  QDCP
1732  }
1733 
1734  32 size (including this field)
1735  32 tag (=frma)
1736  32 type (=QDM2 or QDMC)
1737 
1738  32 size (including this field, in bytes)
1739  32 tag (=QDCA) // maybe mandatory parameters
1740  32 unknown (=1)
1741  32 channels (=2)
1742  32 samplerate (=44100)
1743  32 bitrate (=96000)
1744  32 block size (=4096)
1745  32 frame size (=256) (for one channel)
1746  32 packet size (=1300)
1747 
1748  32 size (including this field, in bytes)
1749  32 tag (=QDCP) // maybe some tuneable parameters
1750  32 float1 (=1.0)
1751  32 zero ?
1752  32 float2 (=1.0)
1753  32 float3 (=1.0)
1754  32 unknown (27)
1755  32 unknown (8)
1756  32 zero ?
1757  */
1758 
1759  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1760  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1761  return -1;
1762  }
1763 
1764  extradata = avctx->extradata;
1765  extradata_size = avctx->extradata_size;
1766 
1767  while (extradata_size > 7) {
1768  if (!memcmp(extradata, "frmaQDM", 7))
1769  break;
1770  extradata++;
1771  extradata_size--;
1772  }
1773 
1774  if (extradata_size < 12) {
1775  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1776  extradata_size);
1777  return -1;
1778  }
1779 
1780  if (memcmp(extradata, "frmaQDM", 7)) {
1781  av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1782  return -1;
1783  }
1784 
1785  if (extradata[7] == 'C') {
1786 // s->is_qdmc = 1;
1787  av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1788  return -1;
1789  }
1790 
1791  extradata += 8;
1792  extradata_size -= 8;
1793 
1794  size = AV_RB32(extradata);
1795 
1796  if(size > extradata_size){
1797  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1798  extradata_size, size);
1799  return -1;
1800  }
1801 
1802  extradata += 4;
1803  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1804  if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1805  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1806  return -1;
1807  }
1808 
1809  extradata += 8;
1810 
1811  avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1812  extradata += 4;
1813  if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1814  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1815  return AVERROR_INVALIDDATA;
1816  }
1817  avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1819 
1820  avctx->sample_rate = AV_RB32(extradata);
1821  extradata += 4;
1822 
1823  avctx->bit_rate = AV_RB32(extradata);
1824  extradata += 4;
1825 
1826  s->group_size = AV_RB32(extradata);
1827  extradata += 4;
1828 
1829  s->fft_size = AV_RB32(extradata);
1830  extradata += 4;
1831 
1832  s->checksum_size = AV_RB32(extradata);
1833  if (s->checksum_size >= 1U << 28) {
1834  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1835  return AVERROR_INVALIDDATA;
1836  }
1837 
1838  s->fft_order = av_log2(s->fft_size) + 1;
1839 
1840  // something like max decodable tones
1841  s->group_order = av_log2(s->group_size) + 1;
1842  s->frame_size = s->group_size / 16; // 16 iterations per super block
1843 
1845  return AVERROR_INVALIDDATA;
1846 
1847  s->sub_sampling = s->fft_order - 7;
1848  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1849 
1850  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1851  case 0: tmp = 40; break;
1852  case 1: tmp = 48; break;
1853  case 2: tmp = 56; break;
1854  case 3: tmp = 72; break;
1855  case 4: tmp = 80; break;
1856  case 5: tmp = 100;break;
1857  default: tmp=s->sub_sampling; break;
1858  }
1859  tmp_val = 0;
1860  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1861  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1862  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1863  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1864  s->cm_table_select = tmp_val;
1865 
1866  if (s->sub_sampling == 0)
1867  tmp = 7999;
1868  else
1869  tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1870  /*
1871  0: 7999 -> 0
1872  1: 20000 -> 2
1873  2: 28000 -> 2
1874  */
1875  if (tmp < 8000)
1876  s->coeff_per_sb_select = 0;
1877  else if (tmp <= 16000)
1878  s->coeff_per_sb_select = 1;
1879  else
1880  s->coeff_per_sb_select = 2;
1881 
1882  // Fail on unknown fft order
1883  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1884  av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1885  return -1;
1886  }
1887 
1889  ff_mpadsp_init(&s->mpadsp);
1890 
1891  qdm2_init(s);
1892 
1893  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1894 
1895  return 0;
1896 }
1897 
1898 
1900 {
1901  QDM2Context *s = avctx->priv_data;
1902 
1903  ff_rdft_end(&s->rdft_ctx);
1904 
1905  return 0;
1906 }
1907 
1908 
1909 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1910 {
1911  int ch, i;
1912  const int frame_size = (q->frame_size * q->channels);
1913 
1914  if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1915  return -1;
1916 
1917  /* select input buffer */
1918  q->compressed_data = in;
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  AVFrame *frame = data;
1976  const uint8_t *buf = avpkt->data;
1977  int buf_size = avpkt->size;
1978  QDM2Context *s = avctx->priv_data;
1979  int16_t *out;
1980  int i, ret;
1981 
1982  if(!buf)
1983  return 0;
1984  if(buf_size < s->checksum_size)
1985  return -1;
1986 
1987  /* get output buffer */
1988  frame->nb_samples = 16 * s->frame_size;
1989  if ((ret = ff_get_buffer(avctx, frame)) < 0) {
1990  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1991  return ret;
1992  }
1993  out = (int16_t *)frame->data[0];
1994 
1995  for (i = 0; i < 16; i++) {
1996  if (qdm2_decode(s, buf, out) < 0)
1997  return -1;
1998  out += s->channels * s->frame_size;
1999  }
2000 
2001  *got_frame_ptr = 1;
2002 
2003  return s->checksum_size;
2004 }
2005 
2007 {
2008  .name = "qdm2",
2009  .type = AVMEDIA_TYPE_AUDIO,
2010  .id = AV_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 };