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 "internal.h"
41 #include "get_bits.h"
42 #include "dsputil.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 BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
74 
75 #define SAMPLES_NEEDED \
76  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
77 
78 #define SAMPLES_NEEDED_2(why) \
79  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
80 
81 #define QDM2_MAX_FRAME_SIZE 512
82 
83 typedef int8_t sb_int8_array[2][30][64];
84 
88 typedef struct {
89  int type;
90  unsigned int size;
91  const uint8_t *data;
93 
97 typedef struct QDM2SubPNode {
99  struct QDM2SubPNode *next;
100 } QDM2SubPNode;
101 
102 typedef struct {
103  float re;
104  float im;
105 } QDM2Complex;
106 
107 typedef struct {
108  float level;
110  const float *table;
111  int phase;
113  int duration;
114  short time_index;
115  short cutoff;
116 } FFTTone;
117 
118 typedef struct {
119  int16_t sub_packet;
120  uint8_t channel;
121  int16_t offset;
122  int16_t exp;
123  uint8_t phase;
125 
126 typedef struct {
128 } QDM2FFT;
129 
133 typedef struct {
135 
138  int channels;
140  int fft_size;
142 
145  int fft_order;
152 
154  QDM2SubPacket sub_packets[16];
155  QDM2SubPNode sub_packet_list_A[16];
156  QDM2SubPNode sub_packet_list_B[16];
158  QDM2SubPNode sub_packet_list_C[16];
159  QDM2SubPNode sub_packet_list_D[16];
160 
162  FFTTone fft_tones[1000];
165  FFTCoefficient fft_coefs[1000];
167  int fft_coefs_min_index[5];
168  int fft_coefs_max_index[5];
169  int fft_level_exp[6];
172 
174  const uint8_t *compressed_data;
176  float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
177 
180  DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
181  int synth_buf_offset[MPA_MAX_CHANNELS];
182  DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
184 
186  float tone_level[MPA_MAX_CHANNELS][30][64];
187  int8_t coding_method[MPA_MAX_CHANNELS][30][64];
188  int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
189  int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
190  int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
191  int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
192  int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
193  int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
194  int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
195 
196  // Flags
200 
202  int noise_idx;
203 } QDM2Context;
204 
205 
207 
221 
222 static const uint16_t qdm2_vlc_offs[] = {
223  0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
224 };
225 
226 static av_cold void qdm2_init_vlc(void)
227 {
228  static int vlcs_initialized = 0;
229  static VLC_TYPE qdm2_table[3838][2];
230 
231  if (!vlcs_initialized) {
232 
233  vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
234  vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
235  init_vlc (&vlc_tab_level, 8, 24,
238 
239  vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
240  vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
241  init_vlc (&vlc_tab_diff, 8, 37,
242  vlc_tab_diff_huffbits, 1, 1,
244 
245  vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
246  vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
247  init_vlc (&vlc_tab_run, 5, 6,
248  vlc_tab_run_huffbits, 1, 1,
250 
251  fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
252  fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
253  init_vlc (&fft_level_exp_alt_vlc, 8, 28,
256 
257 
258  fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
259  fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
260  init_vlc (&fft_level_exp_vlc, 8, 20,
263 
264  fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
265  fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
266  init_vlc (&fft_stereo_exp_vlc, 6, 7,
269 
270  fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
271  fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
272  init_vlc (&fft_stereo_phase_vlc, 6, 9,
275 
276  vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
277  vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
278  init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
281 
282  vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
283  vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
284  init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
287 
288  vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
289  vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
290  init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
293 
294  vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
295  vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
296  init_vlc (&vlc_tab_type30, 6, 9,
299 
300  vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
301  vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
302  init_vlc (&vlc_tab_type34, 5, 10,
305 
306  vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
307  vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
308  init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
311 
312  vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
313  vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
314  init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
317 
318  vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
319  vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
320  init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
323 
324  vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
325  vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
326  init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
329 
330  vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
331  vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
332  init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
335 
336  vlcs_initialized=1;
337  }
338 }
339 
340 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
341 {
342  int value;
343 
344  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
345 
346  /* stage-2, 3 bits exponent escape sequence */
347  if (value-- == 0)
348  value = get_bits (gb, get_bits (gb, 3) + 1);
349 
350  /* stage-3, optional */
351  if (flag) {
352  int 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 int fix_coding_method_array(int sb, int channels,
503  sb_int8_array coding_method)
504 {
505  int j,k;
506  int ch;
507  int run, case_val;
508  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};
509 
510  for (ch = 0; ch < channels; ch++) {
511  for (j = 0; j < 64; ) {
512  if (coding_method[ch][sb][j] < 8)
513  return -1;
514  if ((coding_method[ch][sb][j] - 8) > 22) {
515  run = 1;
516  case_val = 8;
517  } else {
518  switch (switchtable[coding_method[ch][sb][j]-8]) {
519  case 0: run = 10; case_val = 10; break;
520  case 1: run = 1; case_val = 16; break;
521  case 2: run = 5; case_val = 24; break;
522  case 3: run = 3; case_val = 30; break;
523  case 4: run = 1; case_val = 30; break;
524  case 5: run = 1; case_val = 8; break;
525  default: run = 1; case_val = 8; break;
526  }
527  }
528  for (k = 0; k < run; k++)
529  if (j + k < 128)
530  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
531  if (k > 0) {
533  //not debugged, almost never used
534  memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
535  memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
536  }
537  j += run;
538  }
539  }
540  return 0;
541 }
542 
543 
551 static void fill_tone_level_array (QDM2Context *q, int flag)
552 {
553  int i, sb, ch, sb_used;
554  int tmp, tab;
555 
556  // This should never happen
557  if (q->nb_channels <= 0)
558  return;
559 
560  for (ch = 0; ch < q->nb_channels; ch++)
561  for (sb = 0; sb < 30; sb++)
562  for (i = 0; i < 8; i++) {
564  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
566  else
567  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
568  if(tmp < 0)
569  tmp += 0xff;
570  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
571  }
572 
573  sb_used = QDM2_SB_USED(q->sub_sampling);
574 
575  if ((q->superblocktype_2_3 != 0) && !flag) {
576  for (sb = 0; sb < sb_used; sb++)
577  for (ch = 0; ch < q->nb_channels; ch++)
578  for (i = 0; i < 64; i++) {
579  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
580  if (q->tone_level_idx[ch][sb][i] < 0)
581  q->tone_level[ch][sb][i] = 0;
582  else
583  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
584  }
585  } else {
586  tab = q->superblocktype_2_3 ? 0 : 1;
587  for (sb = 0; sb < sb_used; sb++) {
588  if ((sb >= 4) && (sb <= 23)) {
589  for (ch = 0; ch < q->nb_channels; ch++)
590  for (i = 0; i < 64; i++) {
591  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
592  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
593  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
594  q->tone_level_idx_hi2[ch][sb - 4];
595  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
596  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
597  q->tone_level[ch][sb][i] = 0;
598  else
599  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
600  }
601  } else {
602  if (sb > 4) {
603  for (ch = 0; ch < q->nb_channels; ch++)
604  for (i = 0; i < 64; i++) {
605  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
606  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
607  q->tone_level_idx_hi2[ch][sb - 4];
608  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
609  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
610  q->tone_level[ch][sb][i] = 0;
611  else
612  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
613  }
614  } else {
615  for (ch = 0; ch < q->nb_channels; ch++)
616  for (i = 0; i < 64; i++) {
617  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
618  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
619  q->tone_level[ch][sb][i] = 0;
620  else
621  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
622  }
623  }
624  }
625  }
626  }
627 
628  return;
629 }
630 
631 
646 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
647  sb_int8_array coding_method, int nb_channels,
648  int c, int superblocktype_2_3, int cm_table_select)
649 {
650  int ch, sb, j;
651  int tmp, acc, esp_40, comp;
652  int add1, add2, add3, add4;
653  int64_t multres;
654 
655  // This should never happen
656  if (nb_channels <= 0)
657  return;
658 
659  if (!superblocktype_2_3) {
660  /* This case is untested, no samples available */
662  for (ch = 0; ch < nb_channels; ch++)
663  for (sb = 0; sb < 30; sb++) {
664  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
665  add1 = tone_level_idx[ch][sb][j] - 10;
666  if (add1 < 0)
667  add1 = 0;
668  add2 = add3 = add4 = 0;
669  if (sb > 1) {
670  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
671  if (add2 < 0)
672  add2 = 0;
673  }
674  if (sb > 0) {
675  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
676  if (add3 < 0)
677  add3 = 0;
678  }
679  if (sb < 29) {
680  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
681  if (add4 < 0)
682  add4 = 0;
683  }
684  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
685  if (tmp < 0)
686  tmp = 0;
687  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
688  }
689  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
690  }
691  acc = 0;
692  for (ch = 0; ch < nb_channels; ch++)
693  for (sb = 0; sb < 30; sb++)
694  for (j = 0; j < 64; j++)
695  acc += tone_level_idx_temp[ch][sb][j];
696 
697  multres = 0x66666667 * (acc * 10);
698  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
699  for (ch = 0; ch < nb_channels; ch++)
700  for (sb = 0; sb < 30; sb++)
701  for (j = 0; j < 64; j++) {
702  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
703  if (comp < 0)
704  comp += 0xff;
705  comp /= 256; // signed shift
706  switch(sb) {
707  case 0:
708  if (comp < 30)
709  comp = 30;
710  comp += 15;
711  break;
712  case 1:
713  if (comp < 24)
714  comp = 24;
715  comp += 10;
716  break;
717  case 2:
718  case 3:
719  case 4:
720  if (comp < 16)
721  comp = 16;
722  }
723  if (comp <= 5)
724  tmp = 0;
725  else if (comp <= 10)
726  tmp = 10;
727  else if (comp <= 16)
728  tmp = 16;
729  else if (comp <= 24)
730  tmp = -1;
731  else
732  tmp = 0;
733  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
734  }
735  for (sb = 0; sb < 30; sb++)
736  fix_coding_method_array(sb, nb_channels, coding_method);
737  for (ch = 0; ch < nb_channels; ch++)
738  for (sb = 0; sb < 30; sb++)
739  for (j = 0; j < 64; j++)
740  if (sb >= 10) {
741  if (coding_method[ch][sb][j] < 10)
742  coding_method[ch][sb][j] = 10;
743  } else {
744  if (sb >= 2) {
745  if (coding_method[ch][sb][j] < 16)
746  coding_method[ch][sb][j] = 16;
747  } else {
748  if (coding_method[ch][sb][j] < 30)
749  coding_method[ch][sb][j] = 30;
750  }
751  }
752  } else { // superblocktype_2_3 != 0
753  for (ch = 0; ch < nb_channels; ch++)
754  for (sb = 0; sb < 30; sb++)
755  for (j = 0; j < 64; j++)
756  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
757  }
758 
759  return;
760 }
761 
762 
774 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
775 {
776  int sb, j, k, n, ch, run, channels;
777  int joined_stereo, zero_encoding;
778  int type34_first;
779  float type34_div = 0;
780  float type34_predictor;
781  float samples[10], sign_bits[16];
782 
783  if (length == 0) {
784  // If no data use noise
785  for (sb=sb_min; sb < sb_max; sb++)
787 
788  return;
789  }
790 
791  for (sb = sb_min; sb < sb_max; sb++) {
792  channels = q->nb_channels;
793 
794  if (q->nb_channels <= 1 || sb < 12)
795  joined_stereo = 0;
796  else if (sb >= 24)
797  joined_stereo = 1;
798  else
799  joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
800 
801  if (joined_stereo) {
802  if (BITS_LEFT(length,gb) >= 16)
803  for (j = 0; j < 16; j++)
804  sign_bits[j] = get_bits1 (gb);
805 
806  for (j = 0; j < 64; j++)
807  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
808  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
809 
811  q->coding_method)) {
813  continue;
814  }
815  channels = 1;
816  }
817 
818  for (ch = 0; ch < channels; ch++) {
820  zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
821  type34_predictor = 0.0;
822  type34_first = 1;
823 
824  for (j = 0; j < 128; ) {
825  switch (q->coding_method[ch][sb][j / 2]) {
826  case 8:
827  if (BITS_LEFT(length,gb) >= 10) {
828  if (zero_encoding) {
829  for (k = 0; k < 5; k++) {
830  if ((j + 2 * k) >= 128)
831  break;
832  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
833  }
834  } else {
835  n = get_bits(gb, 8);
836  for (k = 0; k < 5; k++)
837  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
838  }
839  for (k = 0; k < 5; k++)
840  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
841  } else {
842  for (k = 0; k < 10; k++)
843  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
844  }
845  run = 10;
846  break;
847 
848  case 10:
849  if (BITS_LEFT(length,gb) >= 1) {
850  float f = 0.81;
851 
852  if (get_bits1(gb))
853  f = -f;
854  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
855  samples[0] = f;
856  } else {
857  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
858  }
859  run = 1;
860  break;
861 
862  case 16:
863  if (BITS_LEFT(length,gb) >= 10) {
864  if (zero_encoding) {
865  for (k = 0; k < 5; k++) {
866  if ((j + k) >= 128)
867  break;
868  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
869  }
870  } else {
871  n = get_bits (gb, 8);
872  for (k = 0; k < 5; k++)
873  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
874  }
875  } else {
876  for (k = 0; k < 5; k++)
877  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
878  }
879  run = 5;
880  break;
881 
882  case 24:
883  if (BITS_LEFT(length,gb) >= 7) {
884  n = get_bits(gb, 7);
885  for (k = 0; k < 3; k++)
886  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
887  } else {
888  for (k = 0; k < 3; k++)
889  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
890  }
891  run = 3;
892  break;
893 
894  case 30:
895  if (BITS_LEFT(length,gb) >= 4) {
896  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
897  if (index < FF_ARRAY_ELEMS(type30_dequant)) {
898  samples[0] = type30_dequant[index];
899  } else
900  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
901  } else
902  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
903 
904  run = 1;
905  break;
906 
907  case 34:
908  if (BITS_LEFT(length,gb) >= 7) {
909  if (type34_first) {
910  type34_div = (float)(1 << get_bits(gb, 2));
911  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
912  type34_predictor = samples[0];
913  type34_first = 0;
914  } else {
915  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
916  if (index < FF_ARRAY_ELEMS(type34_delta)) {
917  samples[0] = type34_delta[index] / type34_div + type34_predictor;
918  type34_predictor = samples[0];
919  } else
920  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
921  }
922  } else {
923  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
924  }
925  run = 1;
926  break;
927 
928  default:
929  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
930  run = 1;
931  break;
932  }
933 
934  if (joined_stereo) {
935  for (k = 0; k < run && j + k < 128; k++) {
936  q->sb_samples[0][j + k][sb] =
937  q->tone_level[0][sb][(j + k) / 2] * samples[k];
938  if (q->nb_channels == 2) {
939  if (sign_bits[(j + k) / 8])
940  q->sb_samples[1][j + k][sb] =
941  q->tone_level[1][sb][(j + k) / 2] * -samples[k];
942  else
943  q->sb_samples[1][j + k][sb] =
944  q->tone_level[1][sb][(j + k) / 2] * samples[k];
945  }
946  }
947  } else {
948  for (k = 0; k < run; k++)
949  if ((j + k) < 128)
950  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
951  }
952 
953  j += run;
954  } // j loop
955  } // channel loop
956  } // subband loop
957 }
958 
959 
969 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
970 {
971  int i, k, run, level, diff;
972 
973  if (BITS_LEFT(length,gb) < 16)
974  return;
975  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
976 
977  quantized_coeffs[0] = level;
978 
979  for (i = 0; i < 7; ) {
980  if (BITS_LEFT(length,gb) < 16)
981  break;
982  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
983 
984  if (BITS_LEFT(length,gb) < 16)
985  break;
986  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
987 
988  for (k = 1; k <= run; k++)
989  quantized_coeffs[i + k] = (level + ((k * diff) / run));
990 
991  level += diff;
992  i += run;
993  }
994 }
995 
996 
1007 {
1008  int sb, j, k, n, ch;
1009 
1010  for (ch = 0; ch < q->nb_channels; ch++) {
1011  init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1012 
1013  if (BITS_LEFT(length,gb) < 16) {
1014  memset(q->quantized_coeffs[ch][0], 0, 8);
1015  break;
1016  }
1017  }
1018 
1019  n = q->sub_sampling + 1;
1020 
1021  for (sb = 0; sb < n; sb++)
1022  for (ch = 0; ch < q->nb_channels; ch++)
1023  for (j = 0; j < 8; j++) {
1024  if (BITS_LEFT(length,gb) < 1)
1025  break;
1026  if (get_bits1(gb)) {
1027  for (k=0; k < 8; k++) {
1028  if (BITS_LEFT(length,gb) < 16)
1029  break;
1030  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1031  }
1032  } else {
1033  for (k=0; k < 8; k++)
1034  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1035  }
1036  }
1037 
1038  n = QDM2_SB_USED(q->sub_sampling) - 4;
1039 
1040  for (sb = 0; sb < n; sb++)
1041  for (ch = 0; ch < q->nb_channels; ch++) {
1042  if (BITS_LEFT(length,gb) < 16)
1043  break;
1044  q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1045  if (sb > 19)
1046  q->tone_level_idx_hi2[ch][sb] -= 16;
1047  else
1048  for (j = 0; j < 8; j++)
1049  q->tone_level_idx_mid[ch][sb][j] = -16;
1050  }
1051 
1052  n = QDM2_SB_USED(q->sub_sampling) - 5;
1053 
1054  for (sb = 0; sb < n; sb++)
1055  for (ch = 0; ch < q->nb_channels; ch++)
1056  for (j = 0; j < 8; j++) {
1057  if (BITS_LEFT(length,gb) < 16)
1058  break;
1059  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1060  }
1061 }
1062 
1070 {
1071  GetBitContext gb;
1072  int i, j, k, n, ch, run, level, diff;
1073 
1074  init_get_bits(&gb, node->packet->data, node->packet->size*8);
1075 
1076  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1077 
1078  for (i = 1; i < n; i++)
1079  for (ch=0; ch < q->nb_channels; ch++) {
1080  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1081  q->quantized_coeffs[ch][i][0] = level;
1082 
1083  for (j = 0; j < (8 - 1); ) {
1084  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1085  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1086 
1087  for (k = 1; k <= run; k++)
1088  q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1089 
1090  level += diff;
1091  j += run;
1092  }
1093  }
1094 
1095  for (ch = 0; ch < q->nb_channels; ch++)
1096  for (i = 0; i < 8; i++)
1097  q->quantized_coeffs[ch][0][i] = 0;
1098 }
1099 
1100 
1108 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1109 {
1110  GetBitContext gb;
1111 
1112  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1113 
1114  if (length != 0) {
1115  init_tone_level_dequantization(q, &gb, length);
1116  fill_tone_level_array(q, 1);
1117  } else {
1118  fill_tone_level_array(q, 0);
1119  }
1120 }
1121 
1122 
1130 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1131 {
1132  GetBitContext gb;
1133 
1134  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1135  if (length >= 32) {
1136  int c = get_bits (&gb, 13);
1137 
1138  if (c > 3)
1141  }
1142 
1143  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1144 }
1145 
1146 
1154 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1155 {
1156  GetBitContext gb;
1157 
1158  init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1159  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1160 }
1161 
1162 /*
1163  * Process new subpackets for synthesis filter
1164  *
1165  * @param q context
1166  * @param list list with synthesis filter packets (list D)
1167  */
1169 {
1170  QDM2SubPNode *nodes[4];
1171 
1172  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1173  if (nodes[0] != NULL)
1174  process_subpacket_9(q, nodes[0]);
1175 
1176  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1177  if (nodes[1] != NULL)
1178  process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1179  else
1180  process_subpacket_10(q, NULL, 0);
1181 
1182  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1183  if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1184  process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1185  else
1186  process_subpacket_11(q, NULL, 0);
1187 
1188  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1189  if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1190  process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1191  else
1192  process_subpacket_12(q, NULL, 0);
1193 }
1194 
1195 
1196 /*
1197  * Decode superblock, fill packet lists.
1198  *
1199  * @param q context
1200  */
1202 {
1203  GetBitContext gb;
1204  QDM2SubPacket header, *packet;
1205  int i, packet_bytes, sub_packet_size, sub_packets_D;
1206  unsigned int next_index = 0;
1207 
1208  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1209  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1210  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1211 
1212  q->sub_packets_B = 0;
1213  sub_packets_D = 0;
1214 
1215  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1216 
1218  qdm2_decode_sub_packet_header(&gb, &header);
1219 
1220  if (header.type < 2 || header.type >= 8) {
1221  q->has_errors = 1;
1222  av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1223  return;
1224  }
1225 
1226  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1227  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1228 
1229  init_get_bits(&gb, header.data, header.size*8);
1230 
1231  if (header.type == 2 || header.type == 4 || header.type == 5) {
1232  int csum = 257 * get_bits(&gb, 8);
1233  csum += 2 * get_bits(&gb, 8);
1234 
1235  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1236 
1237  if (csum != 0) {
1238  q->has_errors = 1;
1239  av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1240  return;
1241  }
1242  }
1243 
1244  q->sub_packet_list_B[0].packet = NULL;
1245  q->sub_packet_list_D[0].packet = NULL;
1246 
1247  for (i = 0; i < 6; i++)
1248  if (--q->fft_level_exp[i] < 0)
1249  q->fft_level_exp[i] = 0;
1250 
1251  for (i = 0; packet_bytes > 0; i++) {
1252  int j;
1253 
1254  if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1255  SAMPLES_NEEDED_2("too many packet bytes");
1256  return;
1257  }
1258 
1259  q->sub_packet_list_A[i].next = NULL;
1260 
1261  if (i > 0) {
1262  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1263 
1264  /* seek to next block */
1265  init_get_bits(&gb, header.data, header.size*8);
1266  skip_bits(&gb, next_index*8);
1267 
1268  if (next_index >= header.size)
1269  break;
1270  }
1271 
1272  /* decode subpacket */
1273  packet = &q->sub_packets[i];
1274  qdm2_decode_sub_packet_header(&gb, packet);
1275  next_index = packet->size + get_bits_count(&gb) / 8;
1276  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1277 
1278  if (packet->type == 0)
1279  break;
1280 
1281  if (sub_packet_size > packet_bytes) {
1282  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1283  break;
1284  packet->size += packet_bytes - sub_packet_size;
1285  }
1286 
1287  packet_bytes -= sub_packet_size;
1288 
1289  /* add subpacket to 'all subpackets' list */
1290  q->sub_packet_list_A[i].packet = packet;
1291 
1292  /* add subpacket to related list */
1293  if (packet->type == 8) {
1294  SAMPLES_NEEDED_2("packet type 8");
1295  return;
1296  } else if (packet->type >= 9 && packet->type <= 12) {
1297  /* packets for MPEG Audio like Synthesis Filter */
1298  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1299  } else if (packet->type == 13) {
1300  for (j = 0; j < 6; j++)
1301  q->fft_level_exp[j] = get_bits(&gb, 6);
1302  } else if (packet->type == 14) {
1303  for (j = 0; j < 6; j++)
1304  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1305  } else if (packet->type == 15) {
1306  SAMPLES_NEEDED_2("packet type 15")
1307  return;
1308  } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1309  /* packets for FFT */
1311  }
1312  } // Packet bytes loop
1313 
1314 /* **************************************************************** */
1315  if (q->sub_packet_list_D[0].packet != NULL) {
1317  q->do_synth_filter = 1;
1318  } else if (q->do_synth_filter) {
1319  process_subpacket_10(q, NULL, 0);
1320  process_subpacket_11(q, NULL, 0);
1321  process_subpacket_12(q, NULL, 0);
1322  }
1323 /* **************************************************************** */
1324 }
1325 
1326 
1327 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1328  int offset, int duration, int channel,
1329  int exp, int phase)
1330 {
1331  if (q->fft_coefs_min_index[duration] < 0)
1333 
1334  q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1335  q->fft_coefs[q->fft_coefs_index].channel = channel;
1336  q->fft_coefs[q->fft_coefs_index].offset = offset;
1337  q->fft_coefs[q->fft_coefs_index].exp = exp;
1338  q->fft_coefs[q->fft_coefs_index].phase = phase;
1339  q->fft_coefs_index++;
1340 }
1341 
1342 
1344 {
1345  int channel, stereo, phase, exp;
1346  int local_int_4, local_int_8, stereo_phase, local_int_10;
1347  int local_int_14, stereo_exp, local_int_20, local_int_28;
1348  int n, offset;
1349 
1350  local_int_4 = 0;
1351  local_int_28 = 0;
1352  local_int_20 = 2;
1353  local_int_8 = (4 - duration);
1354  local_int_10 = 1 << (q->group_order - duration - 1);
1355  offset = 1;
1356 
1357  while (1) {
1358  if (q->superblocktype_2_3) {
1359  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1360  offset = 1;
1361  if (n == 0) {
1362  local_int_4 += local_int_10;
1363  local_int_28 += (1 << local_int_8);
1364  } else {
1365  local_int_4 += 8*local_int_10;
1366  local_int_28 += (8 << local_int_8);
1367  }
1368  }
1369  offset += (n - 2);
1370  } else {
1371  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1372  while (offset >= (local_int_10 - 1)) {
1373  offset += (1 - (local_int_10 - 1));
1374  local_int_4 += local_int_10;
1375  local_int_28 += (1 << local_int_8);
1376  }
1377  }
1378 
1379  if (local_int_4 >= q->group_size)
1380  return;
1381 
1382  local_int_14 = (offset >> local_int_8);
1383  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1384  return;
1385 
1386  if (q->nb_channels > 1) {
1387  channel = get_bits1(gb);
1388  stereo = get_bits1(gb);
1389  } else {
1390  channel = 0;
1391  stereo = 0;
1392  }
1393 
1394  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1395  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1396  exp = (exp < 0) ? 0 : exp;
1397 
1398  phase = get_bits(gb, 3);
1399  stereo_exp = 0;
1400  stereo_phase = 0;
1401 
1402  if (stereo) {
1403  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1404  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1405  if (stereo_phase < 0)
1406  stereo_phase += 8;
1407  }
1408 
1409  if (q->frequency_range > (local_int_14 + 1)) {
1410  int sub_packet = (local_int_20 + local_int_28);
1411 
1412  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1413  if (stereo)
1414  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1415  }
1416 
1417  offset++;
1418  }
1419 }
1420 
1421 
1423 {
1424  int i, j, min, max, value, type, unknown_flag;
1425  GetBitContext gb;
1426 
1427  if (q->sub_packet_list_B[0].packet == NULL)
1428  return;
1429 
1430  /* reset minimum indexes for FFT coefficients */
1431  q->fft_coefs_index = 0;
1432  for (i=0; i < 5; i++)
1433  q->fft_coefs_min_index[i] = -1;
1434 
1435  /* process subpackets ordered by type, largest type first */
1436  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1437  QDM2SubPacket *packet= NULL;
1438 
1439  /* find subpacket with largest type less than max */
1440  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1441  value = q->sub_packet_list_B[j].packet->type;
1442  if (value > min && value < max) {
1443  min = value;
1444  packet = q->sub_packet_list_B[j].packet;
1445  }
1446  }
1447 
1448  max = min;
1449 
1450  /* check for errors (?) */
1451  if (!packet)
1452  return;
1453 
1454  if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1455  return;
1456 
1457  /* decode FFT tones */
1458  init_get_bits (&gb, packet->data, packet->size*8);
1459 
1460  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1461  unknown_flag = 1;
1462  else
1463  unknown_flag = 0;
1464 
1465  type = packet->type;
1466 
1467  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1468  int duration = q->sub_sampling + 5 - (type & 15);
1469 
1470  if (duration >= 0 && duration < 4)
1471  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1472  } else if (type == 31) {
1473  for (j=0; j < 4; j++)
1474  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1475  } else if (type == 46) {
1476  for (j=0; j < 6; j++)
1477  q->fft_level_exp[j] = get_bits(&gb, 6);
1478  for (j=0; j < 4; j++)
1479  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1480  }
1481  } // Loop on B packets
1482 
1483  /* calculate maximum indexes for FFT coefficients */
1484  for (i = 0, j = -1; i < 5; i++)
1485  if (q->fft_coefs_min_index[i] >= 0) {
1486  if (j >= 0)
1488  j = i;
1489  }
1490  if (j >= 0)
1492 }
1493 
1494 
1496 {
1497  float level, f[6];
1498  int i;
1499  QDM2Complex c;
1500  const double iscale = 2.0*M_PI / 512.0;
1501 
1502  tone->phase += tone->phase_shift;
1503 
1504  /* calculate current level (maximum amplitude) of tone */
1505  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1506  c.im = level * sin(tone->phase*iscale);
1507  c.re = level * cos(tone->phase*iscale);
1508 
1509  /* generate FFT coefficients for tone */
1510  if (tone->duration >= 3 || tone->cutoff >= 3) {
1511  tone->complex[0].im += c.im;
1512  tone->complex[0].re += c.re;
1513  tone->complex[1].im -= c.im;
1514  tone->complex[1].re -= c.re;
1515  } else {
1516  f[1] = -tone->table[4];
1517  f[0] = tone->table[3] - tone->table[0];
1518  f[2] = 1.0 - tone->table[2] - tone->table[3];
1519  f[3] = tone->table[1] + tone->table[4] - 1.0;
1520  f[4] = tone->table[0] - tone->table[1];
1521  f[5] = tone->table[2];
1522  for (i = 0; i < 2; i++) {
1523  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1524  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1525  }
1526  for (i = 0; i < 4; i++) {
1527  tone->complex[i].re += c.re * f[i+2];
1528  tone->complex[i].im += c.im * f[i+2];
1529  }
1530  }
1531 
1532  /* copy the tone if it has not yet died out */
1533  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1534  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1535  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1536  }
1537 }
1538 
1539 
1540 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1541 {
1542  int i, j, ch;
1543  const double iscale = 0.25 * M_PI;
1544 
1545  for (ch = 0; ch < q->channels; ch++) {
1546  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1547  }
1548 
1549 
1550  /* apply FFT tones with duration 4 (1 FFT period) */
1551  if (q->fft_coefs_min_index[4] >= 0)
1552  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1553  float level;
1554  QDM2Complex c;
1555 
1556  if (q->fft_coefs[i].sub_packet != sub_packet)
1557  break;
1558 
1559  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1560  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];
1561 
1562  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1563  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1564  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1565  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1566  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1567  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1568  }
1569 
1570  /* generate existing FFT tones */
1571  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1573  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1574  }
1575 
1576  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1577  for (i = 0; i < 4; i++)
1578  if (q->fft_coefs_min_index[i] >= 0) {
1579  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1580  int offset, four_i;
1581  FFTTone tone;
1582 
1583  if (q->fft_coefs[j].sub_packet != sub_packet)
1584  break;
1585 
1586  four_i = (4 - i);
1587  offset = q->fft_coefs[j].offset >> four_i;
1588  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1589 
1590  if (offset < q->frequency_range) {
1591  if (offset < 2)
1592  tone.cutoff = offset;
1593  else
1594  tone.cutoff = (offset >= 60) ? 3 : 2;
1595 
1596  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];
1597  tone.complex = &q->fft.complex[ch][offset];
1598  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1599  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1600  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1601  tone.duration = i;
1602  tone.time_index = 0;
1603 
1604  qdm2_fft_generate_tone(q, &tone);
1605  }
1606  }
1607  q->fft_coefs_min_index[i] = j;
1608  }
1609 }
1610 
1611 
1612 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1613 {
1614  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1615  int i;
1616  q->fft.complex[channel][0].re *= 2.0f;
1617  q->fft.complex[channel][0].im = 0.0f;
1618  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1619  /* add samples to output buffer */
1620  for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1621  q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1622 }
1623 
1624 
1630 {
1631  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1632 
1633  /* copy sb_samples */
1634  sb_used = QDM2_SB_USED(q->sub_sampling);
1635 
1636  for (ch = 0; ch < q->channels; ch++)
1637  for (i = 0; i < 8; i++)
1638  for (k=sb_used; k < SBLIMIT; k++)
1639  q->sb_samples[ch][(8 * index) + i][k] = 0;
1640 
1641  for (ch = 0; ch < q->nb_channels; ch++) {
1642  float *samples_ptr = q->samples + ch;
1643 
1644  for (i = 0; i < 8; i++) {
1646  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1647  ff_mpa_synth_window_float, &dither_state,
1648  samples_ptr, q->nb_channels,
1649  q->sb_samples[ch][(8 * index) + i]);
1650  samples_ptr += 32 * q->nb_channels;
1651  }
1652  }
1653 
1654  /* add samples to output buffer */
1655  sub_sampling = (4 >> q->sub_sampling);
1656 
1657  for (ch = 0; ch < q->channels; ch++)
1658  for (i = 0; i < q->frame_size; i++)
1659  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1660 }
1661 
1662 
1668 static av_cold void qdm2_init(QDM2Context *q) {
1669  static int initialized = 0;
1670 
1671  if (initialized != 0)
1672  return;
1673  initialized = 1;
1674 
1675  qdm2_init_vlc();
1678  rnd_table_init();
1680 
1681  av_log(NULL, AV_LOG_DEBUG, "init done\n");
1682 }
1683 
1684 
1685 #if 0
1686 static void dump_context(QDM2Context *q)
1687 {
1688  int i;
1689 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1690  PRINT("compressed_data",q->compressed_data);
1691  PRINT("compressed_size",q->compressed_size);
1692  PRINT("frame_size",q->frame_size);
1693  PRINT("checksum_size",q->checksum_size);
1694  PRINT("channels",q->channels);
1695  PRINT("nb_channels",q->nb_channels);
1696  PRINT("fft_frame_size",q->fft_frame_size);
1697  PRINT("fft_size",q->fft_size);
1698  PRINT("sub_sampling",q->sub_sampling);
1699  PRINT("fft_order",q->fft_order);
1700  PRINT("group_order",q->group_order);
1701  PRINT("group_size",q->group_size);
1702  PRINT("sub_packet",q->sub_packet);
1703  PRINT("frequency_range",q->frequency_range);
1704  PRINT("has_errors",q->has_errors);
1705  PRINT("fft_tone_end",q->fft_tone_end);
1706  PRINT("fft_tone_start",q->fft_tone_start);
1707  PRINT("fft_coefs_index",q->fft_coefs_index);
1708  PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1709  PRINT("cm_table_select",q->cm_table_select);
1710  PRINT("noise_idx",q->noise_idx);
1711 
1712  for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1713  {
1714  FFTTone *t = &q->fft_tones[i];
1715 
1716  av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1717  av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1718 // PRINT(" level", t->level);
1719  PRINT(" phase", t->phase);
1720  PRINT(" phase_shift", t->phase_shift);
1721  PRINT(" duration", t->duration);
1722  PRINT(" samples_im", t->samples_im);
1723  PRINT(" samples_re", t->samples_re);
1724  PRINT(" table", t->table);
1725  }
1726 
1727 }
1728 #endif
1729 
1730 
1735 {
1736  QDM2Context *s = avctx->priv_data;
1737  uint8_t *extradata;
1738  int extradata_size;
1739  int tmp_val, tmp, size;
1740 
1741  /* extradata parsing
1742 
1743  Structure:
1744  wave {
1745  frma (QDM2)
1746  QDCA
1747  QDCP
1748  }
1749 
1750  32 size (including this field)
1751  32 tag (=frma)
1752  32 type (=QDM2 or QDMC)
1753 
1754  32 size (including this field, in bytes)
1755  32 tag (=QDCA) // maybe mandatory parameters
1756  32 unknown (=1)
1757  32 channels (=2)
1758  32 samplerate (=44100)
1759  32 bitrate (=96000)
1760  32 block size (=4096)
1761  32 frame size (=256) (for one channel)
1762  32 packet size (=1300)
1763 
1764  32 size (including this field, in bytes)
1765  32 tag (=QDCP) // maybe some tuneable parameters
1766  32 float1 (=1.0)
1767  32 zero ?
1768  32 float2 (=1.0)
1769  32 float3 (=1.0)
1770  32 unknown (27)
1771  32 unknown (8)
1772  32 zero ?
1773  */
1774 
1775  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1776  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1777  return -1;
1778  }
1779 
1780  extradata = avctx->extradata;
1781  extradata_size = avctx->extradata_size;
1782 
1783  while (extradata_size > 7) {
1784  if (!memcmp(extradata, "frmaQDM", 7))
1785  break;
1786  extradata++;
1787  extradata_size--;
1788  }
1789 
1790  if (extradata_size < 12) {
1791  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1792  extradata_size);
1793  return -1;
1794  }
1795 
1796  if (memcmp(extradata, "frmaQDM", 7)) {
1797  av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1798  return -1;
1799  }
1800 
1801  if (extradata[7] == 'C') {
1802 // s->is_qdmc = 1;
1803  av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1804  return -1;
1805  }
1806 
1807  extradata += 8;
1808  extradata_size -= 8;
1809 
1810  size = AV_RB32(extradata);
1811 
1812  if(size > extradata_size){
1813  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1814  extradata_size, size);
1815  return -1;
1816  }
1817 
1818  extradata += 4;
1819  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1820  if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1821  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1822  return -1;
1823  }
1824 
1825  extradata += 8;
1826 
1827  avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1828  extradata += 4;
1829  if (s->channels > MPA_MAX_CHANNELS)
1830  return AVERROR_INVALIDDATA;
1831 
1832  avctx->sample_rate = AV_RB32(extradata);
1833  extradata += 4;
1834 
1835  avctx->bit_rate = AV_RB32(extradata);
1836  extradata += 4;
1837 
1838  s->group_size = AV_RB32(extradata);
1839  extradata += 4;
1840 
1841  s->fft_size = AV_RB32(extradata);
1842  extradata += 4;
1843 
1844  s->checksum_size = AV_RB32(extradata);
1845  if (s->checksum_size >= 1U << 28) {
1846  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1847  return AVERROR_INVALIDDATA;
1848  }
1849 
1850  s->fft_order = av_log2(s->fft_size) + 1;
1851  s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1852 
1853  // something like max decodable tones
1854  s->group_order = av_log2(s->group_size) + 1;
1855  s->frame_size = s->group_size / 16; // 16 iterations per super block
1857  return AVERROR_INVALIDDATA;
1858 
1859  s->sub_sampling = s->fft_order - 7;
1860  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1861 
1862  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1863  case 0: tmp = 40; break;
1864  case 1: tmp = 48; break;
1865  case 2: tmp = 56; break;
1866  case 3: tmp = 72; break;
1867  case 4: tmp = 80; break;
1868  case 5: tmp = 100;break;
1869  default: tmp=s->sub_sampling; break;
1870  }
1871  tmp_val = 0;
1872  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1873  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1874  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1875  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1876  s->cm_table_select = tmp_val;
1877 
1878  if (s->sub_sampling == 0)
1879  tmp = 7999;
1880  else
1881  tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1882  /*
1883  0: 7999 -> 0
1884  1: 20000 -> 2
1885  2: 28000 -> 2
1886  */
1887  if (tmp < 8000)
1888  s->coeff_per_sb_select = 0;
1889  else if (tmp <= 16000)
1890  s->coeff_per_sb_select = 1;
1891  else
1892  s->coeff_per_sb_select = 2;
1893 
1894  // Fail on unknown fft order
1895  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1896  av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1897  return -1;
1898  }
1899  if (s->fft_size != (1 << (s->fft_order - 1))) {
1900  av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1901  return AVERROR_INVALIDDATA;
1902  }
1903 
1905  ff_mpadsp_init(&s->mpadsp);
1906 
1907  qdm2_init(s);
1908 
1909  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1910 
1912  avctx->coded_frame = &s->frame;
1913 
1914 // dump_context(s);
1915  return 0;
1916 }
1917 
1918 
1920 {
1921  QDM2Context *s = avctx->priv_data;
1922 
1923  ff_rdft_end(&s->rdft_ctx);
1924 
1925  return 0;
1926 }
1927 
1928 
1929 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1930 {
1931  int ch, i;
1932  const int frame_size = (q->frame_size * q->channels);
1933 
1934  /* select input buffer */
1935  q->compressed_data = in;
1937 
1938 // dump_context(q);
1939 
1940  /* copy old block, clear new block of output samples */
1941  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1942  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1943 
1944  /* decode block of QDM2 compressed data */
1945  if (q->sub_packet == 0) {
1946  q->has_errors = 0; // zero it for a new super block
1947  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1949  }
1950 
1951  /* parse subpackets */
1952  if (!q->has_errors) {
1953  if (q->sub_packet == 2)
1955 
1957  }
1958 
1959  /* sound synthesis stage 1 (FFT) */
1960  for (ch = 0; ch < q->channels; ch++) {
1961  qdm2_calculate_fft(q, ch, q->sub_packet);
1962 
1963  if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1964  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1965  return -1;
1966  }
1967  }
1968 
1969  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1970  if (!q->has_errors && q->do_synth_filter)
1972 
1973  q->sub_packet = (q->sub_packet + 1) % 16;
1974 
1975  /* clip and convert output float[] to 16bit signed samples */
1976  for (i = 0; i < frame_size; i++) {
1977  int value = (int)q->output_buffer[i];
1978 
1979  if (value > SOFTCLIP_THRESHOLD)
1980  value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1981  else if (value < -SOFTCLIP_THRESHOLD)
1982  value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1983 
1984  out[i] = value;
1985  }
1986 
1987  return 0;
1988 }
1989 
1990 
1991 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1992  int *got_frame_ptr, AVPacket *avpkt)
1993 {
1994  const uint8_t *buf = avpkt->data;
1995  int buf_size = avpkt->size;
1996  QDM2Context *s = avctx->priv_data;
1997  int16_t *out;
1998  int i, ret;
1999 
2000  if(!buf)
2001  return 0;
2002  if(buf_size < s->checksum_size)
2003  return -1;
2004 
2005  /* get output buffer */
2006  s->frame.nb_samples = 16 * s->frame_size;
2007  if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
2008  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
2009  return ret;
2010  }
2011  out = (int16_t *)s->frame.data[0];
2012 
2013  for (i = 0; i < 16; i++) {
2014  if (qdm2_decode(s, buf, out) < 0)
2015  return -1;
2016  out += s->channels * s->frame_size;
2017  }
2018 
2019  *got_frame_ptr = 1;
2020  *(AVFrame *)data = s->frame;
2021 
2022  return s->checksum_size;
2023 }
2024 
2026 {
2027  .name = "qdm2",
2028  .type = AVMEDIA_TYPE_AUDIO,
2029  .id = CODEC_ID_QDM2,
2030  .priv_data_size = sizeof(QDM2Context),
2034  .capabilities = CODEC_CAP_DR1,
2035  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2036 };